From 165c0946a863cdfd056e472d8a9673d7197ef066 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:28:01 +0100 Subject: [PATCH 001/415] ALSA: hda/hdmi: Reduce hda_jack_tbl lookup at unsol event handling Pass hda_jack_tbl object to hdmi_intrinsic_event() along with res from hdmi_unsol_event() so that we can reduce the lookup of the same hda_jack_tbl object again. Minor code refactoring. Reviewed-by: Kai Vehmanen Reviewed-by: Nikhil Mahale Link: https://lore.kernel.org/r/20200206162804.4734-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 18 +++--------------- 1 file changed, 3 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5119a9ae3d8a56..c65d16dea08c51 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -779,21 +779,9 @@ static void jack_callback(struct hda_codec *codec, check_presence_and_report(codec, jack->nid, jack->dev_id); } -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res, + struct hda_jack_tbl *jack) { - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - struct hda_jack_tbl *jack; - - if (codec->dp_mst) { - int dev_entry = - (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; - - jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry); - } else { - jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0); - } - if (!jack) - return; jack->jack_dirty = 1; codec_dbg(codec, @@ -853,7 +841,7 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) } if (subtag == 0) - hdmi_intrinsic_event(codec, res); + hdmi_intrinsic_event(codec, res, jack); else hdmi_non_intrinsic_event(codec, res); } From db8454023b7f9ca6d341cc83ce033a1f0e33d9c3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:28:02 +0100 Subject: [PATCH 002/415] ALSA: hda/hdmi: Don't use standard hda_jack for generic HDMI jacks The current HDMI codec driver code manages the jack detection in two different ways: for Intel codecs with audio component, the driver creates snd_jack objects by itself while the standard hda_jack stuff is used for the rest. This was basically because the audio component doesn't need the pin sense reading and the unsol event handling, hence it just needs to report the corresponding jacks directly. It was a bit messy but not too messy until the driver got DP-MST support for Nvidia that re-uses the part of dyn_pcm_assign feature while keeping the pin sense and the unsol event handling. Now, for DP-MST, we use hda_jack for pin sensing and unsol events but use the own snd_jack objects. Meanwhile for non-DP-MST, hda_jack is used for pin sense and unsol events, and the jacks are bound on hda_jack. Moreover, there is a polling mode support where the unsol event isn't used. For those, we also have special handling. For simplifying those messes, this patch unifies the snd_jack handling over all generic HDMI codes. The driver creates snd_jack objects just like Intel codecs did in the past but now for all devices. For the system without audio component binding, we still need the pin sense and the unsol event handling, and those are still done with the hda_jack table as before. But hda_jack is no longer used for the actual snd_jack handling. Since the hda_jack is no longer used for jack reporting, we removed snd_hda_jack_report_sync() calls, which also allowed to simplify the return type of hda_present_sense() and co. pin_idx_to_pcm_jack() was simplified as well because it behaves same for all cases now. Note that the hda_jack is still used for the simple HDMI codecs; they are really simple enough, so no big reason to change intrusively. Reviewed-by: Nikhil Mahale Link: https://lore.kernel.org/r/20200206162804.4734-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 219 ++++++++++--------------------------- 1 file changed, 57 insertions(+), 162 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c65d16dea08c51..98a8c4f97d6bfd 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -154,7 +154,6 @@ struct hdmi_spec { struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; - bool use_jack_detect; /* jack detection enabled */ bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ struct drm_audio_component_audio_ops drm_audio_ops; @@ -753,7 +752,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, * Unsolicited events */ -static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, int dev_id) @@ -764,8 +763,7 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, if (pin_idx < 0) return; mutex_lock(&spec->pcm_lock); - if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) - snd_hda_jack_report_sync(codec); + hdmi_present_sense(get_pin(spec, pin_idx), 1); mutex_unlock(&spec->pcm_lock); } @@ -1542,35 +1540,38 @@ static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) { struct hdmi_spec *spec = codec->spec; - struct snd_jack *jack = NULL; - struct hda_jack_tbl *jack_tbl; - - /* if !dyn_pcm_assign, get jack from hda_jack_tbl - * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not - * NULL even after snd_hda_jack_tbl_clear() is called to - * free snd_jack. This may cause access invalid memory - * when calling snd_jack_report + + if (per_pin->pcm_idx >= 0) + return spec->pcm_rec[per_pin->pcm_idx].jack; + else + return NULL; +} + +static void do_update_eld(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + struct hdmi_eld *eld) +{ + struct snd_jack *pcm_jack; + bool changed; + + /* + * pcm_idx >=0 before update_eld() means it is in monitor + * disconnected event. Jack must be fetched before update_eld(). */ - if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) { - jack = spec->pcm_rec[per_pin->pcm_idx].jack; - } else if (!spec->dyn_pcm_assign) { - /* - * jack tbl doesn't support DP MST - * DP MST will use dyn_pcm_assign, - * so DP MST will never come here - */ - jack_tbl = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, - per_pin->dev_id); - if (jack_tbl) - jack = jack_tbl->jack; - } - return jack; + pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); + changed = update_eld(codec, per_pin, eld); + if (!pcm_jack) + pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); + if (changed && pcm_jack) + snd_jack_report(pcm_jack, + (eld->monitor_present && eld->eld_valid) ? + SND_JACK_AVOUT : 0); } + /* update ELD and jack state via HD-audio verbs */ -static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, +static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, int repoll) { - struct hda_jack_tbl *jack; struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; @@ -1585,9 +1586,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, * the unsolicited response to avoid custom WARs. */ int present; - bool ret; bool do_repoll = false; - struct snd_jack *pcm_jack = NULL; present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id); @@ -1615,53 +1614,12 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, do_repoll = true; } - if (do_repoll) { + if (do_repoll) schedule_delayed_work(&per_pin->work, msecs_to_jiffies(300)); - } else { - /* - * pcm_idx >=0 before update_eld() means it is in monitor - * disconnected event. Jack must be fetched before - * update_eld(). - */ - pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - update_eld(codec, per_pin, eld); - if (!pcm_jack) - pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - } - - ret = !repoll || !eld->monitor_present || eld->eld_valid; - - jack = snd_hda_jack_tbl_get_mst(codec, pin_nid, per_pin->dev_id); - if (jack) { - jack->block_report = !ret; - jack->pin_sense = (eld->monitor_present && eld->eld_valid) ? - AC_PINSENSE_PRESENCE : 0; - - if (spec->dyn_pcm_assign && pcm_jack && !do_repoll) { - int state = 0; - - if (jack->pin_sense & AC_PINSENSE_PRESENCE) - state = SND_JACK_AVOUT; - snd_jack_report(pcm_jack, state); - } + else + do_update_eld(codec, per_pin, eld); - /* - * snd_hda_jack_pin_sense() call at the beginning of this - * function, updates jack->pins_sense and clears - * jack->jack_dirty, therefore snd_hda_jack_report_sync() will - * not override the jack->pin_sense. - * - * snd_hda_jack_report_sync() is superfluous for dyn_pcm_assign - * case. The jack->pin_sense update was already performed, and - * hda_jack->jack is NULL for dyn_pcm_assign. - * - * Don't call snd_hda_jack_report_sync() for - * dyn_pcm_assign. - */ - ret = ret && !spec->dyn_pcm_assign; - } mutex_unlock(&per_pin->lock); - return ret; } /* update ELD and jack state via audio component */ @@ -1670,8 +1628,6 @@ static void sync_eld_via_acomp(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; - struct snd_jack *jack = NULL; - bool changed; int size; mutex_lock(&per_pin->lock); @@ -1694,21 +1650,11 @@ static void sync_eld_via_acomp(struct hda_codec *codec, eld->eld_size = 0; } - /* pcm_idx >=0 before update_eld() means it is in monitor - * disconnected event. Jack must be fetched before update_eld() - */ - jack = pin_idx_to_pcm_jack(codec, per_pin); - changed = update_eld(codec, per_pin, eld); - if (jack == NULL) - jack = pin_idx_to_pcm_jack(codec, per_pin); - if (changed && jack) - snd_jack_report(jack, - (eld->monitor_present && eld->eld_valid) ? - SND_JACK_AVOUT : 0); + do_update_eld(codec, per_pin, eld); mutex_unlock(&per_pin->lock); } -static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; int ret; @@ -1718,16 +1664,13 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) ret = snd_hda_power_up_pm(codec); if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { snd_hda_power_down_pm(codec); - return false; + return; } - ret = hdmi_present_sense_via_verbs(per_pin, repoll); + hdmi_present_sense_via_verbs(per_pin, repoll); snd_hda_power_down_pm(codec); } else { sync_eld_via_acomp(codec, per_pin); - ret = false; /* don't call snd_hda_jack_report_sync() */ } - - return ret; } static void hdmi_repoll_eld(struct work_struct *work) @@ -1747,8 +1690,7 @@ static void hdmi_repoll_eld(struct work_struct *work) per_pin->repoll_count = 0; mutex_lock(&spec->pcm_lock); - if (hdmi_present_sense(per_pin, per_pin->repoll_count)) - snd_hda_jack_report_sync(per_pin->codec); + hdmi_present_sense(per_pin, per_pin->repoll_count); mutex_unlock(&spec->pcm_lock); } @@ -2194,15 +2136,23 @@ static void free_hdmi_jack_priv(struct snd_jack *jack) pcm->jack = NULL; } -static int add_hdmi_jack_kctl(struct hda_codec *codec, - struct hdmi_spec *spec, - int pcm_idx, - const char *name) +static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) { + char hdmi_str[32] = "HDMI/DP"; + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pcm_idx); struct snd_jack *jack; + int pcmdev = get_pcm_rec(spec, pcm_idx)->device; int err; - err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack, + if (pcmdev > 0) + sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); + if (!spec->dyn_pcm_assign && + !is_jack_detectable(codec, per_pin->pin_nid)) + strncat(hdmi_str, " Phantom", + sizeof(hdmi_str) - strlen(hdmi_str) - 1); + + err = snd_jack_new(codec->card, hdmi_str, SND_JACK_AVOUT, &jack, true, false); if (err < 0) return err; @@ -2213,48 +2163,6 @@ static int add_hdmi_jack_kctl(struct hda_codec *codec, return 0; } -static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) -{ - char hdmi_str[32] = "HDMI/DP"; - struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin; - struct hda_jack_tbl *jack; - int pcmdev = get_pcm_rec(spec, pcm_idx)->device; - bool phantom_jack; - int ret; - - if (pcmdev > 0) - sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); - - if (spec->dyn_pcm_assign) - return add_hdmi_jack_kctl(codec, spec, pcm_idx, hdmi_str); - - /* for !dyn_pcm_assign, we still use hda_jack for compatibility */ - /* if !dyn_pcm_assign, it must be non-MST mode. - * This means pcms and pins are statically mapped. - * And pcm_idx is pin_idx. - */ - per_pin = get_pin(spec, pcm_idx); - phantom_jack = !is_jack_detectable(codec, per_pin->pin_nid); - if (phantom_jack) - strncat(hdmi_str, " Phantom", - sizeof(hdmi_str) - strlen(hdmi_str) - 1); - ret = snd_hda_jack_add_kctl_mst(codec, per_pin->pin_nid, - per_pin->dev_id, hdmi_str, phantom_jack, - 0, NULL); - if (ret < 0) - return ret; - jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, - per_pin->dev_id); - if (jack == NULL) - return 0; - /* assign jack->jack to pcm_rec[].jack to - * align with dyn_pcm_assign mode - */ - spec->pcm_rec[pcm_idx].jack = jack->jack; - return 0; -} - static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2343,7 +2251,6 @@ static int generic_hdmi_init(struct hda_codec *codec) int pin_idx; mutex_lock(&spec->bind_lock); - spec->use_jack_detect = !codec->jackpoll_interval; for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; @@ -2353,12 +2260,8 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); if (codec_has_acomp(codec)) continue; - if (spec->use_jack_detect) - snd_hda_jack_detect_enable(codec, pin_nid, dev_id); - else - snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, - dev_id, - jack_callback); + snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, dev_id, + jack_callback); } mutex_unlock(&spec->bind_lock); return 0; @@ -2520,12 +2423,6 @@ static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid, unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, val); - } else { - /* if no jack entry was defined beforehand, create a new one - * at need (i.e. only when notifier is cleared) - */ - if (!use_acomp) - snd_hda_jack_detect_enable(codec, nid, dev_id); } } @@ -2541,13 +2438,11 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp, spec->use_acomp_notifier = use_acomp; spec->codec->relaxed_resume = use_acomp; /* reprogram each jack detection logic depending on the notifier */ - if (spec->use_jack_detect) { - for (i = 0; i < spec->num_pins; i++) - reprogram_jack_detect(spec->codec, - get_pin(spec, i)->pin_nid, - get_pin(spec, i)->dev_id, - use_acomp); - } + for (i = 0; i < spec->num_pins; i++) + reprogram_jack_detect(spec->codec, + get_pin(spec, i)->pin_nid, + get_pin(spec, i)->dev_id, + use_acomp); mutex_unlock(&spec->bind_lock); } From ae47e2ec5b4504cd91968e97ce2dab4ccd346fb6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:28:03 +0100 Subject: [PATCH 003/415] ALSA: hda/hdmi: Move runtime PM resume into hdmi_present_sense_via_verbs() For improving the readability, move the runtime PM handling code from hdmi_present_sense() to hdmi_present_sense_via_verbs(). Now hdmi_present_sense() became symmetric for both audio-component and legacy cases. Just a minor code refactoring. Reviewed-by: Kai Vehmanen Reviewed-by: Nikhil Mahale Link: https://lore.kernel.org/r/20200206162804.4734-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 98a8c4f97d6bfd..437177294d7843 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1587,6 +1587,11 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, */ int present; bool do_repoll = false; + int ret; + + ret = snd_hda_power_up_pm(codec); + if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) + goto out; present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id); @@ -1620,6 +1625,8 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, do_update_eld(codec, per_pin, eld); mutex_unlock(&per_pin->lock); + out: + snd_hda_power_down_pm(codec); } /* update ELD and jack state via audio component */ @@ -1657,20 +1664,11 @@ static void sync_eld_via_acomp(struct hda_codec *codec, static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; - int ret; - /* no temporary power up/down needed for component notifier */ - if (!codec_has_acomp(codec)) { - ret = snd_hda_power_up_pm(codec); - if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { - snd_hda_power_down_pm(codec); - return; - } + if (!codec_has_acomp(codec)) hdmi_present_sense_via_verbs(per_pin, repoll); - snd_hda_power_down_pm(codec); - } else { + else sync_eld_via_acomp(codec, per_pin); - } } static void hdmi_repoll_eld(struct work_struct *work) From adf615a605017089c8948f393087080e1b61b114 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:28:04 +0100 Subject: [PATCH 004/415] ALSA: hda/hdmi: Move ELD parse and jack reporting into update_eld() This is a final step of the cleanup series: move the HDMI ELD parser call into update_eld() function so that we can unify the calls. The ELD validity check is unified in update_eld(), too. Along with it, the repoll scheduling is moved to update_eld() as well, where sync_eld_via_acomp() just passes 0 for skipping it. Reviewed-by: Kai Vehmanen Reviewed-by: Nikhil Mahale Link: https://lore.kernel.org/r/20200206162804.4734-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 110 ++++++++++++++++--------------------- 1 file changed, 48 insertions(+), 62 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 437177294d7843..bb287a916dae0c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1466,21 +1466,60 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec, per_pin->channels = 0; } +static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + struct hdmi_spec *spec = codec->spec; + + if (per_pin->pcm_idx >= 0) + return spec->pcm_rec[per_pin->pcm_idx].jack; + else + return NULL; +} + /* update per_pin ELD from the given new ELD; * setup info frame and notification accordingly + * also notify ELD kctl and report jack status changes */ -static bool update_eld(struct hda_codec *codec, +static void update_eld(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin, - struct hdmi_eld *eld) + struct hdmi_eld *eld, + int repoll) { struct hdmi_eld *pin_eld = &per_pin->sink_eld; struct hdmi_spec *spec = codec->spec; + struct snd_jack *pcm_jack; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; int pcm_idx; + if (eld->eld_valid) { + if (eld->eld_size <= 0 || + snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, + eld->eld_size) < 0) { + eld->eld_valid = false; + if (repoll) { + schedule_delayed_work(&per_pin->work, + msecs_to_jiffies(300)); + return; + } + } + } + + if (!eld->eld_valid || eld->eld_size <= 0) { + eld->eld_valid = false; + eld->eld_size = 0; + } + /* for monitor disconnection, save pcm_idx firstly */ pcm_idx = per_pin->pcm_idx; + + /* + * pcm_idx >=0 before update_eld() means it is in monitor + * disconnected event. Jack must be fetched before update_eld(). + */ + pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); + if (spec->dyn_pcm_assign) { if (eld->eld_valid) { hdmi_attach_hda_pcm(spec, per_pin); @@ -1495,6 +1534,8 @@ static bool update_eld(struct hda_codec *codec, */ if (pcm_idx == -1) pcm_idx = per_pin->pcm_idx; + if (!pcm_jack) + pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); if (eld->eld_valid) snd_hdmi_show_eld(codec, &eld->info); @@ -1533,36 +1574,8 @@ static bool update_eld(struct hda_codec *codec, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id); - return eld_changed; -} - -static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec, - struct hdmi_spec_per_pin *per_pin) -{ - struct hdmi_spec *spec = codec->spec; - if (per_pin->pcm_idx >= 0) - return spec->pcm_rec[per_pin->pcm_idx].jack; - else - return NULL; -} - -static void do_update_eld(struct hda_codec *codec, - struct hdmi_spec_per_pin *per_pin, - struct hdmi_eld *eld) -{ - struct snd_jack *pcm_jack; - bool changed; - - /* - * pcm_idx >=0 before update_eld() means it is in monitor - * disconnected event. Jack must be fetched before update_eld(). - */ - pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - changed = update_eld(codec, per_pin, eld); - if (!pcm_jack) - pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - if (changed && pcm_jack) + if (eld_changed && pcm_jack) snd_jack_report(pcm_jack, (eld->monitor_present && eld->eld_valid) ? SND_JACK_AVOUT : 0); @@ -1586,7 +1599,6 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, * the unsolicited response to avoid custom WARs. */ int present; - bool do_repoll = false; int ret; ret = snd_hda_power_up_pm(codec); @@ -1610,20 +1622,9 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, if (spec->ops.pin_get_eld(codec, pin_nid, dev_id, eld->eld_buffer, &eld->eld_size) < 0) eld->eld_valid = false; - else { - if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, - eld->eld_size) < 0) - eld->eld_valid = false; - } - if (!eld->eld_valid && repoll) - do_repoll = true; } - if (do_repoll) - schedule_delayed_work(&per_pin->work, msecs_to_jiffies(300)); - else - do_update_eld(codec, per_pin, eld); - + update_eld(codec, per_pin, eld, repoll); mutex_unlock(&per_pin->lock); out: snd_hda_power_down_pm(codec); @@ -1635,29 +1636,14 @@ static void sync_eld_via_acomp(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; - int size; mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, + eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, per_pin->dev_id, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); - if (size > 0) { - size = min(size, ELD_MAX_SIZE); - if (snd_hdmi_parse_eld(codec, &eld->info, - eld->eld_buffer, size) < 0) - size = -EINVAL; - } - - if (size > 0) { - eld->eld_valid = true; - eld->eld_size = size; - } else { - eld->eld_valid = false; - eld->eld_size = 0; - } - - do_update_eld(codec, per_pin, eld); + eld->eld_valid = (eld->eld_size > 0); + update_eld(codec, per_pin, eld, 0); mutex_unlock(&per_pin->lock); } From c5bb086741c1f5cf05630dab7318433b71abb1f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:31:50 +0100 Subject: [PATCH 005/415] ALSA: via82xx: Fix endianness annotations The internal page tables are in little endian, hence they should be __le32 type. This fixes the relevant sparse warnings: sound/pci/via82xx.c:454:60: warning: incorrect type in assignment (different base types) sound/pci/via82xx.c:454:60: expected unsigned int [usertype] sound/pci/via82xx.c:454:60: got restricted __le32 [usertype] .... No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163152.6073-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 6 ++++-- sound/pci/via82xx_modem.c | 6 ++++-- 2 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 799789c8eea979..8b03e2dc503f01 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -414,6 +414,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre { unsigned int i, idx, ofs, rest; struct via82xx *chip = snd_pcm_substream_chip(substream); + __le32 *pgtbl; if (dev->table.area == NULL) { /* the start of each lists must be aligned to 8 bytes, @@ -435,6 +436,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre /* fill the entries */ idx = 0; ofs = 0; + pgtbl = (__le32 *)dev->table.area; for (i = 0; i < periods; i++) { rest = fragsize; /* fill descriptors for a period. @@ -451,7 +453,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre return -EINVAL; } addr = snd_pcm_sgbuf_get_addr(substream, ofs); - ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr); + pgtbl[idx << 1] = cpu_to_le32(addr); r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest); rest -= r; if (! rest) { @@ -466,7 +468,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre "tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); */ - ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); + pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; ofs += r; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 84e589803e2e7a..607b7100db1cfe 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -267,6 +267,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre { unsigned int i, idx, ofs, rest; struct via82xx_modem *chip = snd_pcm_substream_chip(substream); + __le32 *pgtbl; if (dev->table.area == NULL) { /* the start of each lists must be aligned to 8 bytes, @@ -288,6 +289,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre /* fill the entries */ idx = 0; ofs = 0; + pgtbl = (__le32 *)dev->table.area; for (i = 0; i < periods; i++) { rest = fragsize; /* fill descriptors for a period. @@ -304,7 +306,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre return -EINVAL; } addr = snd_pcm_sgbuf_get_addr(substream, ofs); - ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr); + pgtbl[idx << 1] = cpu_to_le32(addr); r = PAGE_SIZE - (ofs % PAGE_SIZE); if (rest < r) r = rest; @@ -321,7 +323,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre "tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); */ - ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); + pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; ofs += r; From 6a7322df2c2825bb2f45d247432a3b7d3f3ed233 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:31:51 +0100 Subject: [PATCH 006/415] ALSA: emu10k1: Fix endianness annotations The internal page tables are little endian, hence they should be __le32 type. This fixes the relevant sparse warning: sound/pci/emu10k1/emu10k1_main.c:2013:51: warning: incorrect type in assignment (different base types) sound/pci/emu10k1/emu10k1_main.c:2013:51: expected unsigned int [usertype] sound/pci/emu10k1/emu10k1_main.c:2013:51: got restricted __le32 [usertype] No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163152.6073-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index a89a7e603ca87e..6ff581733a199c 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1789,6 +1789,7 @@ int snd_emu10k1_create(struct snd_card *card, int idx, err; int is_audigy; size_t page_table_size; + __le32 *pgtbl; unsigned int silent_page; const struct snd_emu_chip_details *c; static const struct snd_device_ops ops = { @@ -2009,8 +2010,9 @@ int snd_emu10k1_create(struct snd_card *card, /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, emu->silent_page.bytes); silent_page = emu->silent_page.addr << emu->address_mode; + pgtbl = (__le32 *)emu->ptb_pages.area; for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++) - ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx); + pgtbl[idx] = cpu_to_le32(silent_page | idx); /* set up voice indices */ for (idx = 0; idx < NUM_G; idx++) { From f4caf8993e92e65cf716cbd7c30091ab6dbeb8c7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:31:52 +0100 Subject: [PATCH 007/415] ALSA: emu8000: Fix the cast to __user pointer Fixes the sparse warnings. The cast to __user pointer needs __force: sound/isa/sb/emu8000_pcm.c:528:9: warning: cast removes address space '' of expression No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163152.6073-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index e377ac93f37fd6..8e8257c574b0e1 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -435,7 +435,7 @@ enum { #define LOOP_WRITE(rec, offset, _buf, count, mode) \ do { \ struct snd_emu8000 *emu = (rec)->emu; \ - unsigned short *buf = (unsigned short *)(_buf); \ + unsigned short *buf = (__force unsigned short *)(_buf); \ snd_emu8000_write_wait(emu, 1); \ EMU8000_SMALW_WRITE(emu, offset); \ while (count > 0) { \ @@ -492,7 +492,7 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs, #define LOOP_WRITE(rec, pos, _buf, count, mode) \ do { \ struct snd_emu8000 *emu = rec->emu; \ - unsigned short *buf = (unsigned short *)(_buf); \ + unsigned short *buf = (__force unsigned short *)(_buf); \ snd_emu8000_write_wait(emu, 1); \ EMU8000_SMALW_WRITE(emu, pos + rec->loop_start[0]); \ if (rec->voices > 1) \ From 8c356c524af88c0ad4d75b30fa201ed43d9a64c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:39:38 +0100 Subject: [PATCH 008/415] ALSA: aloop: Fix PCM format assignment Fix sparse warnings about PCM format assignment regarding the strong typed snd_pcm_format_t: sound/drivers/aloop.c:352:45: warning: restricted snd_pcm_format_t degrades to integer sound/drivers/aloop.c:355:39: warning: incorrect type in assignment (different base types) sound/drivers/aloop.c:355:39: expected unsigned int format sound/drivers/aloop.c:355:39: got restricted snd_pcm_format_t [usertype] format sound/drivers/aloop.c:1435:34: warning: incorrect type in assignment (different base types) sound/drivers/aloop.c:1435:34: expected long max sound/drivers/aloop.c:1435:34: got restricted snd_pcm_format_t [usertype] sound/drivers/aloop.c:1565:39: warning: incorrect type in assignment (different base types) sound/drivers/aloop.c:1565:39: expected unsigned int format sound/drivers/aloop.c:1565:39: got restricted snd_pcm_format_t [usertype] Some code in this driver assigns an integer value to snd_pcm_format_t via control API, and they need to be with the explicit cast. No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163945.6797-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index d78a27271d6d90..251eaf1152e2c1 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -118,7 +118,7 @@ struct loopback_cable { struct loopback_setup { unsigned int notify: 1; unsigned int rate_shift; - unsigned int format; + snd_pcm_format_t format; unsigned int rate; unsigned int channels; struct snd_ctl_elem_id active_id; @@ -1432,7 +1432,7 @@ static int loopback_format_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = SNDRV_PCM_FORMAT_LAST; + uinfo->value.integer.max = (__force int)SNDRV_PCM_FORMAT_LAST; uinfo->value.integer.step = 1; return 0; } @@ -1443,7 +1443,7 @@ static int loopback_format_get(struct snd_kcontrol *kcontrol, struct loopback *loopback = snd_kcontrol_chip(kcontrol); ucontrol->value.integer.value[0] = - loopback->setup[kcontrol->id.subdevice] + (__force int)loopback->setup[kcontrol->id.subdevice] [kcontrol->id.device].format; return 0; } From b9c7d41087bc1755e01f9584f0bdbce0bb8b195d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:39:39 +0100 Subject: [PATCH 009/415] ALSA: pcm: More helper macros for reducing snd_pcm_format_t cast snd_pcm_format_t is a strong-typed integer and requires the explicit cast with __force if converted or compared with a normal integer value. Since most of use cases do iterate over all formats and test / set the mask, provide a couple of new helper macros that do the explicit cast. Link: https://lore.kernel.org/r/20200206163945.6797-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 9 +++++++++ include/sound/pcm_params.h | 7 +++++++ 2 files changed, 16 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index f657ff08f317d0..31a4b300e4c9af 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1415,6 +1415,15 @@ static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format) return 1ULL << (__force int) pcm_format; } +/** + * pcm_for_each_format - helper to iterate for each format type + * @f: the iterator variable in snd_pcm_format_t type + */ +#define pcm_for_each_format(f) \ + for ((f) = SNDRV_PCM_FORMAT_FIRST; \ + (__force int)(f) <= (__force int)SNDRV_PCM_FORMAT_LAST; \ + (f) = (__force snd_pcm_format_t)((__force int)(f) + 1)) + /* printk helpers */ #define pcm_err(pcm, fmt, args...) \ dev_err((pcm)->card->dev, fmt, ##args) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 661450a2095bd2..36f94735d23d2f 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -133,6 +133,13 @@ static inline int snd_mask_test(const struct snd_mask *mask, unsigned int val) return mask->bits[MASK_OFS(val)] & MASK_BIT(val); } +/* Most of drivers need only this one */ +static inline int snd_mask_test_format(const struct snd_mask *mask, + snd_pcm_format_t format) +{ + return snd_mask_test(mask, (__force unsigned int)format); +} + static inline int snd_mask_single(const struct snd_mask *mask) { int i, c = 0; From 5b29f05396bde54788a18633155ffa3f4d745f0d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:39:40 +0100 Subject: [PATCH 010/415] ALSA: usb-audio: Use pcm_for_each_format() macro for PCM format iterations The new macro can fix the sparse warnings gracefully: sound/usb/proc.c:73:31: warning: restricted snd_pcm_format_t degrades to integer sound/usb/proc.c:73:38: warning: restricted snd_pcm_format_t degrades to integer sound/usb/proc.c:73:61: warning: restricted snd_pcm_format_t degrades to integer No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163945.6797-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index ffbf4bd9208c45..4174ad11fca642 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -70,7 +70,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); snd_iprintf(buffer, " Format:"); - for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt) + pcm_for_each_format(fmt) if (fp->formats & pcm_format_to_bits(fmt)) snd_iprintf(buffer, " %s", snd_pcm_format_name(fmt)); From c5f72ef15cb89284b0d7b0c167d9a713bb3781e8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:39:41 +0100 Subject: [PATCH 011/415] ALSA: dummy: Use standard macros for fixing PCM format cast Simplify the code with the new macros for PCM format type iterations. This fixes the sparse warnings nicely: sound/drivers/dummy.c:906:25: warning: restricted snd_pcm_format_t degrades to integer sound/drivers/dummy.c:908:25: warning: incorrect type in argument 1 (different base types) sound/drivers/dummy.c:908:25: expected restricted snd_pcm_format_t [usertype] format sound/drivers/dummy.c:908:25: got int [assigned] i No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163945.6797-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 02ac3f4e0c02f6..b5486de08b97be 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -901,10 +901,10 @@ static int snd_card_dummy_new_mixer(struct snd_dummy *dummy) static void print_formats(struct snd_dummy *dummy, struct snd_info_buffer *buffer) { - int i; + snd_pcm_format_t i; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (dummy->pcm_hw.formats & (1ULL << i)) + pcm_for_each_format(i) { + if (dummy->pcm_hw.formats & pcm_format_to_bits(i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } } From ba71d227f4a250d7975e8296fed3aac9cb6105c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:39:42 +0100 Subject: [PATCH 012/415] ALSA: pcm: Use standard macros for fixing PCM format cast Simplify the code with the new macros for PCM format type iterations. This fixes the sparse warnings nicely: sound/core/pcm_native.c:2302:26: warning: restricted snd_pcm_format_t degrades to integer sound/core/pcm_native.c:2306:54: warning: incorrect type in argument 1 (different base types) sound/core/pcm_native.c:2306:54: expected restricted snd_pcm_format_t [usertype] format sound/core/pcm_native.c:2306:54: got unsigned int [assigned] k .... No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163945.6797-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 19 ++++++++----------- sound/core/pcm_native.c | 15 ++++++++------- 2 files changed, 16 insertions(+), 18 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 13db77771f0fef..707eb2a9d50ccc 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -884,20 +884,17 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) sformat = snd_pcm_plug_slave_format(format, sformat_mask); if ((__force int)sformat < 0 || - !snd_mask_test(sformat_mask, (__force int)sformat)) { - for (sformat = (__force snd_pcm_format_t)0; - (__force int)sformat <= (__force int)SNDRV_PCM_FORMAT_LAST; - sformat = (__force snd_pcm_format_t)((__force int)sformat + 1)) { - if (snd_mask_test(sformat_mask, (__force int)sformat) && + !snd_mask_test_format(sformat_mask, sformat)) { + pcm_for_each_format(sformat) { + if (snd_mask_test_format(sformat_mask, sformat) && snd_pcm_oss_format_to(sformat) >= 0) - break; - } - if ((__force int)sformat > (__force int)SNDRV_PCM_FORMAT_LAST) { - pcm_dbg(substream->pcm, "Cannot find a format!!!\n"); - err = -EINVAL; - goto failure; + goto format_found; } + pcm_dbg(substream->pcm, "Cannot find a format!!!\n"); + err = -EINVAL; + goto failure; } + format_found: err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, (__force int)sformat, 0); if (err < 0) goto failure; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 336406bcb59e2c..900f9cfd4646ef 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2293,21 +2293,21 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - unsigned int k; + snd_pcm_format_t k; const struct snd_interval *i = hw_param_interval_c(params, rule->deps[0]); struct snd_mask m; struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_any(&m); - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + pcm_for_each_format(k) { int bits; - if (! snd_mask_test(mask, k)) + if (!snd_mask_test_format(mask, k)) continue; bits = snd_pcm_format_physical_width(k); if (bits <= 0) continue; /* ignore invalid formats */ if ((unsigned)bits < i->min || (unsigned)bits > i->max) - snd_mask_reset(&m, k); + snd_mask_reset(&m, (__force unsigned)k); } return snd_mask_refine(mask, &m); } @@ -2316,14 +2316,15 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; - unsigned int k; + snd_pcm_format_t k; + t.min = UINT_MAX; t.max = 0; t.openmin = 0; t.openmax = 0; - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + pcm_for_each_format(k) { int bits; - if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) + if (!snd_mask_test_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) continue; bits = snd_pcm_format_physical_width(k); if (bits <= 0) From f9b0c053a29fdc0ad58e816554216523b29930f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:39:43 +0100 Subject: [PATCH 013/415] ALSA: pcm: Use a macro for parameter masks to reduce the needed cast The parameter bit mask needs often explicit cast with __force, e.g. for the PCM subformat type. Instead of adding __force at each place, which is error prone, this patch introduces a new macro and replaces the all bit shift with it. This fixes the sparse warnings like the following: sound/core/pcm_native.c:2508:30: warning: restricted snd_pcm_access_t degrades to integer No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163945.6797-7-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 32 ++++++++++++++++++-------------- 1 file changed, 18 insertions(+), 14 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 900f9cfd4646ef..559633d4702dce 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -228,6 +228,9 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, return err; } +/* macro for simplified cast */ +#define PARAM_MASK_BIT(b) (1U << (__force int)(b)) + static bool hw_support_mmap(struct snd_pcm_substream *substream) { if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) @@ -257,7 +260,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream, return -EINVAL; /* This parameter is not requested to change by a caller. */ - if (!(params->rmask & (1 << k))) + if (!(params->rmask & PARAM_MASK_BIT(k))) continue; if (trace_hw_mask_param_enabled()) @@ -271,7 +274,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream, /* Set corresponding flag so that the caller gets it. */ trace_hw_mask_param(substream, k, 0, &old_mask, m); - params->cmask |= 1 << k; + params->cmask |= PARAM_MASK_BIT(k); } return 0; @@ -293,7 +296,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream, return -EINVAL; /* This parameter is not requested to change by a caller. */ - if (!(params->rmask & (1 << k))) + if (!(params->rmask & PARAM_MASK_BIT(k))) continue; if (trace_hw_interval_param_enabled()) @@ -307,7 +310,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream, /* Set corresponding flag so that the caller gets it. */ trace_hw_interval_param(substream, k, 0, &old_interval, i); - params->cmask |= 1 << k; + params->cmask |= PARAM_MASK_BIT(k); } return 0; @@ -349,7 +352,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, * have 0 so that the parameters are never changed anymore. */ for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) - vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0; + vstamps[k] = (params->rmask & PARAM_MASK_BIT(k)) ? 1 : 0; /* Due to the above design, actual sequence number starts at 2. */ stamp = 2; @@ -417,7 +420,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, hw_param_interval(params, r->var)); } - params->cmask |= (1 << r->var); + params->cmask |= PARAM_MASK_BIT(r->var); vstamps[r->var] = stamp; again = true; } @@ -486,9 +489,9 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, params->info = 0; params->fifo_size = 0; - if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS)) + if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_SAMPLE_BITS)) params->msbits = 0; - if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) { + if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_RATE)) { params->rate_num = 0; params->rate_den = 0; } @@ -2506,16 +2509,16 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) unsigned int mask = 0; if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_INTERLEAVED); if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_NONINTERLEAVED); if (hw_support_mmap(substream)) { if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_INTERLEAVED); if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED); if (hw->info & SNDRV_PCM_INFO_COMPLEX) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_COMPLEX); } err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask); if (err < 0) @@ -2525,7 +2528,8 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; - err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD); + err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, + PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD)); if (err < 0) return err; From 89e0b9a060febbf85b985d2f3aa6afea3473276e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:39:44 +0100 Subject: [PATCH 014/415] ALSA: pcm_dmaengine: Use pcm_for_each_format() macro for PCM format iteration The new macro can fix the sparse warnings gracefully: sound/core/pcm_dmaengine.c:429:50: warning: restricted snd_pcm_format_t degrades to integer sound/core/pcm_dmaengine.c:429:55: warning: restricted snd_pcm_format_t degrades to integer sound/core/pcm_dmaengine.c:429:79: warning: restricted snd_pcm_format_t degrades to integer No functional changes, just sparse warning fixes. Link: https://lore.kernel.org/r/20200206163945.6797-8-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_dmaengine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 5749a8a4978481..b37c70864b5791 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -426,7 +426,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams( * default assumption is that it supports 1, 2 and 4 bytes * widths. */ - for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { + pcm_for_each_format(i) { int bits = snd_pcm_format_physical_width(i); /* From 49d9e43f36fd4b71a4c1888f86c09cf73be73d0b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Feb 2020 17:39:45 +0100 Subject: [PATCH 015/415] ALSA: pcm: Minor refactoring Make a common helper for validating the format type. This reduces the number of cast in the code that are needed for suppressing sparse warnings. No functional changes, just minor refactoring. Link: https://lore.kernel.org/r/20200206163945.6797-9-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_misc.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index a6a5415115343e..cf3e8c26e00ad4 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -42,6 +42,11 @@ struct pcm_format_data { /* we do lots of calculations on snd_pcm_format_t; shut up sparse */ #define INT __force int +static bool valid_format(snd_pcm_format_t format) +{ + return (INT)format >= 0 && (INT)format <= (INT)SNDRV_PCM_FORMAT_LAST; +} + static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { [SNDRV_PCM_FORMAT_S8] = { .width = 8, .phys = 8, .le = -1, .signd = 1, @@ -259,7 +264,7 @@ static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = int snd_pcm_format_signed(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].signd) < 0) return -EINVAL; @@ -307,7 +312,7 @@ EXPORT_SYMBOL(snd_pcm_format_linear); int snd_pcm_format_little_endian(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].le) < 0) return -EINVAL; @@ -343,7 +348,7 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian); int snd_pcm_format_width(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].width) == 0) return -EINVAL; @@ -361,7 +366,7 @@ EXPORT_SYMBOL(snd_pcm_format_width); int snd_pcm_format_physical_width(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].phys) == 0) return -EINVAL; @@ -394,7 +399,7 @@ EXPORT_SYMBOL(snd_pcm_format_size); */ const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format) { - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return NULL; if (! pcm_formats[(INT)format].phys) return NULL; @@ -418,7 +423,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int unsigned char *dst; const unsigned char *pat; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if (samples == 0) return 0; From 0cc629722221701a1b1030477c44dfbaca6c110d Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sat, 8 Feb 2020 22:20:06 +0000 Subject: [PATCH 016/415] ALSA: hdsp: remove redundant assignment to variable err Variable err is being assigned with a value that is never read, it is assigned a new value in the next statement. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20200208222006.37376-1-colin.king@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cc06f0a1a7e46d..227aece17e3908 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3353,7 +3353,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) return; } } else { - int err = -EINVAL; + int err; + err = hdsp_request_fw_loader(hdsp); if (err < 0) { snd_iprintf(buffer, From 4dca80b4df0a7aa4f8865b0bd6f48962c5994b1e Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sat, 8 Feb 2020 22:27:56 +0000 Subject: [PATCH 017/415] ALSA: hda: remove redundant assignment to variable timeout Variable timeout is being assigned with the value 200 that is never read, it is assigned a new value in a following do-loop. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20200208222756.37707-1-colin.king@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2609e391ce5496..9765652a73d740 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -373,7 +373,7 @@ static int azx_get_sync_time(ktime_t *device, u32 wallclk_ctr, wallclk_cycles; bool direction; u32 dma_select; - u32 timeout = 200; + u32 timeout; u32 retry_count = 0; runtime = substream->runtime; From f18b529a662c18a060e29404d64188398bbf35e4 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sat, 8 Feb 2020 22:34:43 +0000 Subject: [PATCH 018/415] ALSA: ali5451: remove redundant variable capture_flag Variable capture_flag is only ever assigned values, it is never read and hence it is redundant. Remove it. Addresses-Coverity ("Unused value") Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20200208223443.38047-1-colin.king@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 4f524a9dbbca04..4462375d2d826c 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1070,7 +1070,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, { struct snd_ali *codec = snd_pcm_substream_chip(substream); struct snd_pcm_substream *s; - unsigned int what, whati, capture_flag; + unsigned int what, whati; struct snd_ali_voice *pvoice, *evoice; unsigned int val; int do_start; @@ -1088,7 +1088,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, return -EINVAL; } - what = whati = capture_flag = 0; + what = whati = 0; snd_pcm_group_for_each_entry(s, substream) { if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) { pvoice = s->runtime->private_data; @@ -1110,8 +1110,6 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, evoice->running = 0; } snd_pcm_trigger_done(s, substream); - if (pvoice->mode) - capture_flag = 1; } } spin_lock(&codec->reg_lock); From 0e023687ca552147a76540377be4c642b1313d53 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sat, 8 Feb 2020 22:42:06 +0000 Subject: [PATCH 019/415] ALSA: info: remove redundant assignment to variable c Variable c is being assigned with the value -1 that is never read, it is assigned a new value in a following while-loop. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20200208224206.38540-1-colin.king@canonical.com Signed-off-by: Takashi Iwai --- sound/core/info.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/info.c b/sound/core/info.c index ca87ae4c30ba23..8c6bc5241df50c 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -604,7 +604,7 @@ int snd_info_card_free(struct snd_card *card) */ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { - int c = -1; + int c; if (snd_BUG_ON(!buffer || !buffer->buffer)) return 1; From 99b4f439a1c62d6a122636c5cd55ee3671dd96f4 Mon Sep 17 00:00:00 2001 From: Yu-Hsuan Hsu Date: Sun, 26 Jan 2020 00:29:17 +0800 Subject: [PATCH 020/415] ASoC: cros_ec_codec: Support setting bclk ratio Support setting bclk ratio from machine drivers. Signed-off-by: Yu-Hsuan Hsu Reviewed-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200125162917.247485-1-yuhsuan@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/cros_ec_codec.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 6a24f570c5e86f..d3dc42aa682565 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -45,6 +45,9 @@ struct cros_ec_codec_priv { /* DMIC */ atomic_t dmic_probed; + /* I2S_RX */ + uint32_t i2s_rx_bclk_ratio; + /* WoV */ bool wov_enabled; uint8_t *wov_audio_shm_p; @@ -259,6 +262,7 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, snd_soc_component_get_drvdata(component); struct ec_param_ec_codec_i2s_rx p; enum ec_codec_i2s_rx_sample_depth depth; + uint32_t bclk; int ret; if (params_rate(params) != 48000) @@ -284,15 +288,29 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - dev_dbg(component->dev, "set bclk to %u\n", - snd_soc_params_to_bclk(params)); + if (priv->i2s_rx_bclk_ratio) + bclk = params_rate(params) * priv->i2s_rx_bclk_ratio; + else + bclk = snd_soc_params_to_bclk(params); + + dev_dbg(component->dev, "set bclk to %u\n", bclk); p.cmd = EC_CODEC_I2S_RX_SET_BCLK; - p.set_bclk_param.bclk = snd_soc_params_to_bclk(params); + p.set_bclk_param.bclk = bclk; return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, (uint8_t *)&p, sizeof(p), NULL, 0); } +static int i2s_rx_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct cros_ec_codec_priv *priv = + snd_soc_component_get_drvdata(component); + + priv->i2s_rx_bclk_ratio = ratio; + return 0; +} + static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; @@ -340,6 +358,7 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops i2s_rx_dai_ops = { .hw_params = i2s_rx_hw_params, .set_fmt = i2s_rx_set_fmt, + .set_bclk_ratio = i2s_rx_set_bclk_ratio, }; static int i2s_rx_event(struct snd_soc_dapm_widget *w, From f40ed2e8db8d51c0b8155bee3a293528d9f7a956 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 29 Jan 2020 16:36:01 -0600 Subject: [PATCH 021/415] ASoC: Intel: sof_pcm512x: add support for SOF platforms with pcm512x Add support for multiple platforms, e.g. Apollolake based, using the pcm512x audio codec. The SOF developers and CI rely on the Up^2 and Hifiberry DAC+ boards based on this codec for tests. Signed-off-by: Kai Vehmanen Signed-off-by: Pan Xiuli Signed-off-by: Ranjani Sridharan Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129223603.2569-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_pcm512x.c | 428 +++++++++++++++++++++++++++ 3 files changed, 444 insertions(+) create mode 100644 sound/soc/intel/boards/sof_pcm512x.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 9ca2567d005951..755e1de19df9e3 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -457,6 +457,20 @@ config SND_SOC_INTEL_SOF_RT5682_MACH with rt5682 codec. Say Y if you have such a device. If unsure select "N". + +config SND_SOC_INTEL_SOF_PCM512x_MACH + tristate "SOF with TI PCM512x codec" + depends on I2C && ACPI + depends on (SND_SOC_SOF_HDA_AUDIO_CODEC && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ + (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + select SND_SOC_PCM512x_I2C + select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC + help + This adds support for ASoC machine driver for SOF platforms + with TI PCM512x I2S audio codec. + Say Y or m if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL if (SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index b74ddd49bd39f4..781e7ec58e47a0 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -7,6 +7,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o hda_dsp_common.o +snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o hda_dsp_common.o snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o hda_dsp_common.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o @@ -37,6 +38,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c new file mode 100644 index 00000000000000..626153bd71e779 --- /dev/null +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -0,0 +1,428 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018-2020 Intel Corporation. + +/* + * Intel SOF Machine Driver for Intel platforms with TI PCM512x codec, + * e.g. Up or Up2 with Hifiberry DAC+ HAT + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/pcm512x.h" +#include "../common/soc-intel-quirks.h" +#include "hda_dsp_common.h" + +#define NAME_SIZE 32 + +#define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) +#define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) + +/* Default: SSP5 */ +static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); + +static bool is_legacy_cpu; + +struct sof_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct sof_card_private { + struct list_head hdmi_pcm_list; +}; + +static int sof_pcm512x_quirk_cb(const struct dmi_system_id *id) +{ + sof_pcm512x_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_pcm512x_quirk_table[] = { + { + .callback = sof_pcm512x_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_PRODUCT_NAME, "UP-CHT01"), + }, + .driver_data = (void *)(SOF_PCM512X_SSP_CODEC(2)), + }, + {} +}; + +static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct sof_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int sof_pcm512x_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *codec = rtd->codec_dai->component; + + snd_soc_component_update_bits(codec, PCM512x_GPIO_EN, 0x08, 0x08); + snd_soc_component_update_bits(codec, PCM512x_GPIO_OUTPUT_4, 0x0f, 0x02); + snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1, + 0x08, 0x08); + + return 0; +} + +static int aif1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *codec = rtd->codec_dai->component; + + snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1, + 0x08, 0x08); + + return 0; +} + +static void aif1_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *codec = rtd->codec_dai->component; + + snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1, + 0x08, 0x00); +} + +static const struct snd_soc_ops sof_pcm512x_ops = { + .startup = aif1_startup, + .shutdown = aif1_shutdown, +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); + struct sof_hdmi_pcm *pcm; + + /* HDMI is not supported by SOF on Baytrail/CherryTrail */ + if (is_legacy_cpu) + return 0; + + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; + + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); + + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); +} +#else +static int sof_card_late_probe(struct snd_soc_card *card) +{ + return 0; +} +#endif + +static const struct snd_kcontrol_new sof_controls[] = { + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route sof_map[] = { + /* Speaker */ + {"Ext Spk", NULL, "OUTR"}, + {"Ext Spk", NULL, "OUTL"}, +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + +/* sof audio machine driver for pcm512x codec */ +static struct snd_soc_card sof_audio_card_pcm512x = { + .name = "pcm512x", + .owner = THIS_MODULE, + .controls = sof_controls, + .num_controls = ARRAY_SIZE(sof_controls), + .dapm_widgets = sof_widgets, + .num_dapm_widgets = ARRAY_SIZE(sof_widgets), + .dapm_routes = sof_map, + .num_dapm_routes = ARRAY_SIZE(sof_map), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +SND_SOC_DAILINK_DEF(pcm512x_component, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-104C5122:00", "pcm512x-hifi"))); +SND_SOC_DAILINK_DEF(dmic_component, + DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int dmic_be_num, + int hdmi_num) +{ + struct snd_soc_dai_link_component *idisp_components; + struct snd_soc_dai_link_component *cpus; + struct snd_soc_dai_link *links; + int i, id = 0; + + links = devm_kcalloc(dev, sof_audio_card_pcm512x.num_links, + sizeof(struct snd_soc_dai_link), GFP_KERNEL); + cpus = devm_kcalloc(dev, sof_audio_card_pcm512x.num_links, + sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); + if (!links || !cpus) + goto devm_err; + + /* codec SSP */ + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_codec); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = pcm512x_component; + links[id].num_codecs = ARRAY_SIZE(pcm512x_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_pcm512x_codec_init; + links[id].ops = &sof_pcm512x_ops; + links[id].nonatomic = true; + links[id].dpcm_playback = 1; + /* + * capture only supported with specific versions of the Hifiberry DAC+ + * links[id].dpcm_capture = 1; + */ + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + if (is_legacy_cpu) { + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "ssp%d-port", + ssp_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + } else { + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + } + id++; + + /* dmic */ + if (dmic_be_num > 0) { + /* at least we have dmic01 */ + links[id].name = "dmic01"; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; + if (dmic_be_num > 1) { + /* set up 2 BE links at most */ + links[id + 1].name = "dmic16k"; + links[id + 1].cpus = &cpus[id + 1]; + links[id + 1].cpus->dai_name = "DMIC16k Pin"; + dmic_be_num = 2; + } + } + + for (i = 0; i < dmic_be_num; i++) { + links[id].id = id; + links[id].num_cpus = 1; + links[id].codecs = dmic_component; + links[id].num_codecs = ARRAY_SIZE(dmic_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].ignore_suspend = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + id++; + } + + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kcalloc(dev, hdmi_num, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!idisp_components) + goto devm_err; + } + for (i = 1; i <= hdmi_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i); + if (!links[id].cpus->dai_name) + goto devm_err; + + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i); + if (!idisp_components[i - 1].dai_name) + goto devm_err; + + links[id].codecs = &idisp_components[i - 1]; + links[id].num_codecs = 1; + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_hdmi_init; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + id++; + } + + return links; +devm_err: + return NULL; +} + +static int sof_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai_links; + struct snd_soc_acpi_mach *mach; + struct sof_card_private *ctx; + int dmic_be_num, hdmi_num; + int ret, ssp_codec; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + hdmi_num = 0; + if (soc_intel_is_byt() || soc_intel_is_cht()) { + is_legacy_cpu = true; + dmic_be_num = 0; + /* default quirk for legacy cpu */ + sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(2); + } else { + dmic_be_num = 2; +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) + hdmi_num = 3; +#endif + } + + dmi_check_system(sof_pcm512x_quirk_table); + + dev_dbg(&pdev->dev, "sof_pcm512x_quirk = %lx\n", sof_pcm512x_quirk); + + ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK; + + /* compute number of dai links */ + sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; + + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, + dmic_be_num, hdmi_num); + if (!dai_links) + return -ENOMEM; + + sof_audio_card_pcm512x.dai_link = dai_links; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + sof_audio_card_pcm512x.dev = &pdev->dev; + mach = (&pdev->dev)->platform_data; + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_pcm512x, + mach->mach_params.platform); + if (ret) + return ret; + + snd_soc_card_set_drvdata(&sof_audio_card_pcm512x, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &sof_audio_card_pcm512x); +} + +static int sof_pcm512x_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_component *component = NULL; + + for_each_card_components(card, component) { + if (!strcmp(component->name, pcm512x_component[0].name)) { + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + +static struct platform_driver sof_audio = { + .probe = sof_audio_probe, + .remove = sof_pcm512x_remove, + .driver = { + .name = "sof_pcm512x", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(sof_audio) + +MODULE_DESCRIPTION("ASoC Intel(R) SOF + PCM512x Machine driver"); +MODULE_AUTHOR("Pierre-Louis Bossart"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_pcm512x"); From 341eb6b787c3883561bc76a7a234bf8ba48b7186 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 29 Jan 2020 16:36:02 -0600 Subject: [PATCH 022/415] ASoC: Intel: BXT: switch pcm512x based boards to sof_pcm512x Switch over Broxton platforms with the pcm512x codec from the legacy bxt-pcm512x to the new sof_pcm512x machine driver. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20200129223603.2569-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index 4a5adae1d785d6..f5092bc48364e2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -65,7 +65,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { }, { .id = "104C5122", - .drv_name = "bxt-pcm512x", + .drv_name = "sof_pcm512x", .sof_fw_filename = "sof-apl.ri", .sof_tplg_filename = "sof-apl-pcm512x.tplg", }, From 9d19426ed8f8cb5d468659caddeeeef4b147669b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 29 Jan 2020 16:36:03 -0600 Subject: [PATCH 023/415] ASoC: Intel: CHT: add support for pcm512x boards Add support for Cherrytrail boards, using the pcm512x audio codec using the new sof_pcm512x machine driver. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129223603.2569-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index d0fb43c2b9f686..2752dc9557334e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -174,6 +174,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .sof_fw_filename = "sof-cht.ri", .sof_tplg_filename = "sof-cht-cx2072x.tplg", }, + { + .id = "104C5122", + .drv_name = "sof_pcm512x", + .sof_fw_filename = "sof-cht.ri", + .sof_tplg_filename = "sof-cht-src-50khz-pcm512x.tplg", + }, + #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) /* * This is always last in the table so that it is selected only when From d2cff470452df5eba8107f267bdb6de159ba09e2 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Sun, 2 Feb 2020 07:39:17 +0000 Subject: [PATCH 024/415] ASoC: Remove unused including Remove including that don't need it. Signed-off-by: YueHaibing Link: https://lore.kernel.org/r/20200202073917.195880-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/mt6660.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index a36c416caad4db..1a3515df17641e 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -4,7 +4,6 @@ #include #include -#include #include #include #include From 918d0aba86ed8c1f4ff7f39e39e5c1b46fff2bc2 Mon Sep 17 00:00:00 2001 From: Nathan Chancellor Date: Mon, 3 Feb 2020 23:01:44 -0700 Subject: [PATCH 025/415] ASoC: wcd934x: Remove some unnecessary NULL checks Clang warns: ../sound/soc/codecs/wcd934x.c:1886:11: warning: address of array 'wcd->rx_chs' will always evaluate to 'true' [-Wpointer-bool-conversion] if (wcd->rx_chs) { ~~ ~~~~~^~~~~~ ../sound/soc/codecs/wcd934x.c:1894:11: warning: address of array 'wcd->tx_chs' will always evaluate to 'true' [-Wpointer-bool-conversion] if (wcd->tx_chs) { ~~ ~~~~~^~~~~~ 2 warnings generated. Arrays that are in the middle of a struct are never NULL so they don't need a check like this. Fixes: a61f3b4f476e ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec") Link: https://github.com/ClangBuiltLinux/linux/issues/854 Signed-off-by: Nathan Chancellor Link: https://lore.kernel.org/r/20200204060143.23393-1-natechancellor@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 158e878abd6c7d..e780ecd554d265 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1883,20 +1883,16 @@ static int wcd934x_set_channel_map(struct snd_soc_dai *dai, return -EINVAL; } - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); } - if (wcd->tx_chs) { - wcd->num_tx_port = tx_num; - for (i = 0; i < tx_num; i++) { - wcd->tx_chs[i].ch_num = tx_slot[i]; - INIT_LIST_HEAD(&wcd->tx_chs[i].list); - } + wcd->num_tx_port = tx_num; + for (i = 0; i < tx_num; i++) { + wcd->tx_chs[i].ch_num = tx_slot[i]; + INIT_LIST_HEAD(&wcd->tx_chs[i].list); } return 0; From bbf53b95ed9521625e5867522cc057bd8f1320b1 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Thu, 6 Feb 2020 14:22:13 +0800 Subject: [PATCH 026/415] ASoC: rl6231: Add new supports on rl6231 1. Increases the max limit of PLL input frequency on RL6231 shared support. 2. Add a new pll preset map. Signed-off-by: Derek Fang Link: https://lore.kernel.org/r/1580970133-14089-1-git-send-email-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rl6231.c | 1 + sound/soc/codecs/rl6231.h | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index a887d5ccb10d6c..d181c217d835b5 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -102,6 +102,7 @@ struct pll_calc_map { static const struct pll_calc_map pll_preset_table[] = { {19200000, 4096000, 23, 14, 1, false}, {19200000, 24576000, 3, 30, 3, false}, + {3840000, 24576000, 3, 30, 0, true}, }; static unsigned int find_best_div(unsigned int in, diff --git a/sound/soc/codecs/rl6231.h b/sound/soc/codecs/rl6231.h index 31a9643b0afd43..6d8ed0377296da 100644 --- a/sound/soc/codecs/rl6231.h +++ b/sound/soc/codecs/rl6231.h @@ -10,7 +10,7 @@ #ifndef __RL6231_H__ #define __RL6231_H__ -#define RL6231_PLL_INP_MAX 40000000 +#define RL6231_PLL_INP_MAX 50000000 #define RL6231_PLL_INP_MIN 256000 #define RL6231_PLL_N_MAX 0x1ff #define RL6231_PLL_K_MAX 0x1f From b5848c814cdb6ea87f77559a143c464101330c7e Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 5 Feb 2020 02:28:56 +0000 Subject: [PATCH 027/415] ASoC: rt5682: Add the field "is_sdw" of private data The field "is_sdw" is used for distinguishing the driver whether is run in soundwire mode or not. That will run the separated setting in runtime to make sure the driver can be run with the same build between i2s mode and soundwire mode. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/980b97e1ab9c4fab8bd345ec2158f1fd@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 148 +++++++++++++++++++++----------------- sound/soc/codecs/rt5682.h | 6 ++ 2 files changed, 90 insertions(+), 64 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index ae6f6121bc1b0f..82a6366201314e 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -56,6 +56,7 @@ struct rt5682_priv { struct delayed_work jack_detect_work; struct delayed_work jd_check_work; struct mutex calibrate_mutex; + bool is_sdw; int sysclk; int sysclk_src; @@ -805,10 +806,11 @@ static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux = static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux = SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum); -static void rt5682_reset(struct regmap *regmap) +static void rt5682_reset(struct rt5682_priv *rt5682) { - regmap_write(regmap, RT5682_RESET, 0); - regmap_write(regmap, RT5682_I2C_MODE, 1); + regmap_write(rt5682->regmap, RT5682_RESET, 0); + if (!rt5682->is_sdw) + regmap_write(rt5682->regmap, RT5682_I2C_MODE, 1); } /** * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters @@ -871,6 +873,8 @@ static int rt5682_button_detect(struct snd_soc_component *component) static void rt5682_enable_push_button_irq(struct snd_soc_component *component, bool enable) { + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + if (enable) { snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN); @@ -880,8 +884,15 @@ static void rt5682_enable_push_button_irq(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR); - snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, - RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN); + if (rt5682->is_sdw) + snd_soc_component_update_bits(component, + RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK | RT5682_IL_IRQ_TYPE_MASK, + RT5682_IL_IRQ_EN | RT5682_IL_IRQ_PUL); + else + snd_soc_component_update_bits(component, + RT5682_IRQ_CTRL_3, RT5682_IL_IRQ_MASK, + RT5682_IL_IRQ_EN); } else { snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS); @@ -999,62 +1010,69 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, rt5682->hs_jack = hs_jack; - if (!hs_jack) { - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK, RT5682_JD1_DIS); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, - RT5682_POW_JDH | RT5682_POW_JDL, 0); - cancel_delayed_work_sync(&rt5682->jack_detect_work); - return 0; - } + if (!rt5682->is_sdw) { + if (!hs_jack) { + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + cancel_delayed_work_sync(&rt5682->jack_detect_work); + return 0; + } - switch (rt5682->pdata.jd_src) { - case RT5682_JD1: - snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, - RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042); - snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3, - RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); - snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, - RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); - regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, - RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + switch (rt5682->pdata.jd_src) { + case RT5682_JD1: + snd_soc_component_update_bits(component, + RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, + RT5682_EXT_JD_SRC_MANUAL); + snd_soc_component_write(component, RT5682_CBJ_CTRL_1, + 0xd042); + snd_soc_component_update_bits(component, + RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, + RT5682_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, + RT5682_SAR_IL_CMD_1, RT5682_SAR_POW_MASK, + RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, RT5682_POW_IRQ | RT5682_POW_JDH | RT5682_POW_ANA, RT5682_POW_IRQ | RT5682_POW_JDH | RT5682_POW_ANA); - regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, - RT5682_PWR_JDH | RT5682_PWR_JDL, - RT5682_PWR_JDH | RT5682_PWR_JDL); - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, - RT5682_JD1_EN | RT5682_JD1_POL_NOR); - regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4, - 0x7f7f, (rt5682->pdata.btndet_delay << 8 | - rt5682->pdata.btndet_delay)); - regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5, - 0x7f7f, (rt5682->pdata.btndet_delay << 8 | - rt5682->pdata.btndet_delay)); - regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6, - 0x7f7f, (rt5682->pdata.btndet_delay << 8 | - rt5682->pdata.btndet_delay)); - regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7, - 0x7f7f, (rt5682->pdata.btndet_delay << 8 | - rt5682->pdata.btndet_delay)); - mod_delayed_work(system_power_efficient_wq, - &rt5682->jack_detect_work, msecs_to_jiffies(250)); - break; + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH | RT5682_PWR_JDL, + RT5682_PWR_JDH | RT5682_PWR_JDL); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, + RT5682_JD1_EN | RT5682_JD1_POL_NOR); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, + msecs_to_jiffies(250)); + break; - case RT5682_JD_NULL: - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK, RT5682_JD1_DIS); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, - RT5682_POW_JDH | RT5682_POW_JDL, 0); - break; + case RT5682_JD_NULL: + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + break; - default: - dev_warn(component->dev, "Wrong JD source\n"); - break; + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } } return 0; @@ -1134,11 +1152,13 @@ static void rt5682_jack_detect_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); - if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3)) - schedule_delayed_work(&rt5682->jd_check_work, 0); - else - cancel_delayed_work_sync(&rt5682->jd_check_work); + if (!rt5682->is_sdw) { + if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682->jd_check_work); + } mutex_unlock(&rt5682->calibrate_mutex); } @@ -2332,7 +2352,7 @@ static void rt5682_remove(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - rt5682_reset(rt5682->regmap); + rt5682_reset(rt5682); } #ifdef CONFIG_PM @@ -2474,7 +2494,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); - rt5682_reset(rt5682->regmap); + rt5682_reset(rt5682); regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); @@ -2586,7 +2606,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - rt5682_reset(rt5682->regmap); + rt5682_reset(rt5682); mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); @@ -2676,7 +2696,7 @@ static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); - rt5682_reset(rt5682->regmap); + rt5682_reset(rt5682); } #ifdef CONFIG_OF diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 18faaa2a49a0ce..4d3a8c41546e7b 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1091,11 +1091,17 @@ #define RT5682_JD1_POL_MASK (0x1 << 13) #define RT5682_JD1_POL_NOR (0x0 << 13) #define RT5682_JD1_POL_INV (0x1 << 13) +#define RT5682_JD1_IRQ_MASK (0x1 << 10) +#define RT5682_JD1_IRQ_LEV (0x0 << 10) +#define RT5682_JD1_IRQ_PUL (0x1 << 10) /* IRQ Control 3 (0x00b8) */ #define RT5682_IL_IRQ_MASK (0x1 << 7) #define RT5682_IL_IRQ_DIS (0x0 << 7) #define RT5682_IL_IRQ_EN (0x1 << 7) +#define RT5682_IL_IRQ_TYPE_MASK (0x1 << 4) +#define RT5682_IL_IRQ_LEV (0x0 << 4) +#define RT5682_IL_IRQ_PUL (0x1 << 4) /* GPIO Control 1 (0x00c0) */ #define RT5682_GP1_PIN_MASK (0x3 << 14) From 028476c861e3eb660d8d104ef39fccb34c04a0d5 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 3 Feb 2020 11:08:09 +0100 Subject: [PATCH 028/415] ASoC: stm32: sai: manage error when getting reset controller Return an error when the SAI driver fails to get a reset controller. Also add an error trace, except on probe defer status. Signed-off-by: Etienne Carriere Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200203100814.22944-2-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index e20267504b1671..b824ba6cb02816 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -197,12 +197,16 @@ static int stm32_sai_probe(struct platform_device *pdev) return sai->irq; /* reset */ - rst = devm_reset_control_get_exclusive(&pdev->dev, NULL); - if (!IS_ERR(rst)) { - reset_control_assert(rst); - udelay(2); - reset_control_deassert(rst); + rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(rst)) { + if (PTR_ERR(rst) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Reset controller error %ld\n", + PTR_ERR(rst)); + return PTR_ERR(rst); } + reset_control_assert(rst); + udelay(2); + reset_control_deassert(rst); /* Enable peripheral clock to allow register access */ ret = clk_prepare_enable(sai->pclk); From 7889c0082e6400ae5d07345e5573548d0999b840 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 3 Feb 2020 11:08:10 +0100 Subject: [PATCH 029/415] ASoC: stm32: spdifrx: manage error when getting reset controller Return an error when the SPDIFRX driver fails to get a reset controller. Also add an error trace, except on probe defer status. Signed-off-by: Etienne Carriere Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200203100814.22944-3-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 3769d9ce5dbef9..9f80ddf3444353 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -978,12 +978,16 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) return ret; } - rst = devm_reset_control_get_exclusive(&pdev->dev, NULL); - if (!IS_ERR(rst)) { - reset_control_assert(rst); - udelay(2); - reset_control_deassert(rst); + rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(rst)) { + if (PTR_ERR(rst) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Reset controller error %ld\n", + PTR_ERR(rst)); + return PTR_ERR(rst); } + reset_control_assert(rst); + udelay(2); + reset_control_deassert(rst); ret = devm_snd_soc_register_component(&pdev->dev, &stm32_spdifrx_component, From 158ecc65c05314cd02fcf67fa54ebef537358e5c Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 3 Feb 2020 11:08:11 +0100 Subject: [PATCH 030/415] ASoC: stm32: i2s: manage error when getting reset controller Return an error when the i2s driver fails to get a reset controller. Also add an error trace, except on probe defer status. Signed-off-by: Etienne Carriere Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200203100814.22944-4-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 3e7226a53e53ae..cdcc00d9a67ed0 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -866,12 +866,16 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, } /* Reset */ - rst = devm_reset_control_get_exclusive(&pdev->dev, NULL); - if (!IS_ERR(rst)) { - reset_control_assert(rst); - udelay(2); - reset_control_deassert(rst); + rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(rst)) { + if (PTR_ERR(rst) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Reset controller error %ld\n", + PTR_ERR(rst)); + return PTR_ERR(rst); } + reset_control_assert(rst); + udelay(2); + reset_control_deassert(rst); return 0; } From 5183e85423070d088aaf1ed07ab260e03d5a4e20 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 3 Feb 2020 11:08:12 +0100 Subject: [PATCH 031/415] ASoC: stm32: sai: improve error management on probe deferral Do not print an error trace when deferring probe for SAI driver. Signed-off-by: Etienne Carriere Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200203100814.22944-5-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai.c | 12 +++++++++--- sound/soc/stm/stm32_sai_sub.c | 11 ++++++++--- 2 files changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index b824ba6cb02816..058757c721f0a5 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -174,20 +174,26 @@ static int stm32_sai_probe(struct platform_device *pdev) if (!STM_SAI_IS_F4(sai)) { sai->pclk = devm_clk_get(&pdev->dev, "pclk"); if (IS_ERR(sai->pclk)) { - dev_err(&pdev->dev, "missing bus clock pclk\n"); + if (PTR_ERR(sai->pclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "missing bus clock pclk: %ld\n", + PTR_ERR(sai->pclk)); return PTR_ERR(sai->pclk); } } sai->clk_x8k = devm_clk_get(&pdev->dev, "x8k"); if (IS_ERR(sai->clk_x8k)) { - dev_err(&pdev->dev, "missing x8k parent clock\n"); + if (PTR_ERR(sai->clk_x8k) != -EPROBE_DEFER) + dev_err(&pdev->dev, "missing x8k parent clock: %ld\n", + PTR_ERR(sai->clk_x8k)); return PTR_ERR(sai->clk_x8k); } sai->clk_x11k = devm_clk_get(&pdev->dev, "x11k"); if (IS_ERR(sai->clk_x11k)) { - dev_err(&pdev->dev, "missing x11k parent clock\n"); + if (PTR_ERR(sai->clk_x11k) != -EPROBE_DEFER) + dev_err(&pdev->dev, "missing x11k parent clock: %ld\n", + PTR_ERR(sai->clk_x11k)); return PTR_ERR(sai->clk_x11k); } diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a8f..0bbf9ed5e48b13 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1380,7 +1380,9 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->regmap = devm_regmap_init_mmio(&pdev->dev, base, sai->regmap_config); if (IS_ERR(sai->regmap)) { - dev_err(&pdev->dev, "Failed to initialize MMIO\n"); + if (PTR_ERR(sai->regmap) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Regmap init error %ld\n", + PTR_ERR(sai->regmap)); return PTR_ERR(sai->regmap); } @@ -1471,7 +1473,9 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, of_node_put(args.np); sai->sai_ck = devm_clk_get(&pdev->dev, "sai_ck"); if (IS_ERR(sai->sai_ck)) { - dev_err(&pdev->dev, "Missing kernel clock sai_ck\n"); + if (PTR_ERR(sai->sai_ck) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Missing kernel clock sai_ck: %ld\n", + PTR_ERR(sai->sai_ck)); return PTR_ERR(sai->sai_ck); } @@ -1553,7 +1557,8 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); return ret; } From d49bd5ed24163a1a1c81d40e84295731ddd17b1c Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 3 Feb 2020 11:08:13 +0100 Subject: [PATCH 032/415] ASoC: stm32: spdifrx: improve error management on probe deferral Do not print an error trace when deferring probe for SPDIFRX driver. Signed-off-by: Etienne Carriere Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200203100814.22944-6-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 9f80ddf3444353..49766afdae6163 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -406,7 +406,9 @@ static int stm32_spdifrx_dma_ctrl_register(struct device *dev, spdifrx->ctrl_chan = dma_request_chan(dev, "rx-ctrl"); if (IS_ERR(spdifrx->ctrl_chan)) { - dev_err(dev, "dma_request_slave_channel failed\n"); + if (PTR_ERR(spdifrx->ctrl_chan) != -EPROBE_DEFER) + dev_err(dev, "dma_request_slave_channel error %ld\n", + PTR_ERR(spdifrx->ctrl_chan)); return PTR_ERR(spdifrx->ctrl_chan); } @@ -929,7 +931,9 @@ static int stm32_spdifrx_parse_of(struct platform_device *pdev, spdifrx->kclk = devm_clk_get(&pdev->dev, "kclk"); if (IS_ERR(spdifrx->kclk)) { - dev_err(&pdev->dev, "Could not get kclk\n"); + if (PTR_ERR(spdifrx->kclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get kclk: %ld\n", + PTR_ERR(spdifrx->kclk)); return PTR_ERR(spdifrx->kclk); } @@ -967,7 +971,9 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) spdifrx->base, spdifrx->regmap_conf); if (IS_ERR(spdifrx->regmap)) { - dev_err(&pdev->dev, "Regmap init failed\n"); + if (PTR_ERR(spdifrx->regmap) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Regmap init error %ld\n", + PTR_ERR(spdifrx->regmap)); return PTR_ERR(spdifrx->regmap); } @@ -1003,7 +1009,8 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) pcm_config = &stm32_spdifrx_pcm_config; ret = devm_snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0); if (ret) { - dev_err(&pdev->dev, "PCM DMA register returned %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); goto error; } From 04dd656e8d506c12f5e97a24089b2991f5f00984 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 3 Feb 2020 11:08:14 +0100 Subject: [PATCH 033/415] ASoC: stm32: i2s: improve error management on probe deferral Do not print an error trace when deferring probe for I2S driver. Signed-off-by: Etienne Carriere Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200203100814.22944-7-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 25 +++++++++++++++++++------ 1 file changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index cdcc00d9a67ed0..2478405727c378 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -831,25 +831,33 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, /* Get clocks */ i2s->pclk = devm_clk_get(&pdev->dev, "pclk"); if (IS_ERR(i2s->pclk)) { - dev_err(&pdev->dev, "Could not get pclk\n"); + if (PTR_ERR(i2s->pclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get pclk: %ld\n", + PTR_ERR(i2s->pclk)); return PTR_ERR(i2s->pclk); } i2s->i2sclk = devm_clk_get(&pdev->dev, "i2sclk"); if (IS_ERR(i2s->i2sclk)) { - dev_err(&pdev->dev, "Could not get i2sclk\n"); + if (PTR_ERR(i2s->i2sclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get i2sclk: %ld\n", + PTR_ERR(i2s->i2sclk)); return PTR_ERR(i2s->i2sclk); } i2s->x8kclk = devm_clk_get(&pdev->dev, "x8k"); if (IS_ERR(i2s->x8kclk)) { - dev_err(&pdev->dev, "missing x8k parent clock\n"); + if (PTR_ERR(i2s->x8kclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get x8k parent clock: %ld\n", + PTR_ERR(i2s->x8kclk)); return PTR_ERR(i2s->x8kclk); } i2s->x11kclk = devm_clk_get(&pdev->dev, "x11k"); if (IS_ERR(i2s->x11kclk)) { - dev_err(&pdev->dev, "missing x11k parent clock\n"); + if (PTR_ERR(i2s->x11kclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get x11k parent clock: %ld\n", + PTR_ERR(i2s->x11kclk)); return PTR_ERR(i2s->x11kclk); } @@ -907,7 +915,9 @@ static int stm32_i2s_probe(struct platform_device *pdev) i2s->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "pclk", i2s->base, i2s->regmap_conf); if (IS_ERR(i2s->regmap)) { - dev_err(&pdev->dev, "regmap init failed\n"); + if (PTR_ERR(i2s->regmap) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Regmap init error %ld\n", + PTR_ERR(i2s->regmap)); return PTR_ERR(i2s->regmap); } @@ -918,8 +928,11 @@ static int stm32_i2s_probe(struct platform_device *pdev) ret = devm_snd_dmaengine_pcm_register(&pdev->dev, &stm32_i2s_pcm_config, 0); - if (ret) + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); return ret; + } /* Set SPI/I2S in i2s mode */ ret = regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, From 3025571edd9df653e1ad649f0638368a39d1bbb5 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sat, 8 Feb 2020 22:07:20 +0000 Subject: [PATCH 034/415] ASoC: Intel: mrfld: return error codes when an error occurs Currently function sst_platform_get_resources always returns zero and error return codes set by the function are never returned. Fix this by returning the error return code in variable ret rather than the hard coded zero. Addresses-Coverity: ("Unused value") Fixes: f533a035e4da ("ASoC: Intel: mrfld - create separate module for pci part") Signed-off-by: Colin Ian King Acked-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200208220720.36657-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_pci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c index d952719bc09862..5862fe96808369 100644 --- a/sound/soc/intel/atom/sst/sst_pci.c +++ b/sound/soc/intel/atom/sst/sst_pci.c @@ -99,7 +99,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); do_release_regions: pci_release_regions(pci); - return 0; + return ret; } /* From 1646484ed2430e37f00945db4755449d54354b57 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sat, 8 Feb 2020 22:15:29 +0000 Subject: [PATCH 035/415] ASoC: rt5659: remove redundant assignment to variable idx Variable idx is being assigned with a value that is never idx, it is assigned a new value a couple of statements later. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20200208221529.37105-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5659.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index e66d08398f7469..89e0f58512fa41 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -1604,7 +1604,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); - int pd, idx = -EINVAL; + int pd, idx; pd = rl6231_get_pre_div(rt5659->regmap, RT5659_ADDA_CLK_1, RT5659_I2S_PD1_SFT); From 0d4416446897a91bb19ba837b97b607caea59a8f Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 10 Feb 2020 10:30:27 +0100 Subject: [PATCH 036/415] spi: pxa2xx: Enable support for compile-testing m68k/allmodconfig: WARNING: unmet direct dependencies detected for SPI_PXA2XX Depends on [n]: SPI [=y] && SPI_MASTER [=y] && (ARCH_PXA || ARCH_MMP || PCI [=n] || ACPI) Selected by [m]: - SND_SOC_INTEL_BDW_RT5677_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_INTEL_HASWELL [=n] || SND_SOC_SOF_BROADWELL [=m]) && I2C [=m] && (I2C_DESIGNWARE_PLATFORM [=m] || COMPILE_TEST [=y]) && (GPIOLIB [=y] || COMPILE_TEST [=y]) && (X86_INTEL_LPSS || COMPILE_TEST [=y]) && SPI_MASTER [=y] This happens because SND_SOC_INTEL_BDW_RT5677_MACH selects SPI_PXA2XX, and the former depends on COMPILE_TEST, while the latter does not. Fix this by enabling compile-testing for SPI_PXA2XX. Fixes: 630db1549356f644 ("ASoC: Intel: bdw-rt5677: fix Kconfig dependencies") Signed-off-by: Geert Uytterhoeven Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200210093027.6672-1-geert@linux-m68k.org Signed-off-by: Mark Brown --- drivers/spi/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/spi/Kconfig b/drivers/spi/Kconfig index d6ed0c355954f0..912cd6e357266f 100644 --- a/drivers/spi/Kconfig +++ b/drivers/spi/Kconfig @@ -551,7 +551,7 @@ config SPI_PPC4xx config SPI_PXA2XX tristate "PXA2xx SSP SPI master" - depends on (ARCH_PXA || ARCH_MMP || PCI || ACPI) + depends on ARCH_PXA || ARCH_MMP || PCI || ACPI || COMPILE_TEST select PXA_SSP if ARCH_PXA || ARCH_MMP help This enables using a PXA2xx or Sodaville SSP port as a SPI master From 9da9ace29ba556d5a2ae6d044070daba5b7d3638 Mon Sep 17 00:00:00 2001 From: Paul Olaru Date: Mon, 10 Feb 2020 11:58:14 +0200 Subject: [PATCH 037/415] ASoC: SOF: Rename i.MX8 platform to i.MX8X i.MX8 and i.MX8X platforms are very similar and were treated the same. Anyhow, we need to account for the differences somehow. Current supported platform is i.MX8QXP which is from i.MX8X family. Rename i.MX8 platform to i.MX8X to prepare for future i.MX8 platforms. Signed-off-by: Paul Olaru Signed-off-by: Daniel Baluta Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200210095817.13226-2-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8.c | 10 +++++----- sound/soc/sof/sof-of-dev.c | 4 ++-- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index b2556f5e2871fa..9ffc2a955e4f9c 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -138,7 +138,7 @@ static int imx8_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) /* * DSP control. */ -static int imx8_run(struct snd_sof_dev *sdev) +static int imx8x_run(struct snd_sof_dev *sdev) { struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private; int ret; @@ -360,13 +360,13 @@ static struct snd_soc_dai_driver imx8_dai[] = { }, }; -/* i.MX8 ops */ -struct snd_sof_dsp_ops sof_imx8_ops = { +/* i.MX8X ops */ +struct snd_sof_dsp_ops sof_imx8x_ops = { /* probe and remove */ .probe = imx8_probe, .remove = imx8_remove, /* DSP core boot */ - .run = imx8_run, + .run = imx8x_run, /* Block IO */ .block_read = sof_block_read, @@ -398,6 +398,6 @@ struct snd_sof_dsp_ops sof_imx8_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP }; -EXPORT_SYMBOL(sof_imx8_ops); +EXPORT_SYMBOL(sof_imx8x_ops); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index 39ea8af6213f32..2da1bd859d98da 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -19,9 +19,9 @@ extern struct snd_sof_dsp_ops sof_imx8_ops; static struct sof_dev_desc sof_of_imx8qxp_desc = { .default_fw_path = "imx/sof", .default_tplg_path = "imx/sof-tplg", - .default_fw_filename = "sof-imx8.ri", + .default_fw_filename = "sof-imx8x.ri", .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", - .ops = &sof_imx8_ops, + .ops = &sof_imx8x_ops, }; #endif From acfa52027bb64b8f93324da2277ff662c7a95679 Mon Sep 17 00:00:00 2001 From: Paul Olaru Date: Mon, 10 Feb 2020 11:58:15 +0200 Subject: [PATCH 038/415] ASoC: SOF: imx8: Add ops for i.MX8QM i.MX8QM and i.MX8QXP are mostly identical platforms with minor hardware differences. One of these differences affects the firmware boot process, requiring the run operation to differ. All other ops are reused. Signed-off-by: Paul Olaru Signed-off-by: Daniel Baluta Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200210095817.13226-3-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8.c | 51 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 9ffc2a955e4f9c..b692752b21788b 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -178,6 +178,24 @@ static int imx8x_run(struct snd_sof_dev *sdev) return 0; } +static int imx8_run(struct snd_sof_dev *sdev) +{ + struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private; + int ret; + + ret = imx_sc_misc_set_control(dsp_priv->sc_ipc, IMX_SC_R_DSP, + IMX_SC_C_OFS_SEL, 0); + if (ret < 0) { + dev_err(sdev->dev, "Error system address offset source select\n"); + return ret; + } + + imx_sc_pm_cpu_start(dsp_priv->sc_ipc, IMX_SC_R_DSP, true, + RESET_VECTOR_VADDR); + + return 0; +} + static int imx8_probe(struct snd_sof_dev *sdev) { struct platform_device *pdev = @@ -360,6 +378,39 @@ static struct snd_soc_dai_driver imx8_dai[] = { }, }; +/* i.MX8 ops */ +struct snd_sof_dsp_ops sof_imx8_ops = { + /* probe and remove */ + .probe = imx8_probe, + .remove = imx8_remove, + /* DSP core boot */ + .run = imx8_run, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* ipc */ + .send_msg = imx8_send_msg, + .fw_ready = sof_fw_ready, + .get_mailbox_offset = imx8_get_mailbox_offset, + .get_window_offset = imx8_get_window_offset, + + .ipc_msg_data = imx8_ipc_msg_data, + .ipc_pcm_params = imx8_ipc_pcm_params, + + /* module loading */ + .load_module = snd_sof_parse_module_memcpy, + .get_bar_index = imx8_get_bar_index, + /* firmware loading */ + .load_firmware = snd_sof_load_firmware_memcpy, + + /* DAI drivers */ + .drv = imx8_dai, + .num_drv = 1, /* we have only 1 ESAI interface on i.MX8 */ +}; +EXPORT_SYMBOL(sof_imx8_ops); + /* i.MX8X ops */ struct snd_sof_dsp_ops sof_imx8x_ops = { /* probe and remove */ From f831ebf2faa598793a7ec327847c61dbeabba601 Mon Sep 17 00:00:00 2001 From: Paul Olaru Date: Mon, 10 Feb 2020 11:58:16 +0200 Subject: [PATCH 039/415] ASoC: SOF: Add i.MX8QM device descriptor Add SOF device and DT descriptors for i.MX8QM platform. Signed-off-by: Paul Olaru Signed-off-by: Daniel Baluta Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200210095817.13226-4-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-of-dev.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index 2da1bd859d98da..16e49f2ee6291b 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -13,6 +13,7 @@ #include "ops.h" extern struct snd_sof_dsp_ops sof_imx8_ops; +extern struct snd_sof_dsp_ops sof_imx8x_ops; /* platform specific devices */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) @@ -23,6 +24,14 @@ static struct sof_dev_desc sof_of_imx8qxp_desc = { .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", .ops = &sof_imx8x_ops, }; + +static struct sof_dev_desc sof_of_imx8qm_desc = { + .default_fw_path = "imx/sof", + .default_tplg_path = "imx/sof-tplg", + .default_fw_filename = "sof-imx8.ri", + .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", + .ops = &sof_imx8_ops, +}; #endif static const struct dev_pm_ops sof_of_pm = { @@ -103,6 +112,7 @@ static int sof_of_remove(struct platform_device *pdev) static const struct of_device_id sof_of_ids[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) { .compatible = "fsl,imx8qxp-dsp", .data = &sof_of_imx8qxp_desc}, + { .compatible = "fsl,imx8qm-dsp", .data = &sof_of_imx8qm_desc}, #endif { } }; From ea00d95200d02ece71f5814d41b14f2eb16d598b Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Fri, 7 Feb 2020 10:13:51 +0100 Subject: [PATCH 040/415] ASoC: Use imply for SND_SOC_ALL_CODECS Currently SND_SOC_ALL_CODECS selects the config symbols for all codec drivers. As "select" bypasses dependencies, lots of "select" statements need explicit dependencies, which are hard to get right, and hard to maintain[*]. Fix this by using "imply" instead, which is a weak version of "select", and which obeys dependencies of target symbols. Add dependencies to invisible symbols that are currently selected only if their dependencies are fulfilled. [*] See e.g. commit 13426feaf46c48fc ("ASoC: wcd934x: Add missing COMMON_CLK dependency to SND_SOC_ALL_CODECS"). Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/20200207091351.18133-1-geert@linux-m68k.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 520 ++++++++++++++++++++------------------- 1 file changed, 263 insertions(+), 257 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7e90f5d8309713..7a14b1c416b55e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -14,262 +14,262 @@ menu "CODEC drivers" config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" depends on COMPILE_TEST - select SND_SOC_88PM860X if MFD_88PM860X - select SND_SOC_L3 - select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC - select SND_SOC_AD1836 if SPI_MASTER - select SND_SOC_AD193X_SPI if SPI_MASTER - select SND_SOC_AD193X_I2C if I2C - select SND_SOC_AD1980 if SND_SOC_AC97_BUS - select SND_SOC_AD73311 - select SND_SOC_ADAU1373 if I2C - select SND_SOC_ADAU1761_I2C if I2C - select SND_SOC_ADAU1761_SPI if SPI - select SND_SOC_ADAU1781_I2C if I2C - select SND_SOC_ADAU1781_SPI if SPI - select SND_SOC_ADAV801 if SPI_MASTER - select SND_SOC_ADAV803 if I2C - select SND_SOC_ADAU1977_SPI if SPI_MASTER - select SND_SOC_ADAU1977_I2C if I2C - select SND_SOC_ADAU1701 if I2C - select SND_SOC_ADAU7002 - select SND_SOC_ADAU7118_I2C if I2C - select SND_SOC_ADAU7118_HW - select SND_SOC_ADS117X - select SND_SOC_AK4104 if SPI_MASTER - select SND_SOC_AK4118 if I2C - select SND_SOC_AK4458 if I2C - select SND_SOC_AK4535 if I2C - select SND_SOC_AK4554 - select SND_SOC_AK4613 if I2C - select SND_SOC_AK4641 if I2C - select SND_SOC_AK4642 if I2C - select SND_SOC_AK4671 if I2C - select SND_SOC_AK5386 - select SND_SOC_AK5558 if I2C - select SND_SOC_ALC5623 if I2C - select SND_SOC_ALC5632 if I2C - select SND_SOC_BT_SCO - select SND_SOC_BD28623 - select SND_SOC_CQ0093VC - select SND_SOC_CROS_EC_CODEC if CROS_EC - select SND_SOC_CS35L32 if I2C - select SND_SOC_CS35L33 if I2C - select SND_SOC_CS35L34 if I2C - select SND_SOC_CS35L35 if I2C - select SND_SOC_CS35L36 if I2C - select SND_SOC_CS42L42 if I2C - select SND_SOC_CS42L51_I2C if I2C - select SND_SOC_CS42L52 if I2C && INPUT - select SND_SOC_CS42L56 if I2C && INPUT - select SND_SOC_CS42L73 if I2C - select SND_SOC_CS4265 if I2C - select SND_SOC_CS4270 if I2C - select SND_SOC_CS4271_I2C if I2C - select SND_SOC_CS4271_SPI if SPI_MASTER - select SND_SOC_CS42XX8_I2C if I2C - select SND_SOC_CS43130 if I2C - select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI - select SND_SOC_CS4349 if I2C - select SND_SOC_CS47L15 if MFD_CS47L15 - select SND_SOC_CS47L24 if MFD_CS47L24 - select SND_SOC_CS47L35 if MFD_CS47L35 - select SND_SOC_CS47L85 if MFD_CS47L85 - select SND_SOC_CS47L90 if MFD_CS47L90 - select SND_SOC_CS47L92 if MFD_CS47L92 - select SND_SOC_CS53L30 if I2C - select SND_SOC_CX20442 if TTY - select SND_SOC_CX2072X if I2C - select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI - select SND_SOC_DA7213 if I2C - select SND_SOC_DA7218 if I2C - select SND_SOC_DA7219 if I2C - select SND_SOC_DA732X if I2C - select SND_SOC_DA9055 if I2C - select SND_SOC_DMIC if GPIOLIB - select SND_SOC_ES8316 if I2C - select SND_SOC_ES8328_SPI if SPI_MASTER - select SND_SOC_ES8328_I2C if I2C - select SND_SOC_ES7134 - select SND_SOC_ES7241 - select SND_SOC_GTM601 - select SND_SOC_HDAC_HDMI - select SND_SOC_HDAC_HDA - select SND_SOC_ICS43432 - select SND_SOC_INNO_RK3036 - select SND_SOC_ISABELLE if I2C - select SND_SOC_JZ4740_CODEC - select SND_SOC_JZ4725B_CODEC - select SND_SOC_JZ4770_CODEC - select SND_SOC_LM4857 if I2C - select SND_SOC_LM49453 if I2C - select SND_SOC_LOCHNAGAR_SC if MFD_LOCHNAGAR - select SND_SOC_MAX98088 if I2C - select SND_SOC_MAX98090 if I2C - select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A if GPIOLIB - select SND_SOC_MAX98371 if I2C - select SND_SOC_MAX98504 if I2C - select SND_SOC_MAX9867 if I2C - select SND_SOC_MAX98925 if I2C - select SND_SOC_MAX98926 if I2C - select SND_SOC_MAX98927 if I2C - select SND_SOC_MAX98373 if I2C - select SND_SOC_MAX9850 if I2C - select SND_SOC_MAX9860 if I2C - select SND_SOC_MAX9759 - select SND_SOC_MAX9768 if I2C - select SND_SOC_MAX9877 if I2C - select SND_SOC_MC13783 if MFD_MC13XXX - select SND_SOC_ML26124 if I2C - select SND_SOC_MT6351 if MTK_PMIC_WRAP - select SND_SOC_MT6358 if MTK_PMIC_WRAP - select SND_SOC_MT6660 if I2C - select SND_SOC_NAU8540 if I2C - select SND_SOC_NAU8810 if I2C - select SND_SOC_NAU8822 if I2C - select SND_SOC_NAU8824 if I2C - select SND_SOC_NAU8825 if I2C - select SND_SOC_HDMI_CODEC - select SND_SOC_PCM1681 if I2C - select SND_SOC_PCM1789_I2C if I2C - select SND_SOC_PCM179X_I2C if I2C - select SND_SOC_PCM179X_SPI if SPI_MASTER - select SND_SOC_PCM186X_I2C if I2C - select SND_SOC_PCM186X_SPI if SPI_MASTER - select SND_SOC_PCM3008 - select SND_SOC_PCM3060_I2C if I2C - select SND_SOC_PCM3060_SPI if SPI_MASTER - select SND_SOC_PCM3168A_I2C if I2C - select SND_SOC_PCM3168A_SPI if SPI_MASTER - select SND_SOC_PCM5102A - select SND_SOC_PCM512x_I2C if I2C - select SND_SOC_PCM512x_SPI if SPI_MASTER - select SND_SOC_RK3328 - select SND_SOC_RT274 if I2C - select SND_SOC_RT286 if I2C - select SND_SOC_RT298 if I2C - select SND_SOC_RT1011 if I2C - select SND_SOC_RT1015 if I2C - select SND_SOC_RT1305 if I2C - select SND_SOC_RT1308 if I2C - select SND_SOC_RT5514 if I2C - select SND_SOC_RT5616 if I2C - select SND_SOC_RT5631 if I2C - select SND_SOC_RT5640 if I2C - select SND_SOC_RT5645 if I2C - select SND_SOC_RT5651 if I2C - select SND_SOC_RT5659 if I2C - select SND_SOC_RT5660 if I2C - select SND_SOC_RT5663 if I2C - select SND_SOC_RT5665 if I2C - select SND_SOC_RT5668 if I2C - select SND_SOC_RT5670 if I2C - select SND_SOC_RT5677 if I2C && SPI_MASTER - select SND_SOC_RT5682 if I2C - select SND_SOC_RT700_SDW if SOUNDWIRE - select SND_SOC_RT711_SDW if SOUNDWIRE - select SND_SOC_RT715_SDW if SOUNDWIRE - select SND_SOC_RT1308_SDW if SOUNDWIRE - select SND_SOC_SGTL5000 if I2C - select SND_SOC_SI476X if MFD_SI476X_CORE - select SND_SOC_SIMPLE_AMPLIFIER - select SND_SOC_SIRF_AUDIO_CODEC - select SND_SOC_SPDIF - select SND_SOC_SSM2305 - select SND_SOC_SSM2518 if I2C - select SND_SOC_SSM2602_SPI if SPI_MASTER - select SND_SOC_SSM2602_I2C if I2C - select SND_SOC_SSM4567 if I2C - select SND_SOC_STA32X if I2C - select SND_SOC_STA350 if I2C - select SND_SOC_STA529 if I2C - select SND_SOC_STAC9766 if SND_SOC_AC97_BUS - select SND_SOC_STI_SAS - select SND_SOC_TAS2552 if I2C - select SND_SOC_TAS2562 if I2C - select SND_SOC_TAS2770 if I2C - select SND_SOC_TAS5086 if I2C - select SND_SOC_TAS571X if I2C - select SND_SOC_TAS5720 if I2C - select SND_SOC_TAS6424 if I2C - select SND_SOC_TDA7419 if I2C - select SND_SOC_TFA9879 if I2C - select SND_SOC_TLV320AIC23_I2C if I2C - select SND_SOC_TLV320AIC23_SPI if SPI_MASTER - select SND_SOC_TLV320AIC26 if SPI_MASTER - select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK - select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK - select SND_SOC_TLV320AIC3X if I2C - select SND_SOC_TPA6130A2 if I2C - select SND_SOC_TLV320DAC33 if I2C - select SND_SOC_TSCS42XX if I2C - select SND_SOC_TSCS454 if I2C - select SND_SOC_TS3A227E if I2C - select SND_SOC_TWL4030 if TWL4030_CORE - select SND_SOC_TWL6040 if TWL6040_CORE - select SND_SOC_UDA1334 if GPIOLIB - select SND_SOC_UDA134X - select SND_SOC_UDA1380 if I2C - select SND_SOC_WCD9335 if SLIMBUS - select SND_SOC_WCD934X if MFD_WCD934X && COMMON_CLK - select SND_SOC_WL1273 if MFD_WL1273_CORE - select SND_SOC_WM0010 if SPI_MASTER - select SND_SOC_WM1250_EV1 if I2C - select SND_SOC_WM2000 if I2C - select SND_SOC_WM2200 if I2C - select SND_SOC_WM5100 if I2C - select SND_SOC_WM5102 if MFD_WM5102 - select SND_SOC_WM5110 if MFD_WM5110 - select SND_SOC_WM8350 if MFD_WM8350 - select SND_SOC_WM8400 if MFD_WM8400 - select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8523 if I2C - select SND_SOC_WM8524 if GPIOLIB - select SND_SOC_WM8580 if I2C - select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8727 - select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8737 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8770 if SPI_MASTER - select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8782 - select SND_SOC_WM8804_I2C if I2C - select SND_SOC_WM8804_SPI if SPI_MASTER - select SND_SOC_WM8900 if I2C - select SND_SOC_WM8903 if I2C - select SND_SOC_WM8904 if I2C - select SND_SOC_WM8940 if I2C - select SND_SOC_WM8955 if I2C - select SND_SOC_WM8960 if I2C - select SND_SOC_WM8961 if I2C - select SND_SOC_WM8962 if I2C && INPUT - select SND_SOC_WM8971 if I2C - select SND_SOC_WM8974 if I2C - select SND_SOC_WM8978 if I2C - select SND_SOC_WM8983 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8990 if I2C - select SND_SOC_WM8991 if I2C - select SND_SOC_WM8993 if I2C - select SND_SOC_WM8994 if MFD_WM8994 - select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8996 if I2C - select SND_SOC_WM8997 if MFD_WM8997 - select SND_SOC_WM8998 if MFD_WM8998 - select SND_SOC_WM9081 if I2C - select SND_SOC_WM9090 if I2C - select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - select SND_SOC_WSA881X if SOUNDWIRE + imply SND_SOC_88PM860X + imply SND_SOC_L3 + imply SND_SOC_AB8500_CODEC + imply SND_SOC_AC97_CODEC + imply SND_SOC_AD1836 + imply SND_SOC_AD193X_SPI + imply SND_SOC_AD193X_I2C + imply SND_SOC_AD1980 + imply SND_SOC_AD73311 + imply SND_SOC_ADAU1373 + imply SND_SOC_ADAU1761_I2C + imply SND_SOC_ADAU1761_SPI + imply SND_SOC_ADAU1781_I2C + imply SND_SOC_ADAU1781_SPI + imply SND_SOC_ADAV801 + imply SND_SOC_ADAV803 + imply SND_SOC_ADAU1977_SPI + imply SND_SOC_ADAU1977_I2C + imply SND_SOC_ADAU1701 + imply SND_SOC_ADAU7002 + imply SND_SOC_ADAU7118_I2C + imply SND_SOC_ADAU7118_HW + imply SND_SOC_ADS117X + imply SND_SOC_AK4104 + imply SND_SOC_AK4118 + imply SND_SOC_AK4458 + imply SND_SOC_AK4535 + imply SND_SOC_AK4554 + imply SND_SOC_AK4613 + imply SND_SOC_AK4641 + imply SND_SOC_AK4642 + imply SND_SOC_AK4671 + imply SND_SOC_AK5386 + imply SND_SOC_AK5558 + imply SND_SOC_ALC5623 + imply SND_SOC_ALC5632 + imply SND_SOC_BT_SCO + imply SND_SOC_BD28623 + imply SND_SOC_CQ0093VC + imply SND_SOC_CROS_EC_CODEC + imply SND_SOC_CS35L32 + imply SND_SOC_CS35L33 + imply SND_SOC_CS35L34 + imply SND_SOC_CS35L35 + imply SND_SOC_CS35L36 + imply SND_SOC_CS42L42 + imply SND_SOC_CS42L51_I2C + imply SND_SOC_CS42L52 + imply SND_SOC_CS42L56 + imply SND_SOC_CS42L73 + imply SND_SOC_CS4265 + imply SND_SOC_CS4270 + imply SND_SOC_CS4271_I2C + imply SND_SOC_CS4271_SPI + imply SND_SOC_CS42XX8_I2C + imply SND_SOC_CS43130 + imply SND_SOC_CS4341 + imply SND_SOC_CS4349 + imply SND_SOC_CS47L15 + imply SND_SOC_CS47L24 + imply SND_SOC_CS47L35 + imply SND_SOC_CS47L85 + imply SND_SOC_CS47L90 + imply SND_SOC_CS47L92 + imply SND_SOC_CS53L30 + imply SND_SOC_CX20442 + imply SND_SOC_CX2072X + imply SND_SOC_DA7210 + imply SND_SOC_DA7213 + imply SND_SOC_DA7218 + imply SND_SOC_DA7219 + imply SND_SOC_DA732X + imply SND_SOC_DA9055 + imply SND_SOC_DMIC + imply SND_SOC_ES8316 + imply SND_SOC_ES8328_SPI + imply SND_SOC_ES8328_I2C + imply SND_SOC_ES7134 + imply SND_SOC_ES7241 + imply SND_SOC_GTM601 + imply SND_SOC_HDAC_HDMI + imply SND_SOC_HDAC_HDA + imply SND_SOC_ICS43432 + imply SND_SOC_INNO_RK3036 + imply SND_SOC_ISABELLE + imply SND_SOC_JZ4740_CODEC + imply SND_SOC_JZ4725B_CODEC + imply SND_SOC_JZ4770_CODEC + imply SND_SOC_LM4857 + imply SND_SOC_LM49453 + imply SND_SOC_LOCHNAGAR_SC + imply SND_SOC_MAX98088 + imply SND_SOC_MAX98090 + imply SND_SOC_MAX98095 + imply SND_SOC_MAX98357A + imply SND_SOC_MAX98371 + imply SND_SOC_MAX98504 + imply SND_SOC_MAX9867 + imply SND_SOC_MAX98925 + imply SND_SOC_MAX98926 + imply SND_SOC_MAX98927 + imply SND_SOC_MAX98373 + imply SND_SOC_MAX9850 + imply SND_SOC_MAX9860 + imply SND_SOC_MAX9759 + imply SND_SOC_MAX9768 + imply SND_SOC_MAX9877 + imply SND_SOC_MC13783 + imply SND_SOC_ML26124 + imply SND_SOC_MT6351 + imply SND_SOC_MT6358 + imply SND_SOC_MT6660 + imply SND_SOC_NAU8540 + imply SND_SOC_NAU8810 + imply SND_SOC_NAU8822 + imply SND_SOC_NAU8824 + imply SND_SOC_NAU8825 + imply SND_SOC_HDMI_CODEC + imply SND_SOC_PCM1681 + imply SND_SOC_PCM1789_I2C + imply SND_SOC_PCM179X_I2C + imply SND_SOC_PCM179X_SPI + imply SND_SOC_PCM186X_I2C + imply SND_SOC_PCM186X_SPI + imply SND_SOC_PCM3008 + imply SND_SOC_PCM3060_I2C + imply SND_SOC_PCM3060_SPI + imply SND_SOC_PCM3168A_I2C + imply SND_SOC_PCM3168A_SPI + imply SND_SOC_PCM5102A + imply SND_SOC_PCM512x_I2C + imply SND_SOC_PCM512x_SPI + imply SND_SOC_RK3328 + imply SND_SOC_RT274 + imply SND_SOC_RT286 + imply SND_SOC_RT298 + imply SND_SOC_RT1011 + imply SND_SOC_RT1015 + imply SND_SOC_RT1305 + imply SND_SOC_RT1308 + imply SND_SOC_RT5514 + imply SND_SOC_RT5616 + imply SND_SOC_RT5631 + imply SND_SOC_RT5640 + imply SND_SOC_RT5645 + imply SND_SOC_RT5651 + imply SND_SOC_RT5659 + imply SND_SOC_RT5660 + imply SND_SOC_RT5663 + imply SND_SOC_RT5665 + imply SND_SOC_RT5668 + imply SND_SOC_RT5670 + imply SND_SOC_RT5677 + imply SND_SOC_RT5682 + imply SND_SOC_RT700_SDW + imply SND_SOC_RT711_SDW + imply SND_SOC_RT715_SDW + imply SND_SOC_RT1308_SDW + imply SND_SOC_SGTL5000 + imply SND_SOC_SI476X + imply SND_SOC_SIMPLE_AMPLIFIER + imply SND_SOC_SIRF_AUDIO_CODEC + imply SND_SOC_SPDIF + imply SND_SOC_SSM2305 + imply SND_SOC_SSM2518 + imply SND_SOC_SSM2602_SPI + imply SND_SOC_SSM2602_I2C + imply SND_SOC_SSM4567 + imply SND_SOC_STA32X + imply SND_SOC_STA350 + imply SND_SOC_STA529 + imply SND_SOC_STAC9766 + imply SND_SOC_STI_SAS + imply SND_SOC_TAS2552 + imply SND_SOC_TAS2562 + imply SND_SOC_TAS2770 + imply SND_SOC_TAS5086 + imply SND_SOC_TAS571X + imply SND_SOC_TAS5720 + imply SND_SOC_TAS6424 + imply SND_SOC_TDA7419 + imply SND_SOC_TFA9879 + imply SND_SOC_TLV320AIC23_I2C + imply SND_SOC_TLV320AIC23_SPI + imply SND_SOC_TLV320AIC26 + imply SND_SOC_TLV320AIC31XX + imply SND_SOC_TLV320AIC32X4_I2C + imply SND_SOC_TLV320AIC32X4_SPI + imply SND_SOC_TLV320AIC3X + imply SND_SOC_TPA6130A2 + imply SND_SOC_TLV320DAC33 + imply SND_SOC_TSCS42XX + imply SND_SOC_TSCS454 + imply SND_SOC_TS3A227E + imply SND_SOC_TWL4030 + imply SND_SOC_TWL6040 + imply SND_SOC_UDA1334 + imply SND_SOC_UDA134X + imply SND_SOC_UDA1380 + imply SND_SOC_WCD9335 + imply SND_SOC_WCD934X + imply SND_SOC_WL1273 + imply SND_SOC_WM0010 + imply SND_SOC_WM1250_EV1 + imply SND_SOC_WM2000 + imply SND_SOC_WM2200 + imply SND_SOC_WM5100 + imply SND_SOC_WM5102 + imply SND_SOC_WM5110 + imply SND_SOC_WM8350 + imply SND_SOC_WM8400 + imply SND_SOC_WM8510 + imply SND_SOC_WM8523 + imply SND_SOC_WM8524 + imply SND_SOC_WM8580 + imply SND_SOC_WM8711 + imply SND_SOC_WM8727 + imply SND_SOC_WM8728 + imply SND_SOC_WM8731 + imply SND_SOC_WM8737 + imply SND_SOC_WM8741 + imply SND_SOC_WM8750 + imply SND_SOC_WM8753 + imply SND_SOC_WM8770 + imply SND_SOC_WM8776 + imply SND_SOC_WM8782 + imply SND_SOC_WM8804_I2C + imply SND_SOC_WM8804_SPI + imply SND_SOC_WM8900 + imply SND_SOC_WM8903 + imply SND_SOC_WM8904 + imply SND_SOC_WM8940 + imply SND_SOC_WM8955 + imply SND_SOC_WM8960 + imply SND_SOC_WM8961 + imply SND_SOC_WM8962 + imply SND_SOC_WM8971 + imply SND_SOC_WM8974 + imply SND_SOC_WM8978 + imply SND_SOC_WM8983 + imply SND_SOC_WM8985 + imply SND_SOC_WM8988 + imply SND_SOC_WM8990 + imply SND_SOC_WM8991 + imply SND_SOC_WM8993 + imply SND_SOC_WM8994 + imply SND_SOC_WM8995 + imply SND_SOC_WM8996 + imply SND_SOC_WM8997 + imply SND_SOC_WM8998 + imply SND_SOC_WM9081 + imply SND_SOC_WM9090 + imply SND_SOC_WM9705 + imply SND_SOC_WM9712 + imply SND_SOC_WM9713 + imply SND_SOC_WSA881X help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -283,6 +283,7 @@ config SND_SOC_ALL_CODECS config SND_SOC_88PM860X tristate + depends on MFD_88PM860X config SND_SOC_ARIZONA tristate @@ -1301,11 +1302,13 @@ config SND_SOC_TSCS454 Add support for Tempo Semiconductor's TSCS454 audio CODEC. config SND_SOC_TWL4030 - select MFD_TWL4030_AUDIO tristate + depends on TWL4030_CORE + select MFD_TWL4030_AUDIO config SND_SOC_TWL6040 tristate + depends on TWL6040_CORE config SND_SOC_UDA1334 tristate "NXP UDA1334 DAC" @@ -1366,9 +1369,12 @@ config SND_SOC_WM5110 config SND_SOC_WM8350 tristate + depends on MFD_WM8350 config SND_SOC_WM8400 tristate + # FIXME nothing selects SND_SOC_WM8400?? + depends on MFD_WM8400 config SND_SOC_WM8510 tristate "Wolfson Microelectronics WM8510 CODEC" From 2619e03703475b7e0a6f73f85e642859cd25dfc8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2020 16:09:49 +0200 Subject: [PATCH 041/415] ASoC: ti: Add udma-pcm platform driver for UDMA Platform driver glue for platforms using UDMA (am654 and j721e). Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20200210140950.11090-2-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/soc/ti/Kconfig | 4 ++++ sound/soc/ti/Makefile | 2 ++ sound/soc/ti/udma-pcm.c | 43 +++++++++++++++++++++++++++++++++++++++++ sound/soc/ti/udma-pcm.h | 18 +++++++++++++++++ 4 files changed, 67 insertions(+) create mode 100644 sound/soc/ti/udma-pcm.c create mode 100644 sound/soc/ti/udma-pcm.h diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 29f61053ab62e3..bf85a160a5237f 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -10,6 +10,10 @@ config SND_SOC_TI_SDMA_PCM tristate select SND_SOC_GENERIC_DMAENGINE_PCM +config SND_SOC_TI_UDMA_PCM + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + comment "Texas Instruments DAI support for:" config SND_SOC_DAVINCI_ASP tristate "daVinci Audio Serial Port (ASP) or McBSP support" diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile index 08c44d56ef3ecf..ea48c6679cc72f 100644 --- a/sound/soc/ti/Makefile +++ b/sound/soc/ti/Makefile @@ -3,9 +3,11 @@ # Platform drivers snd-soc-ti-edma-objs := edma-pcm.o snd-soc-ti-sdma-objs := sdma-pcm.o +snd-soc-ti-udma-objs := udma-pcm.o obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o +obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o # CPU DAI drivers snd-soc-davinci-asp-objs := davinci-i2s.o diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c new file mode 100644 index 00000000000000..39830caaaf7cbd --- /dev/null +++ b/sound/soc/ti/udma-pcm.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com + * Author: Peter Ujfalusi + */ + +#include +#include +#include +#include +#include +#include + +#include "udma-pcm.h" + +static const struct snd_pcm_hardware udma_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | + SNDRV_PCM_INFO_INTERLEAVED, + .buffer_bytes_max = SIZE_MAX, + .period_bytes_min = 32, + .period_bytes_max = SZ_64K, + .periods_min = 2, + .periods_max = UINT_MAX, +}; + +static const struct snd_dmaengine_pcm_config udma_dmaengine_pcm_config = { + .pcm_hardware = &udma_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, +}; + +int udma_pcm_platform_register(struct device *dev) +{ + return devm_snd_dmaengine_pcm_register(dev, &udma_dmaengine_pcm_config, + 0); +} +EXPORT_SYMBOL_GPL(udma_pcm_platform_register); + +MODULE_AUTHOR("Peter Ujfalusi "); +MODULE_DESCRIPTION("UDMA PCM ASoC platform driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h new file mode 100644 index 00000000000000..54111e7312c106 --- /dev/null +++ b/sound/soc/ti/udma-pcm.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com + */ + +#ifndef __UDMA_PCM_H__ +#define __UDMA_PCM_H__ + +#if IS_ENABLED(CONFIG_SND_SOC_TI_UDMA_PCM) +int udma_pcm_platform_register(struct device *dev); +#else +static inline int udma_pcm_platform_register(struct device *dev) +{ + return 0; +} +#endif /* CONFIG_SND_SOC_TI_UDMA_PCM */ + +#endif /* __UDMA_PCM_H__ */ From fb0c3c6e2007da156d023e91da42c173ea33b102 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2020 16:09:50 +0200 Subject: [PATCH 042/415] ASoC: ti: davinci-mcasp: Add support for platforms using UDMA k3 devices including am654 and j721e are using UDMA Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20200210140950.11090-3-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/soc/ti/Kconfig | 4 +++- sound/soc/ti/davinci-mcasp.c | 7 +++++++ 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index bf85a160a5237f..c5408c129f342d 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only menu "Audio support for Texas Instruments SoCs" -depends on DMA_OMAP || TI_EDMA || COMPILE_TEST +depends on DMA_OMAP || TI_EDMA || TI_K3_UDMA || COMPILE_TEST config SND_SOC_TI_EDMA_PCM tristate @@ -28,6 +28,7 @@ config SND_SOC_DAVINCI_MCASP tristate "Multichannel Audio Serial Port (McASP) support" select SND_SOC_TI_EDMA_PCM select SND_SOC_TI_SDMA_PCM + select SND_SOC_TI_UDMA_PCM help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: @@ -35,6 +36,7 @@ config SND_SOC_DAVINCI_MCASP - Sitara line of SoCs (AM335x, AM438x, etc) - DRA7x devices - Keystone devices + - K3 devices (am654, j721e) config SND_SOC_DAVINCI_VCIF tristate "daVinci Voice Interface (VCIF) support" diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index e1e937eb1dc1ac..d9c3a3210a2405 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -38,6 +38,7 @@ #include "edma-pcm.h" #include "sdma-pcm.h" +#include "udma-pcm.h" #include "davinci-mcasp.h" #define MCASP_MAX_AFIFO_DEPTH 64 @@ -1875,6 +1876,7 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( enum { PCM_EDMA, PCM_SDMA, + PCM_UDMA, }; static const char *sdma_prefix = "ti,omap"; @@ -1912,6 +1914,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp); if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix))) return PCM_SDMA; + else if (strstr(tmp, "udmap")) + return PCM_UDMA; return PCM_EDMA; } @@ -2371,6 +2375,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) case PCM_SDMA: ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx"); break; + case PCM_UDMA: + ret = udma_pcm_platform_register(&pdev->dev); + break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); case -EPROBE_DEFER: From c8b60c6d93b8104f5a8d9fbb4f52ad88df918a44 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Thu, 6 Feb 2020 11:17:52 +0800 Subject: [PATCH 043/415] ASoC: mediatek: mt8173-rt5650: support HDMI jack reporting Uses hdmi-codec to support HDMI jack reporting. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200206102509.3.I253f51edff62df1d88005de12ba601aa029b1e99@changeid Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ef6f2367528679..849b050a54d149 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -11,6 +11,7 @@ #include #include #include +#include #include "../../codecs/rt5645.h" #define MCLK_FOR_CODECS 12288000 @@ -98,7 +99,7 @@ static const struct snd_soc_ops mt8173_rt5650_ops = { .hw_params = mt8173_rt5650_hw_params, }; -static struct snd_soc_jack mt8173_rt5650_jack; +static struct snd_soc_jack mt8173_rt5650_jack, mt8173_rt5650_hdmi_jack; static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { @@ -144,6 +145,19 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) &mt8173_rt5650_jack); } +static int mt8173_rt5650_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + + ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, + &mt8173_rt5650_hdmi_jack, NULL, 0); + if (ret) + return ret; + + return hdmi_codec_set_jack_detect(rtd->codec_dai->component, + &mt8173_rt5650_hdmi_jack); +} + enum { DAI_LINK_PLAYBACK, DAI_LINK_CAPTURE, @@ -222,6 +236,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .name = "HDMI BE", .no_pcm = 1, .dpcm_playback = 1, + .init = mt8173_rt5650_hdmi_init, SND_SOC_DAILINK_REG(hdmi_be), }, }; From da22a95313197a349c557b98e3bee4e2b04d4f9d Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Mon, 10 Feb 2020 23:04:21 +0800 Subject: [PATCH 044/415] ASoC: wcd934x: Remove set but not unused variable 'hph_comp_ctrl7' sound/soc/codecs/wcd934x.c: In function wcd934x_codec_hphdelay_lutbypass: sound/soc/codecs/wcd934x.c:3395:6: warning: variable hph_comp_ctrl7 set but not used [-Wunused-but-set-variable] commit da3e83f8bb86 ("ASoC: wcd934x: add audio routings") involved this unused variable. Reported-by: Hulk Robot Signed-off-by: YueHaibing Link: https://lore.kernel.org/r/20200210150421.34680-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index e780ecd554d265..aefaadfba8a106 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -3388,18 +3388,15 @@ static void wcd934x_codec_hphdelay_lutbypass(struct snd_soc_component *comp, { u8 hph_dly_mask; u16 hph_lut_bypass_reg = 0; - u16 hph_comp_ctrl7 = 0; switch (interp_idx) { case INTERP_HPHL: hph_dly_mask = 1; hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHL_COMP_LUT; - hph_comp_ctrl7 = WCD934X_CDC_COMPANDER1_CTL7; break; case INTERP_HPHR: hph_dly_mask = 2; hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHR_COMP_LUT; - hph_comp_ctrl7 = WCD934X_CDC_COMPANDER2_CTL7; break; default: return; From f4d95de415b286090c1bf739c20a5ea2aefda834 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 10 Feb 2020 09:24:22 +0000 Subject: [PATCH 045/415] ASoC: ti: davinci-mcasp: remove redundant assignment to variable ret The assignment to ret is redundant as it is not used in the error return path and hence can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King Acked-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20200210092423.327499-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-mcasp.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index d9c3a3210a2405..734ffe925c4dbf 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1765,10 +1765,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( } else if (match) { pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata), GFP_KERNEL); - if (!pdata) { - ret = -ENOMEM; - return pdata; - } + if (!pdata) + return NULL; } else { /* control shouldn't reach here. something is wrong */ ret = -EINVAL; From f9eb06cd0cdd50a3125bc9c62cdc997dc461eae7 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Thu, 6 Feb 2020 11:17:50 +0800 Subject: [PATCH 046/415] drm/mediatek: exit earlier if failed to register audio driver Exits earlier if register_audio_driver() returns errors. Signed-off-by: Tzung-Bi Shih Acked-by: CK Hu Link: https://lore.kernel.org/r/20200206102509.1.Ieba8d422486264eb7aaa3aa257620a1b0c74c6db@changeid Signed-off-by: Mark Brown --- drivers/gpu/drm/mediatek/mtk_hdmi.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index 5e4a4dbda443f9..fcec06e63e0cb9 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -1659,7 +1659,7 @@ static const struct hdmi_codec_ops mtk_hdmi_audio_codec_ops = { .get_eld = mtk_hdmi_audio_get_eld, }; -static void mtk_hdmi_register_audio_driver(struct device *dev) +static int mtk_hdmi_register_audio_driver(struct device *dev) { struct hdmi_codec_pdata codec_data = { .ops = &mtk_hdmi_audio_codec_ops, @@ -1672,9 +1672,10 @@ static void mtk_hdmi_register_audio_driver(struct device *dev) PLATFORM_DEVID_AUTO, &codec_data, sizeof(codec_data)); if (IS_ERR(pdev)) - return; + return PTR_ERR(pdev); DRM_INFO("%s driver bound to HDMI\n", HDMI_CODEC_DRV_NAME); + return 0; } static int mtk_drm_hdmi_probe(struct platform_device *pdev) @@ -1708,7 +1709,11 @@ static int mtk_drm_hdmi_probe(struct platform_device *pdev) return ret; } - mtk_hdmi_register_audio_driver(dev); + ret = mtk_hdmi_register_audio_driver(dev); + if (ret) { + dev_err(dev, "Failed to register audio driver: %d\n", ret); + return ret; + } hdmi->bridge.funcs = &mtk_hdmi_bridge_funcs; hdmi->bridge.of_node = pdev->dev.of_node; From 5d3c644773925c3568617435e42a9404a114c428 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Thu, 6 Feb 2020 11:17:51 +0800 Subject: [PATCH 047/415] drm/mediatek: support HDMI jack status reporting 1. Provides a callback (i.e. mtk_hdmi_audio_hook_plugged_cb) to hdmi-codec. When ASoC machine driver calls hdmi_codec_set_jack_detect(), the callback will be invoked to save plugged_cb and codec_dev parameters. +---------+ set_jack_ +------------+ plugged_cb +----------+ | machine | ----------> | hdmi-codec | ----------> | mtk-hdmi | +---------+ detect() +------------+ codec_dev +----------+ 2. When there is any jack status changes, mtk-hdmi will call the plugged_cb() to notify hdmi-codec. And then hdmi-codec will call snd_soc_jack_report(). +----------+ plugged_cb +------------+ | mtk-hdmi | ----------> | hdmi-codec | -> snd_soc_jack_report() +----------+ codec_dev +------------+ connector_status Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200206102509.2.I230fd59de28e73934a91cb01424e25b9e84727f4@changeid Signed-off-by: Mark Brown --- drivers/gpu/drm/mediatek/mtk_hdmi.c | 34 ++++++++++++++++++++++++++--- 1 file changed, 31 insertions(+), 3 deletions(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index fcec06e63e0cb9..03aeb73005ef88 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -169,6 +169,8 @@ struct mtk_hdmi { bool audio_enable; bool powered; bool enabled; + hdmi_codec_plugged_cb plugged_cb; + struct device *codec_dev; }; static inline struct mtk_hdmi *hdmi_ctx_from_bridge(struct drm_bridge *b) @@ -1194,13 +1196,23 @@ static void mtk_hdmi_clk_disable_audio(struct mtk_hdmi *hdmi) clk_disable_unprepare(hdmi->clk[MTK_HDMI_CLK_AUD_SPDIF]); } +static enum drm_connector_status +mtk_hdmi_update_plugged_status(struct mtk_hdmi *hdmi) +{ + bool connected = mtk_cec_hpd_high(hdmi->cec_dev); + + if (hdmi->plugged_cb && hdmi->codec_dev) + hdmi->plugged_cb(hdmi->codec_dev, connected); + + return connected ? + connector_status_connected : connector_status_disconnected; +} + static enum drm_connector_status hdmi_conn_detect(struct drm_connector *conn, bool force) { struct mtk_hdmi *hdmi = hdmi_ctx_from_conn(conn); - - return mtk_cec_hpd_high(hdmi->cec_dev) ? - connector_status_connected : connector_status_disconnected; + return mtk_hdmi_update_plugged_status(hdmi); } static void hdmi_conn_destroy(struct drm_connector *conn) @@ -1651,20 +1663,36 @@ static int mtk_hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, return 0; } +static int mtk_hdmi_audio_hook_plugged_cb(struct device *dev, void *data, + hdmi_codec_plugged_cb fn, + struct device *codec_dev) +{ + struct mtk_hdmi *hdmi = data; + + hdmi->plugged_cb = fn; + hdmi->codec_dev = codec_dev; + mtk_hdmi_update_plugged_status(hdmi); + + return 0; +} + static const struct hdmi_codec_ops mtk_hdmi_audio_codec_ops = { .hw_params = mtk_hdmi_audio_hw_params, .audio_startup = mtk_hdmi_audio_startup, .audio_shutdown = mtk_hdmi_audio_shutdown, .digital_mute = mtk_hdmi_audio_digital_mute, .get_eld = mtk_hdmi_audio_get_eld, + .hook_plugged_cb = mtk_hdmi_audio_hook_plugged_cb, }; static int mtk_hdmi_register_audio_driver(struct device *dev) { + struct mtk_hdmi *hdmi = dev_get_drvdata(dev); struct hdmi_codec_pdata codec_data = { .ops = &mtk_hdmi_audio_codec_ops, .max_i2s_channels = 2, .i2s = 1, + .data = hdmi, }; struct platform_device *pdev; From 3f06501ea4d2d8add203e66d225274f106cb4029 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:18 -0600 Subject: [PATCH 048/415] ASoC: SOF: Do not reset hw_params for streams that ignored suspend Setting the prepared flag to false marks the streams for the hw_params to be reset upon resuming. In the case of the D0i3-compatible streams that ignored suspend to keep the pipeline active in the DSP during suspend, this should not be done. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0d8f65b9ae25c8..345e42ee478372 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -39,6 +39,13 @@ int sof_set_hw_params_upon_resume(struct device *dev) */ list_for_each_entry(spcm, &sdev->pcm_list, list) { for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + /* + * do not reset hw_params upon resume for streams that + * were kept running during suspend + */ + if (spcm->stream[dir].suspend_ignored) + continue; + substream = spcm->stream[dir].substream; if (!substream || !substream->runtime) continue; From fb9a81192d44ae9f334b1d88651dec427fa751d8 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:19 -0600 Subject: [PATCH 049/415] ASoC: SOF: pm: Unify suspend/resume routines Unify the suspend/resume routines for both the D0I3/D3 DSP targets in sof_suspend()/sof_resume(). Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 91 ++++++++++++++++++++-------------------------- 1 file changed, 39 insertions(+), 52 deletions(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index a0cde053b61aac..5b186bceedb910 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -50,6 +50,7 @@ static void sof_cache_debugfs(struct snd_sof_dev *sdev) static int sof_resume(struct device *dev, bool runtime_resume) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + enum sof_d0_substate old_d0_substate = sdev->d0_substate; int ret; /* do nothing if dsp resume callbacks are not set */ @@ -60,6 +61,17 @@ static int sof_resume(struct device *dev, bool runtime_resume) if (sdev->first_boot) return 0; + /* resume from D0I3 */ + if (!runtime_resume && old_d0_substate == SOF_DSP_D0I3) { + ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I0); + if (ret < 0 && ret != -ENOTSUPP) { + dev_err(sdev->dev, + "error: failed to resume from D0I3 %d\n", + ret); + return ret; + } + } + /* * if the runtime_resume flag is set, call the runtime_resume routine * or else call the system resume routine @@ -74,6 +86,10 @@ static int sof_resume(struct device *dev, bool runtime_resume) return ret; } + /* Nothing further to do if resuming from D0I3 */ + if (!runtime_resume && old_d0_substate == SOF_DSP_D0I3) + return 0; + sdev->fw_state = SOF_FW_BOOT_PREPARE; /* load the firmware */ @@ -140,10 +156,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) return 0; if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) - goto power_down; - - /* release trace */ - snd_sof_release_trace(sdev); + goto suspend; /* set restore_stream for all streams during system suspend */ if (!runtime_suspend) { @@ -156,6 +169,22 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) } } + if (snd_sof_dsp_d0i3_on_suspend(sdev)) { + /* suspend to D0i3 */ + ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I3); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to enter D0I3, %d\n", + ret); + return ret; + } + + /* Skip to platform-specific suspend if DSP is entering D0I3 */ + goto suspend; + } + + /* release trace */ + snd_sof_release_trace(sdev); + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) /* cache debugfs contents during runtime suspend */ if (runtime_suspend) @@ -179,13 +208,13 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) ret); } -power_down: +suspend: /* return if the DSP was not probed successfully */ if (sdev->fw_state == SOF_FW_BOOT_NOT_STARTED) return 0; - /* power down all DSP cores */ + /* platform-specific suspend */ if (runtime_suspend) ret = snd_sof_dsp_runtime_suspend(sdev); else @@ -195,6 +224,10 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) "error: failed to power down DSP during suspend %d\n", ret); + /* Do not reset FW state if DSP is in D0I3 */ + if (sdev->d0_substate == SOF_DSP_D0I3) + return ret; + /* reset FW state */ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; @@ -275,58 +308,12 @@ EXPORT_SYMBOL(snd_sof_set_d0_substate); int snd_sof_resume(struct device *dev) { - struct snd_sof_dev *sdev = dev_get_drvdata(dev); - int ret; - - if (snd_sof_dsp_d0i3_on_suspend(sdev)) { - /* resume from D0I3 */ - dev_dbg(sdev->dev, "DSP will exit from D0i3...\n"); - ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I0); - if (ret == -ENOTSUPP) { - /* fallback to resume from D3 */ - dev_dbg(sdev->dev, "D0i3 not supported, fall back to resume from D3...\n"); - goto d3_resume; - } else if (ret < 0) { - dev_err(sdev->dev, "error: failed to exit from D0I3 %d\n", - ret); - return ret; - } - - /* platform-specific resume from D0i3 */ - return snd_sof_dsp_resume(sdev); - } - -d3_resume: - /* resume from D3 */ return sof_resume(dev, false); } EXPORT_SYMBOL(snd_sof_resume); int snd_sof_suspend(struct device *dev) { - struct snd_sof_dev *sdev = dev_get_drvdata(dev); - int ret; - - if (snd_sof_dsp_d0i3_on_suspend(sdev)) { - /* suspend to D0i3 */ - dev_dbg(sdev->dev, "DSP is trying to enter D0i3...\n"); - ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I3); - if (ret == -ENOTSUPP) { - /* fallback to D3 suspend */ - dev_dbg(sdev->dev, "D0i3 not supported, fall back to D3...\n"); - goto d3_suspend; - } else if (ret < 0) { - dev_err(sdev->dev, "error: failed to enter D0I3, %d\n", - ret); - return ret; - } - - /* platform-specific suspend to D0i3 */ - return snd_sof_dsp_suspend(sdev); - } - -d3_suspend: - /* suspend to D3 */ return sof_suspend(dev, false); } EXPORT_SYMBOL(snd_sof_suspend); From 043ae13bbd558971ce91596ce09c03d6ef6a4a0c Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:20 -0600 Subject: [PATCH 050/415] ASoC: SOF: Add system_suspend_target field to struct snd_sof_dev Add the system_suspend_target field to struct snd_sof_dev to track the intended system suspend power target. This will be used as one of the criteria for determining the final DSP power state. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 4 ++-- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/pm.c | 9 ++++++--- sound/soc/sof/sof-priv.h | 12 ++++++++++-- 4 files changed, 19 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 4a4d318f97ffa5..fddf2c48904f02 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -481,7 +481,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct pci_dev *pci = to_pci_dev(sdev->dev); - if (sdev->s0_suspend) { + if (sdev->system_suspend_target == SOF_SUSPEND_S0IX) { /* restore L1SEN bit */ if (hda->l1_support_changed) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -530,7 +530,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev) struct pci_dev *pci = to_pci_dev(sdev->dev); int ret; - if (sdev->s0_suspend) { + if (sdev->system_suspend_target == SOF_SUSPEND_S0IX) { /* enable L1SEN to make sure the system can enter S0Ix */ hda->l1_support_changed = snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 29435ba2d32927..db3df02c7398f1 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -372,7 +372,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component, stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START; break; case SNDRV_PCM_TRIGGER_SUSPEND: - if (sdev->s0_suspend && + if (sdev->system_suspend_target == SOF_SUSPEND_S0IX && spcm->stream[substream->stream].d0i3_compatible) { /* * trap the event, not sending trigger stop to diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 5b186bceedb910..c86ac1e84bd7e9 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -323,10 +323,13 @@ int snd_sof_prepare(struct device *dev) struct snd_sof_dev *sdev = dev_get_drvdata(dev); #if defined(CONFIG_ACPI) - sdev->s0_suspend = acpi_target_system_state() == ACPI_STATE_S0; + if (acpi_target_system_state() == ACPI_STATE_S0) + sdev->system_suspend_target = SOF_SUSPEND_S0IX; + else + sdev->system_suspend_target = SOF_SUSPEND_S3; #else /* will suspend to S3 by default */ - sdev->s0_suspend = false; + sdev->system_suspend_target = SOF_SUSPEND_S3; #endif return 0; @@ -337,6 +340,6 @@ void snd_sof_complete(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); - sdev->s0_suspend = false; + sdev->system_suspend_target = SOF_SUSPEND_NONE; } EXPORT_SYMBOL(snd_sof_complete); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index bc2337cf1142d5..1839cc51957d19 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -60,6 +60,13 @@ enum sof_d0_substate { SOF_DSP_D0I3, /* DSP D0i3(low power) substate*/ }; +/* System suspend target state */ +enum sof_system_suspend_state { + SOF_SUSPEND_NONE = 0, + SOF_SUSPEND_S0IX, + SOF_SUSPEND_S3, +}; + struct snd_sof_dev; struct snd_sof_ipc_msg; struct snd_sof_ipc; @@ -325,8 +332,9 @@ struct snd_sof_dev { /* power states related */ enum sof_d0_substate d0_substate; - /* flag to track if the intended power target of suspend is S0ix */ - bool s0_suspend; + + /* Intended power target of system suspend */ + enum sof_system_suspend_state system_suspend_target; /* DSP firmware boot */ wait_queue_head_t boot_wait; From 700d167739a099cdf12ed15c25fec7f4cb563d42 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:21 -0600 Subject: [PATCH 051/415] ASoC: SOF: pm: Introduce DSP power states Add a new enum sof_dsp_power_states for all the possible the DSP device states. The SOF driver currently handles only the D0 and D3 states and support for other states will be added later as needed. Also, add a helper to determine the target DSP power state based on the system suspend target. The snd_sof_dsp_d0i3_on_suspend() function is renamed to snd_sof_stream_suspend_ignored() to be more indicative of what it does and it used to determine the target DSP state during system suspend. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 38 +++++++++++++++++++++++++++++++++++++- sound/soc/sof/sof-audio.c | 2 +- sound/soc/sof/sof-audio.h | 2 +- sound/soc/sof/sof-priv.h | 10 ++++++++++ 4 files changed, 49 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index c86ac1e84bd7e9..bec25cb6beeccf 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -12,6 +12,42 @@ #include "sof-priv.h" #include "sof-audio.h" +/* + * Helper function to determine the target DSP state during + * system suspend. This function only cares about the device + * D-states. Platform-specific substates, if any, should be + * handled by the platform-specific parts. + */ +static u32 snd_sof_dsp_power_target(struct snd_sof_dev *sdev) +{ + u32 target_dsp_state; + + switch (sdev->system_suspend_target) { + case SOF_SUSPEND_S3: + /* DSP should be in D3 if the system is suspending to S3 */ + target_dsp_state = SOF_DSP_PM_D3; + break; + case SOF_SUSPEND_S0IX: + /* + * Currently, the only criterion for retaining the DSP in D0 + * is that there are streams that ignored the suspend trigger. + * Additional criteria such Soundwire clock-stop mode and + * device suspend latency considerations will be added later. + */ + if (snd_sof_stream_suspend_ignored(sdev)) + target_dsp_state = SOF_DSP_PM_D0; + else + target_dsp_state = SOF_DSP_PM_D3; + break; + default: + /* This case would be during runtime suspend */ + target_dsp_state = SOF_DSP_PM_D3; + break; + } + + return target_dsp_state; +} + static int sof_send_pm_ctx_ipc(struct snd_sof_dev *sdev, int cmd) { struct sof_ipc_pm_ctx pm_ctx; @@ -169,7 +205,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) } } - if (snd_sof_dsp_d0i3_on_suspend(sdev)) { + if (snd_sof_dsp_power_target(sdev) == SOF_DSP_PM_D0) { /* suspend to D0i3 */ ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I3); if (ret < 0) { diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 345e42ee478372..d16571ca129cec 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -11,7 +11,7 @@ #include "sof-audio.h" #include "ops.h" -bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev) +bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev) { struct snd_sof_pcm *spcm; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index a62fb2da6a6ece..a2702afbd9a121 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -202,7 +202,7 @@ int snd_sof_ipc_set_get_comp_data(struct snd_sof_control *scontrol, /* PM */ int sof_restore_pipelines(struct device *dev); int sof_set_hw_params_upon_resume(struct device *dev); -bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev); +bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev); /* Machine driver enumeration */ int sof_machine_register(struct snd_sof_dev *sdev, void *pdata); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 1839cc51957d19..a7c6109acd98f3 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -54,6 +54,16 @@ extern int sof_core_debug; (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) || \ IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)) +/* DSP power state */ +enum sof_dsp_power_states { + SOF_DSP_PM_D0, + SOF_DSP_PM_D1, + SOF_DSP_PM_D2, + SOF_DSP_PM_D3_HOT, + SOF_DSP_PM_D3, + SOF_DSP_PM_D3_COLD, +}; + /* DSP D0ix sub-state */ enum sof_d0_substate { SOF_DSP_D0I0 = 0, /* DSP default D0 substate */ From 61e285caf40fef18e8bd7cea5237ee6723609a1c Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:22 -0600 Subject: [PATCH 052/415] ASoC: SOF: Move DSP power state transitions to platform-specific ops The DSP device substates such as D0I0/D0I3 are platform-specific. Therefore, the d0_substate field of struct snd_sof_dev is replaced with the dsp_power_state field which represents the current state of the DSP. This field holds both the device state and the platform-specific substate values. With the DSP device substates being platform-specific, the DSP power state transitions need to be performed in the platform-specific suspend/resume ops as well. In order to achieve this, the ops signature has to be modified to pass the target device state as an argument. The target substate will be determined by the platform-specific ops before performing the transition. For example, in the case of the system suspending to S0IX, the top-level SOF device suspend callback needs to only determine if the DSP will be entering D3 or remain in D0. The target substate in case the device needs to remain in D0 (D0I0 or D0I3) will be determined by the platform-specific suspend op. With the addition of the extended set of power states for the DSP, the set_power_state op for HDA platforms has to be extended to handle only the appropriate state transitions. So, the implementation for the Intel HDA platforms is also modified. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 4 +- sound/soc/sof/intel/hda-dsp.c | 223 +++++++++++++++++++++++++++++++--- sound/soc/sof/intel/hda.h | 10 +- sound/soc/sof/ops.h | 16 +-- sound/soc/sof/pm.c | 92 ++------------ sound/soc/sof/sof-priv.h | 18 ++- 6 files changed, 242 insertions(+), 121 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 34cefbaf2d2afc..1d07450aff7774 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -286,8 +286,8 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) /* initialize sof device */ sdev->dev = dev; - /* initialize default D0 sub-state */ - sdev->d0_substate = SOF_DSP_D0I0; + /* initialize default DSP power state */ + sdev->dsp_power_state.state = SOF_DSP_PM_D0; sdev->pdata = plat_data; sdev->first_boot = true; diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index fddf2c48904f02..8c00e128a7b047 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -338,13 +338,10 @@ static int hda_dsp_send_pm_gate_ipc(struct snd_sof_dev *sdev, u32 flags) sizeof(pm_gate), &reply, sizeof(reply)); } -int hda_dsp_set_power_state(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate) +static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) { struct hdac_bus *bus = sof_to_bus(sdev); - u32 flags; int ret; - u8 value; /* Write to D0I3C after Command-In-Progress bit is cleared */ ret = hda_dsp_wait_d0i3c_done(sdev); @@ -354,7 +351,6 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev, } /* Update D0I3C register */ - value = d0_substate == SOF_DSP_D0I3 ? SOF_HDA_VS_D0I3C_I3 : 0; snd_hdac_chip_updateb(bus, VS_D0I3C, SOF_HDA_VS_D0I3C_I3, value); /* Wait for cmd in progress to be cleared before exiting the function */ @@ -367,20 +363,160 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev, dev_vdbg(bus->dev, "D0I3C updated, register = 0x%x\n", snd_hdac_chip_readb(bus, VS_D0I3C)); - if (d0_substate == SOF_DSP_D0I0) - flags = HDA_PM_PPG;/* prevent power gating in D0 */ - else - flags = HDA_PM_NO_DMA_TRACE;/* disable DMA trace in D0I3*/ + return 0; +} - /* sending pm_gate IPC */ - ret = hda_dsp_send_pm_gate_ipc(sdev, flags); +static int hda_dsp_set_D0_state(struct snd_sof_dev *sdev, + const struct sof_dsp_power_state *target_state) +{ + u32 flags = 0; + int ret; + u8 value = 0; + + /* + * Sanity check for illegal state transitions + * The only allowed transitions are: + * 1. D3 -> D0I0 + * 2. D0I0 -> D0I3 + * 3. D0I3 -> D0I0 + */ + switch (sdev->dsp_power_state.state) { + case SOF_DSP_PM_D0: + /* Follow the sequence below for D0 substate transitions */ + break; + case SOF_DSP_PM_D3: + /* Follow regular flow for D3 -> D0 transition */ + return 0; + default: + dev_err(sdev->dev, "error: transition from %d to %d not allowed\n", + sdev->dsp_power_state.state, target_state->state); + return -EINVAL; + } + + /* Set flags and register value for D0 target substate */ + if (target_state->substate == SOF_HDA_DSP_PM_D0I3) { + value = SOF_HDA_VS_D0I3C_I3; + + /* disable DMA trace in D0I3 */ + flags = HDA_PM_NO_DMA_TRACE; + } else { + /* prevent power gating in D0I0 */ + flags = HDA_PM_PPG; + } + + /* update D0I3C register */ + ret = hda_dsp_update_d0i3c_register(sdev, value); if (ret < 0) + return ret; + + /* + * Notify the DSP of the state change. + * If this IPC fails, revert the D0I3C register update in order + * to prevent partial state change. + */ + ret = hda_dsp_send_pm_gate_ipc(sdev, flags); + if (ret < 0) { dev_err(sdev->dev, "error: PM_GATE ipc error %d\n", ret); + goto revert; + } + + return ret; + +revert: + /* fallback to the previous register value */ + value = value ? 0 : SOF_HDA_VS_D0I3C_I3; + + /* + * This can fail but return the IPC error to signal that + * the state change failed. + */ + hda_dsp_update_d0i3c_register(sdev, value); return ret; } +/* + * All DSP power state transitions are initiated by the driver. + * If the requested state change fails, the error is simply returned. + * Further state transitions are attempted only when the set_power_save() op + * is called again either because of a new IPC sent to the DSP or + * during system suspend/resume. + */ +int hda_dsp_set_power_state(struct snd_sof_dev *sdev, + const struct sof_dsp_power_state *target_state) +{ + int ret = 0; + + /* Nothing to do if the DSP is already in the requested state */ + if (target_state->state == sdev->dsp_power_state.state && + target_state->substate == sdev->dsp_power_state.substate) + return 0; + + switch (target_state->state) { + case SOF_DSP_PM_D0: + ret = hda_dsp_set_D0_state(sdev, target_state); + break; + case SOF_DSP_PM_D3: + /* The only allowed transition is: D0I0 -> D3 */ + if (sdev->dsp_power_state.state == SOF_DSP_PM_D0 && + sdev->dsp_power_state.substate == SOF_HDA_DSP_PM_D0I0) + break; + + dev_err(sdev->dev, + "error: transition from %d to %d not allowed\n", + sdev->dsp_power_state.state, target_state->state); + return -EINVAL; + default: + dev_err(sdev->dev, "error: target state unsupported %d\n", + target_state->state); + return -EINVAL; + } + if (ret < 0) { + dev_err(sdev->dev, + "failed to set requested target DSP state %d substate %d\n", + target_state->state, target_state->substate); + return ret; + } + + sdev->dsp_power_state = *target_state; + dev_dbg(sdev->dev, "New DSP state %d substate %d\n", + target_state->state, target_state->substate); + return ret; +} + +/* + * Audio DSP states may transform as below:- + * + * D0I3 compatible stream + * Runtime +---------------------+ opened only, timeout + * suspend | +--------------------+ + * +------------+ D0(active) | | + * | | <---------------+ | + * | +--------> | | | + * | |Runtime +--^--+---------^--+--+ The last | | + * | |resume | | | | opened D0I3 | | + * | | | | | | compatible | | + * | | resume| | | | stream closed | | + * | | from | | D3 | | | | + * | | D3 | |suspend | | d0i3 | | + * | | | | | |suspend | | + * | | | | | | | | + * | | | | | | | | + * +-v---+-----------+--v-------+ | | +------+----v----+ + * | | | +-----------> | + * | D3 (suspended) | | | D0I3 +-----+ + * | | +--------------+ | | + * | | resume from | | | + * +-------------------^--------+ d0i3 suspend +----------------+ | + * | | + * | D3 suspend | + * +------------------------------------------------+ + * + * d0i3_suspend = s0_suspend && D0I3 stream opened, + * D3 suspend = !d0i3_suspend, + */ + static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; @@ -480,8 +616,22 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct pci_dev *pci = to_pci_dev(sdev->dev); + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + .substate = SOF_HDA_DSP_PM_D0I0, + }; + int ret; + + /* resume from D0I3 */ + if (sdev->dsp_power_state.state == SOF_DSP_PM_D0) { + /* Set DSP power state */ + ret = hda_dsp_set_power_state(sdev, &target_state); + if (ret < 0) { + dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", + target_state.state, target_state.substate); + return ret; + } - if (sdev->system_suspend_target == SOF_SUSPEND_S0IX) { /* restore L1SEN bit */ if (hda->l1_support_changed) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -495,13 +645,27 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) } /* init hda controller. DSP cores will be powered up during fw boot */ - return hda_resume(sdev, false); + ret = hda_resume(sdev, false); + if (ret < 0) + return ret; + + hda_dsp_set_power_state(sdev, &target_state); + return ret; } int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) { + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + }; + int ret; + /* init hda controller. DSP cores will be powered up during fw boot */ - return hda_resume(sdev, true); + ret = hda_resume(sdev, true); + if (ret < 0) + return ret; + + return hda_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) @@ -519,18 +683,41 @@ int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) { + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D3, + }; + int ret; + /* stop hda controller and power dsp off */ - return hda_suspend(sdev, true); + ret = hda_suspend(sdev, true); + if (ret < 0) + return ret; + + return hda_dsp_set_power_state(sdev, &target_state); } -int hda_dsp_suspend(struct snd_sof_dev *sdev) +int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct hdac_bus *bus = sof_to_bus(sdev); struct pci_dev *pci = to_pci_dev(sdev->dev); + const struct sof_dsp_power_state target_dsp_state = { + .state = target_state, + .substate = target_state == SOF_DSP_PM_D0 ? + SOF_HDA_DSP_PM_D0I3 : 0, + }; int ret; - if (sdev->system_suspend_target == SOF_SUSPEND_S0IX) { + if (target_state == SOF_DSP_PM_D0) { + /* Set DSP power state */ + ret = hda_dsp_set_power_state(sdev, &target_dsp_state); + if (ret < 0) { + dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", + target_dsp_state.state, + target_dsp_state.substate); + return ret; + } + /* enable L1SEN to make sure the system can enter S0Ix */ hda->l1_support_changed = snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -551,7 +738,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev) return ret; } - return 0; + return hda_dsp_set_power_state(sdev, &target_dsp_state); } int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 6191d9192faee2..02c2a7eadb1bc6 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -392,6 +392,12 @@ struct sof_intel_dsp_bdl { #define SOF_HDA_PLAYBACK 0 #define SOF_HDA_CAPTURE 1 +/* HDA DSP D0 substate */ +enum sof_hda_D0_substate { + SOF_HDA_DSP_PM_D0I0, /* default D0 substate */ + SOF_HDA_DSP_PM_D0I3, /* low power D0 substate */ +}; + /* represents DSP HDA controller frontend - i.e. host facing control */ struct sof_intel_hda_dev { @@ -469,9 +475,9 @@ void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev); void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev); int hda_dsp_set_power_state(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate); + const struct sof_dsp_power_state *target_state); -int hda_dsp_suspend(struct snd_sof_dev *sdev); +int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state); int hda_dsp_resume(struct snd_sof_dev *sdev); int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev); int hda_dsp_runtime_resume(struct snd_sof_dev *sdev); diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index e929a6e0058fc1..7f532bcc8e9dd3 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -146,10 +146,11 @@ static inline int snd_sof_dsp_resume(struct snd_sof_dev *sdev) return 0; } -static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev) +static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev, + u32 target_state) { if (sof_ops(sdev)->suspend) - return sof_ops(sdev)->suspend(sdev); + return sof_ops(sdev)->suspend(sdev, target_state); return 0; } @@ -193,14 +194,15 @@ static inline int snd_sof_dsp_set_clk(struct snd_sof_dev *sdev, u32 freq) return 0; } -static inline int snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev, - enum sof_d0_substate substate) +static inline int +snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev, + const struct sof_dsp_power_state *target_state) { if (sof_ops(sdev)->set_power_state) - return sof_ops(sdev)->set_power_state(sdev, substate); + return sof_ops(sdev)->set_power_state(sdev, target_state); - /* D0 substate is not supported */ - return -ENOTSUPP; + /* D0 substate is not supported, do nothing here. */ + return 0; } /* debug */ diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index bec25cb6beeccf..c410822d9920db 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -86,7 +86,7 @@ static void sof_cache_debugfs(struct snd_sof_dev *sdev) static int sof_resume(struct device *dev, bool runtime_resume) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); - enum sof_d0_substate old_d0_substate = sdev->d0_substate; + u32 old_state = sdev->dsp_power_state.state; int ret; /* do nothing if dsp resume callbacks are not set */ @@ -97,17 +97,6 @@ static int sof_resume(struct device *dev, bool runtime_resume) if (sdev->first_boot) return 0; - /* resume from D0I3 */ - if (!runtime_resume && old_d0_substate == SOF_DSP_D0I3) { - ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I0); - if (ret < 0 && ret != -ENOTSUPP) { - dev_err(sdev->dev, - "error: failed to resume from D0I3 %d\n", - ret); - return ret; - } - } - /* * if the runtime_resume flag is set, call the runtime_resume routine * or else call the system resume routine @@ -122,8 +111,8 @@ static int sof_resume(struct device *dev, bool runtime_resume) return ret; } - /* Nothing further to do if resuming from D0I3 */ - if (!runtime_resume && old_d0_substate == SOF_DSP_D0I3) + /* Nothing further to do if resuming from a low-power D0 substate */ + if (!runtime_resume && old_state == SOF_DSP_PM_D0) return 0; sdev->fw_state = SOF_FW_BOOT_PREPARE; @@ -176,15 +165,13 @@ static int sof_resume(struct device *dev, bool runtime_resume) "error: ctx_restore ipc error during resume %d\n", ret); - /* initialize default D0 sub-state */ - sdev->d0_substate = SOF_DSP_D0I0; - return ret; } static int sof_suspend(struct device *dev, bool runtime_suspend) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + u32 target_state = 0; int ret; /* do nothing if dsp suspend callback is not set */ @@ -205,18 +192,11 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) } } - if (snd_sof_dsp_power_target(sdev) == SOF_DSP_PM_D0) { - /* suspend to D0i3 */ - ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I3); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to enter D0I3, %d\n", - ret); - return ret; - } + target_state = snd_sof_dsp_power_target(sdev); - /* Skip to platform-specific suspend if DSP is entering D0I3 */ + /* Skip to platform-specific suspend if DSP is entering D0 */ + if (target_state == SOF_DSP_PM_D0) goto suspend; - } /* release trace */ snd_sof_release_trace(sdev); @@ -254,14 +234,14 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) if (runtime_suspend) ret = snd_sof_dsp_runtime_suspend(sdev); else - ret = snd_sof_dsp_suspend(sdev); + ret = snd_sof_dsp_suspend(sdev, target_state); if (ret < 0) dev_err(sdev->dev, "error: failed to power down DSP during suspend %d\n", ret); - /* Do not reset FW state if DSP is in D0I3 */ - if (sdev->d0_substate == SOF_DSP_D0I3) + /* Do not reset FW state if DSP is in D0 */ + if (target_state == SOF_DSP_PM_D0) return ret; /* reset FW state */ @@ -290,58 +270,6 @@ int snd_sof_runtime_resume(struct device *dev) } EXPORT_SYMBOL(snd_sof_runtime_resume); -int snd_sof_set_d0_substate(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate) -{ - int ret; - - if (sdev->d0_substate == d0_substate) - return 0; - - /* do platform specific set_state */ - ret = snd_sof_dsp_set_power_state(sdev, d0_substate); - if (ret < 0) - return ret; - - /* update dsp D0 sub-state */ - sdev->d0_substate = d0_substate; - - return 0; -} -EXPORT_SYMBOL(snd_sof_set_d0_substate); - -/* - * Audio DSP states may transform as below:- - * - * D0I3 compatible stream - * Runtime +---------------------+ opened only, timeout - * suspend | +--------------------+ - * +------------+ D0(active) | | - * | | <---------------+ | - * | +--------> | | | - * | |Runtime +--^--+---------^--+--+ The last | | - * | |resume | | | | opened D0I3 | | - * | | | | | | compatible | | - * | | resume| | | | stream closed | | - * | | from | | D3 | | | | - * | | D3 | |suspend | | d0i3 | | - * | | | | | |suspend | | - * | | | | | | | | - * | | | | | | | | - * +-v---+-----------+--v-------+ | | +------+----v----+ - * | | | +-----------> | - * | D3 (suspended) | | | D0I3 +-----+ - * | | +--------------+ | | - * | | resume from | | | - * +-------------------^--------+ d0i3 suspend +----------------+ | - * | | - * | D3 suspend | - * +------------------------------------------------+ - * - * d0i3_suspend = s0_suspend && D0I3 stream opened, - * D3 suspend = !d0i3_suspend, - */ - int snd_sof_resume(struct device *dev) { return sof_resume(dev, false); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index a7c6109acd98f3..ef33aaadbc310d 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -64,10 +64,9 @@ enum sof_dsp_power_states { SOF_DSP_PM_D3_COLD, }; -/* DSP D0ix sub-state */ -enum sof_d0_substate { - SOF_DSP_D0I0 = 0, /* DSP default D0 substate */ - SOF_DSP_D0I3, /* DSP D0i3(low power) substate*/ +struct sof_dsp_power_state { + u32 state; + u32 substate; /* platform-specific */ }; /* System suspend target state */ @@ -186,14 +185,15 @@ struct snd_sof_dsp_ops { int (*post_fw_run)(struct snd_sof_dev *sof_dev); /* optional */ /* DSP PM */ - int (*suspend)(struct snd_sof_dev *sof_dev); /* optional */ + int (*suspend)(struct snd_sof_dev *sof_dev, + u32 target_state); /* optional */ int (*resume)(struct snd_sof_dev *sof_dev); /* optional */ int (*runtime_suspend)(struct snd_sof_dev *sof_dev); /* optional */ int (*runtime_resume)(struct snd_sof_dev *sof_dev); /* optional */ int (*runtime_idle)(struct snd_sof_dev *sof_dev); /* optional */ int (*set_hw_params_upon_resume)(struct snd_sof_dev *sdev); /* optional */ int (*set_power_state)(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate); /* optional */ + const struct sof_dsp_power_state *target_state); /* optional */ /* DSP clocking */ int (*set_clk)(struct snd_sof_dev *sof_dev, u32 freq); /* optional */ @@ -340,8 +340,8 @@ struct snd_sof_dev { */ struct snd_soc_component_driver plat_drv; - /* power states related */ - enum sof_d0_substate d0_substate; + /* current DSP power state */ + struct sof_dsp_power_state dsp_power_state; /* Intended power target of system suspend */ enum sof_system_suspend_state system_suspend_target; @@ -435,8 +435,6 @@ int snd_sof_resume(struct device *dev); int snd_sof_suspend(struct device *dev); int snd_sof_prepare(struct device *dev); void snd_sof_complete(struct device *dev); -int snd_sof_set_d0_substate(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate); void snd_sof_new_platform_drv(struct snd_sof_dev *sdev); From de23a838d8d61767c6232f229f019eb46401cb93 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:23 -0600 Subject: [PATCH 053/415] ASoC: SOF: audio: Add helper to check if only D0i3 streams are active Add a helper function to check if only D0i3-compatible streams are active. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/sof/sof-audio.h | 1 + 2 files changed, 34 insertions(+) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index d16571ca129cec..75f2ef2bd94be0 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -11,6 +11,39 @@ #include "sof-audio.h" #include "ops.h" +/* + * helper to determine if there are only D0i3 compatible + * streams active + */ +bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev) +{ + struct snd_pcm_substream *substream; + struct snd_sof_pcm *spcm; + bool d0i3_compatible_active = false; + int dir; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + substream = spcm->stream[dir].substream; + if (!substream || !substream->runtime) + continue; + + /* + * substream->runtime being not NULL indicates that + * that the stream is open. No need to check the + * stream state. + */ + if (!spcm->stream[dir].d0i3_compatible) + return false; + + d0i3_compatible_active = true; + } + } + + return d0i3_compatible_active; +} +EXPORT_SYMBOL(snd_sof_dsp_only_d0i3_compatible_stream_active); + bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev) { struct snd_sof_pcm *spcm; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index a2702afbd9a121..eacd10e4da11a3 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -203,6 +203,7 @@ int snd_sof_ipc_set_get_comp_data(struct snd_sof_control *scontrol, int sof_restore_pipelines(struct device *dev); int sof_set_hw_params_upon_resume(struct device *dev); bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev); +bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev); /* Machine driver enumeration */ int sof_machine_register(struct snd_sof_dev *sdev, void *pdata); From 207bf12f642f39e749ca65d3efca9d48311e629f Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:24 -0600 Subject: [PATCH 054/415] ASoC: SOF: Intel: hda: Amend the DSP state transition diagram Amend the DSP state transition diagram in preparation for introducing the feature to support opportunistic DSP D0I3 state when the system is in S0. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 36 +++++++++++++++++------------------ 1 file changed, 17 insertions(+), 19 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 8c00e128a7b047..7b8425330ae0b1 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -488,33 +488,31 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev, /* * Audio DSP states may transform as below:- * - * D0I3 compatible stream - * Runtime +---------------------+ opened only, timeout + * Opportunistic D0I3 in S0 + * Runtime +---------------------+ Delayed D0i3 work timeout * suspend | +--------------------+ - * +------------+ D0(active) | | + * +------------+ D0I0(active) | | * | | <---------------+ | - * | +--------> | | | - * | |Runtime +--^--+---------^--+--+ The last | | - * | |resume | | | | opened D0I3 | | - * | | | | | | compatible | | - * | | resume| | | | stream closed | | - * | | from | | D3 | | | | - * | | D3 | |suspend | | d0i3 | | + * | +--------> | New IPC | | + * | |Runtime +--^--+---------^--+--+ (via mailbox) | | + * | |resume | | | | | | + * | | | | | | | | + * | | System| | | | | | + * | | resume| | S3/S0IX | | | | + * | | | | suspend | | S0IX | | * | | | | | |suspend | | * | | | | | | | | * | | | | | | | | * +-v---+-----------+--v-------+ | | +------+----v----+ * | | | +-----------> | - * | D3 (suspended) | | | D0I3 +-----+ - * | | +--------------+ | | - * | | resume from | | | - * +-------------------^--------+ d0i3 suspend +----------------+ | - * | | - * | D3 suspend | - * +------------------------------------------------+ + * | D3 (suspended) | | | D0I3 | + * | | +--------------+ | + * | | System resume | | + * +----------------------------+ +----------------+ * - * d0i3_suspend = s0_suspend && D0I3 stream opened, - * D3 suspend = !d0i3_suspend, + * S0IX suspend: The DSP is in D0I3 if any D0I3-compatible streams + * ignored the suspend trigger. Otherwise the DSP + * is in D3. */ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) From 63e51fd33fef04b634a0c32ae491ab16a19cb17c Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:25 -0600 Subject: [PATCH 055/415] ASoC: SOF: Intel: cnl: Implement feature to support DSP D0i3 in S0 This patch implements support for DSP D0i3 when the system is in S0. The basic idea is to schedule a delayed work after every successful IPC TX that checks if there are only D0I3-compatible streams active and if so transition the DSP to D0I3. With the introduction of DSP D0I3 in S0, we need to ensure that the DSP is in D0I0 before sending any new IPCs. The exception for this would be the compact IPCs that are used to set the DSP in D0I3/D0I0 states. Signed-off-by: Keyon Jie Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 37 +++++++++++++++++++++++++++------ sound/soc/sof/intel/hda-dsp.c | 39 +++++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda.c | 5 +++++ sound/soc/sof/intel/hda.h | 11 ++++++++++ sound/soc/sof/ipc.c | 29 ++++++++++++++++++++++++-- sound/soc/sof/sof-priv.h | 3 +++ 6 files changed, 114 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 9e2d8afe053510..8a59fec7291999 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -171,23 +171,48 @@ static bool cnl_compact_ipc_compress(struct snd_sof_ipc_msg *msg, static int cnl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) { + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; + struct sof_ipc_cmd_hdr *hdr; u32 dr = 0; u32 dd = 0; + /* + * Currently the only compact IPC supported is the PM_GATE + * IPC which is used for transitioning the DSP between the + * D0I0 and D0I3 states. And these are sent only during the + * set_power_state() op. Therefore, there will never be a case + * that a compact IPC results in the DSP exiting D0I3 without + * the host and FW being in sync. + */ if (cnl_compact_ipc_compress(msg, &dr, &dd)) { /* send the message via IPC registers */ snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDD, dd); snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR, CNL_DSP_REG_HIPCIDR_BUSY | dr); - } else { - /* send the message via mailbox */ - sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, - msg->msg_size); - snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR, - CNL_DSP_REG_HIPCIDR_BUSY); + return 0; } + /* send the message via mailbox */ + sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, + msg->msg_size); + snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR, + CNL_DSP_REG_HIPCIDR_BUSY); + + hdr = msg->msg_data; + + /* + * Use mod_delayed_work() to schedule the delayed work + * to avoid scheduling multiple workqueue items when + * IPCs are sent at a high-rate. mod_delayed_work() + * modifies the timer if the work is pending. + * Also, a new delayed work should not be queued after the + * the CTX_SAVE IPC, which is sent before the DSP enters D3. + */ + if (hdr->cmd != (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE)) + mod_delayed_work(system_wq, &hdev->d0i3_work, + msecs_to_jiffies(SOF_HDA_D0I3_WORK_DELAY_MS)); + return 0; } diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 7b8425330ae0b1..ee604be715b9e9 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -17,6 +17,7 @@ #include #include +#include "../sof-audio.h" #include "../ops.h" #include "hda.h" #include "hda-ipc.h" @@ -334,8 +335,9 @@ static int hda_dsp_send_pm_gate_ipc(struct snd_sof_dev *sdev, u32 flags) pm_gate.flags = flags; /* send pm_gate ipc to dsp */ - return sof_ipc_tx_message(sdev->ipc, pm_gate.hdr.cmd, &pm_gate, - sizeof(pm_gate), &reply, sizeof(reply)); + return sof_ipc_tx_message_no_pm(sdev->ipc, pm_gate.hdr.cmd, + &pm_gate, sizeof(pm_gate), &reply, + sizeof(reply)); } static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) @@ -706,6 +708,9 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) }; int ret; + /* cancel any attempt for DSP D0I3 */ + cancel_delayed_work_sync(&hda->d0i3_work); + if (target_state == SOF_DSP_PM_D0) { /* Set DSP power state */ ret = hda_dsp_set_power_state(sdev, &target_dsp_state); @@ -780,3 +785,33 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) #endif return 0; } + +void hda_dsp_d0i3_work(struct work_struct *work) +{ + struct sof_intel_hda_dev *hdev = container_of(work, + struct sof_intel_hda_dev, + d0i3_work.work); + struct hdac_bus *bus = &hdev->hbus.core; + struct snd_sof_dev *sdev = dev_get_drvdata(bus->dev); + struct sof_dsp_power_state target_state; + int ret; + + target_state.state = SOF_DSP_PM_D0; + + /* DSP can enter D0I3 iff only D0I3-compatible streams are active */ + if (snd_sof_dsp_only_d0i3_compatible_stream_active(sdev)) + target_state.substate = SOF_HDA_DSP_PM_D0I3; + else + target_state.substate = SOF_HDA_DSP_PM_D0I0; + + /* remain in D0I0 */ + if (target_state.substate == SOF_HDA_DSP_PM_D0I0) + return; + + /* This can fail but error cannot be propagated */ + ret = hda_dsp_set_power_state(sdev, &target_state); + if (ret < 0) + dev_err_ratelimited(sdev->dev, + "error: failed to set DSP state %d substate %d\n", + target_state.state, target_state.substate); +} diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 65b86dd044f101..2b8754a7658433 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -598,6 +598,8 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) /* set default mailbox offset for FW ready message */ sdev->dsp_box.offset = HDA_DSP_MBOX_UPLINK_OFFSET; + INIT_DELAYED_WORK(&hdev->d0i3_work, hda_dsp_d0i3_work); + return 0; free_ipc_irq: @@ -622,6 +624,9 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) struct pci_dev *pci = to_pci_dev(sdev->dev); const struct sof_intel_dsp_desc *chip = hda->desc; + /* cancel any attempt for DSP D0I3 */ + cancel_delayed_work_sync(&hda->d0i3_work); + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* codec removal, invoke bus_device_remove */ snd_hdac_ext_bus_device_remove(bus); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 02c2a7eadb1bc6..a46b66437a3d20 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -392,6 +392,13 @@ struct sof_intel_dsp_bdl { #define SOF_HDA_PLAYBACK 0 #define SOF_HDA_CAPTURE 1 +/* + * Time in ms for opportunistic D0I3 entry delay. + * This has been deliberately chosen to be long to avoid race conditions. + * Could be optimized in future. + */ +#define SOF_HDA_D0I3_WORK_DELAY_MS 5000 + /* HDA DSP D0 substate */ enum sof_hda_D0_substate { SOF_HDA_DSP_PM_D0I0, /* default D0 substate */ @@ -420,6 +427,9 @@ struct sof_intel_hda_dev { /* DMIC device */ struct platform_device *dmic_dev; + + /* delayed work to enter D0I3 opportunistically */ + struct delayed_work d0i3_work; }; static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s) @@ -487,6 +497,7 @@ void hda_dsp_dump_skl(struct snd_sof_dev *sdev, u32 flags); void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags); void hda_ipc_dump(struct snd_sof_dev *sdev); void hda_ipc_irq_dump(struct snd_sof_dev *sdev); +void hda_dsp_d0i3_work(struct work_struct *work); /* * DSP PCM Operations. diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index b63fc529b4568e..22d296f957610f 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -268,7 +268,6 @@ static int sof_ipc_tx_message_unlocked(struct snd_sof_ipc *ipc, u32 header, spin_unlock_irq(&sdev->ipc_lock); if (ret < 0) { - /* So far IPC TX never fails, consider making the above void */ dev_err_ratelimited(sdev->dev, "error: ipc tx failed with error %d\n", ret); @@ -288,6 +287,32 @@ static int sof_ipc_tx_message_unlocked(struct snd_sof_ipc *ipc, u32 header, int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, void *msg_data, size_t msg_bytes, void *reply_data, size_t reply_bytes) +{ + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + }; + int ret; + + /* ensure the DSP is in D0 before sending a new IPC */ + ret = snd_sof_dsp_set_power_state(ipc->sdev, &target_state); + if (ret < 0) { + dev_err(ipc->sdev->dev, "error: resuming DSP %d\n", ret); + return ret; + } + + return sof_ipc_tx_message_no_pm(ipc, header, msg_data, msg_bytes, + reply_data, reply_bytes); +} +EXPORT_SYMBOL(sof_ipc_tx_message); + +/* + * send IPC message from host to DSP without modifying the DSP state. + * This will be used for IPC's that can be handled by the DSP + * even in a low-power D0 substate. + */ +int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header, + void *msg_data, size_t msg_bytes, + void *reply_data, size_t reply_bytes) { int ret; @@ -305,7 +330,7 @@ int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, return ret; } -EXPORT_SYMBOL(sof_ipc_tx_message); +EXPORT_SYMBOL(sof_ipc_tx_message_no_pm); /* handle reply message from DSP */ int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index ef33aaadbc310d..00084471d0de19 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -470,6 +470,9 @@ int snd_sof_ipc_valid(struct snd_sof_dev *sdev); int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, void *msg_data, size_t msg_bytes, void *reply_data, size_t reply_bytes); +int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header, + void *msg_data, size_t msg_bytes, + void *reply_data, size_t reply_bytes); /* * Trace/debug From 851fd87324430dfe56cd55dfd05a8114ac82d168 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 29 Jan 2020 16:07:26 -0600 Subject: [PATCH 056/415] ASoC: SOF: Intel: hda: Allow trace DMA in S0 when DSP is in D0I3 for debug Trace DMA is disabled by default when the DSP is in D0I3. Add a debug option to keep trace DMA enabled when the DSP is in D0I3 during S0. Signed-off-by: Ranjani Sridharan Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200129220726.31792-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 35 ++++++++++++++++++++++++++++++++--- 1 file changed, 32 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index ee604be715b9e9..14228b4931d6b5 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -15,6 +15,7 @@ * Hardware interface for generic Intel audio DSP HDA IP */ +#include #include #include #include "../sof-audio.h" @@ -22,6 +23,13 @@ #include "hda.h" #include "hda-ipc.h" +static bool hda_enable_trace_D0I3_S0; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG) +module_param_named(enable_trace_D0I3_S0, hda_enable_trace_D0I3_S0, bool, 0444); +MODULE_PARM_DESC(enable_trace_D0I3_S0, + "SOF HDA enable trace when the DSP is in D0I3 in S0"); +#endif + /* * DSP Core control. */ @@ -399,8 +407,14 @@ static int hda_dsp_set_D0_state(struct snd_sof_dev *sdev, if (target_state->substate == SOF_HDA_DSP_PM_D0I3) { value = SOF_HDA_VS_D0I3C_I3; - /* disable DMA trace in D0I3 */ - flags = HDA_PM_NO_DMA_TRACE; + /* + * Trace DMA is disabled by default when the DSP enters D0I3. + * But it can be kept enabled when the DSP enters D0I3 while the + * system is in S0 for debug. + */ + if (hda_enable_trace_D0I3_S0 && + sdev->system_suspend_target != SOF_SUSPEND_NONE) + flags = HDA_PM_NO_DMA_TRACE; } else { /* prevent power gating in D0I0 */ flags = HDA_PM_PPG; @@ -450,11 +464,26 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev, { int ret = 0; - /* Nothing to do if the DSP is already in the requested state */ + /* + * When the DSP is already in D0I3 and the target state is D0I3, + * it could be the case that the DSP is in D0I3 during S0 + * and the system is suspending to S0Ix. Therefore, + * hda_dsp_set_D0_state() must be called to disable trace DMA + * by sending the PM_GATE IPC to the FW. + */ + if (target_state->substate == SOF_HDA_DSP_PM_D0I3 && + sdev->system_suspend_target == SOF_SUSPEND_S0IX) + goto set_state; + + /* + * For all other cases, return without doing anything if + * the DSP is already in the target state. + */ if (target_state->state == sdev->dsp_power_state.state && target_state->substate == sdev->dsp_power_state.substate) return 0; +set_state: switch (target_state->state) { case SOF_DSP_PM_D0: ret = hda_dsp_set_D0_state(sdev, target_state); From fa1f875c120fa44572c561d86022af2f6b0774c7 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2020 17:14:02 +0200 Subject: [PATCH 057/415] ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback Some DMA engines can have big FIFOs which adds to the latency. The DMAengine framework can report the FIFO utilization in bytes. Use this information for the delay reporting. Signed-off-by: Peter Ujfalusi Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200210151402.29634-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/core/pcm_dmaengine.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 5749a8a4978481..d8be7b48816283 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -247,9 +247,14 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { + struct snd_pcm_runtime *runtime = substream->runtime; + buf_size = snd_pcm_lib_buffer_bytes(substream); if (state.residue > 0 && state.residue <= buf_size) pos = buf_size - state.residue; + + runtime->delay = bytes_to_frames(runtime, + state.in_flight_bytes); } return bytes_to_frames(substream->runtime, pos); From 7a5aaba4a4f45acc8192beb8a4b1bd4a58b67ce3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Feb 2020 12:14:12 +0900 Subject: [PATCH 058/415] ASoC: soc-pcm: add snd_soc_runtime_action() ALSA SoC has snd_soc_runtime_activate() / snd_soc_runtime_deactivate(). These increment or decrement DAI/Component activity, but the code difference is only +1 or -1. This patch adds common snd_soc_runtime_action() which can get +1 or -1 as parameter, and use it from snd_soc_runtime_activate/deactivate() to avoid duplicate implementation. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/87blq7ceyq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 67 ++++++++++++++++++--------------------------- 1 file changed, 26 insertions(+), 41 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ff1b7c7078e5f1..4d26558fcbfc26 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -82,17 +82,8 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd, return 0; } -/** - * snd_soc_runtime_activate() - Increment active count for PCM runtime components - * @rtd: ASoC PCM runtime that is activated - * @stream: Direction of the PCM stream - * - * Increments the active count for all the DAIs and components attached to a PCM - * runtime. Should typically be called when a stream is opened. - * - * Must be called with the rtd->card->pcm_mutex being held - */ -void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) +static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, + int stream, int action) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; @@ -101,23 +92,38 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) lockdep_assert_held(&rtd->card->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active++; + cpu_dai->playback_active += action; for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->playback_active++; + codec_dai->playback_active += action; } else { - cpu_dai->capture_active++; + cpu_dai->capture_active += action; for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->capture_active++; + codec_dai->capture_active += action; } - cpu_dai->active++; - cpu_dai->component->active++; + cpu_dai->active += action; + cpu_dai->component->active += action; for_each_rtd_codec_dai(rtd, i, codec_dai) { - codec_dai->active++; - codec_dai->component->active++; + codec_dai->active += action; + codec_dai->component->active += action; } } +/** + * snd_soc_runtime_activate() - Increment active count for PCM runtime components + * @rtd: ASoC PCM runtime that is activated + * @stream: Direction of the PCM stream + * + * Increments the active count for all the DAIs and components attached to a PCM + * runtime. Should typically be called when a stream is opened. + * + * Must be called with the rtd->card->pcm_mutex being held + */ +void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) +{ + snd_soc_runtime_action(rtd, stream, 1); +} + /** * snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components * @rtd: ASoC PCM runtime that is deactivated @@ -130,28 +136,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) */ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int i; - - lockdep_assert_held(&rtd->card->pcm_mutex); - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active--; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->capture_active--; - } - - cpu_dai->active--; - cpu_dai->component->active--; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - codec_dai->component->active--; - codec_dai->active--; - } + snd_soc_runtime_action(rtd, stream, -1); } /** From 5c25bd641a7b195b5ed71ce9d6955618bae7b7d3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Feb 2020 12:14:18 +0900 Subject: [PATCH 059/415] ASoC: soc-pcm: adjustment for DAI member 0 reset commit 3635bf09a89cf ("ASoC: soc-pcm: add symmetry for channels and sample bits") set 0 not only to dai->rate but also to dai->channels and dai->sample_bits if DAI was not active at soc_pcm_close(). and commit d3383420c969c ("ASoC: soc-pcm: move DAIs parameters cleaning into hw_free()") moved it from soc_pcm_close() to soc_pcm_hw_free(). These happen at v3.14. But, maybe because of branch merge conflict or something similar happen then, soc_pcm_close() still has old settings (care only dai->rate, doesn't care dai->channels/sample_bits). This is 100% duplicated operation. This patch removes soc_pcm_close() side operation which supposed to already moved to soc_pcm_hw_free(). Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/87a75rceyl.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4d26558fcbfc26..2a4f7ac5f56379 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -687,15 +687,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) snd_soc_runtime_deactivate(rtd, substream->stream); - /* clear the corresponding DAIs rate when inactive */ - if (!cpu_dai->active) - cpu_dai->rate = 0; - - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (!codec_dai->active) - codec_dai->rate = 0; - } - snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream); snd_soc_dai_shutdown(cpu_dai, substream); From 09e88f8a5c56ac5258935a5a543868c20a55d4dd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Feb 2020 12:14:22 +0900 Subject: [PATCH 060/415] ASoC: soc-pcm: add for_each_dapm_widgets() macro This patch adds new for_each_dapm_widgets() macro and use it. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/878slbceyg.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 5 +++++ sound/soc/soc-dapm.c | 8 ++------ sound/soc/soc-pcm.c | 17 +++++++++-------- 3 files changed, 16 insertions(+), 14 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2a306c6f3fbcb7..9439e75945f6ce 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -693,6 +693,11 @@ struct snd_soc_dapm_widget_list { struct snd_soc_dapm_widget *widgets[0]; }; +#define for_each_dapm_widgets(list, i, widget) \ + for ((i) = 0; \ + (i) < list->num_widgets && (widget = list->widgets[i]); \ + (i)++) + struct snd_soc_dapm_stats { int power_checks; int path_checks; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bc20ad9abf8bdd..cc17a3730d3d2a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1724,9 +1724,7 @@ static void dapm_widget_update(struct snd_soc_card *card) wlist = dapm_kcontrol_get_wlist(update->kcontrol); - for (wi = 0; wi < wlist->num_widgets; wi++) { - w = wlist->widgets[wi]; - + for_each_dapm_widgets(wlist, wi, w) { if (w->event && (w->event_flags & SND_SOC_DAPM_PRE_REG)) { ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG); if (ret != 0) @@ -1753,9 +1751,7 @@ static void dapm_widget_update(struct snd_soc_card *card) w->name, ret); } - for (wi = 0; wi < wlist->num_widgets; wi++) { - w = wlist->widgets[wi]; - + for_each_dapm_widgets(wlist, wi, w) { if (w->event && (w->event_flags & SND_SOC_DAPM_POST_REG)) { ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG); if (ret != 0) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2a4f7ac5f56379..7a490c05d4e99c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1306,12 +1306,12 @@ static inline struct snd_soc_dapm_widget * static int widget_in_list(struct snd_soc_dapm_widget_list *list, struct snd_soc_dapm_widget *widget) { + struct snd_soc_dapm_widget *w; int i; - for (i = 0; i < list->num_widgets; i++) { - if (widget == list->widgets[i]) + for_each_dapm_widgets(list, i, w) + if (widget == w) return 1; - } return 0; } @@ -1422,12 +1422,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_card *card = fe->card; struct snd_soc_dapm_widget_list *list = *list_; struct snd_soc_pcm_runtime *be; + struct snd_soc_dapm_widget *widget; int i, new = 0, err; /* Create any new FE <--> BE connections */ - for (i = 0; i < list->num_widgets; i++) { + for_each_dapm_widgets(list, i, widget) { - switch (list->widgets[i]->id) { + switch (widget->id) { case snd_soc_dapm_dai_in: if (stream != SNDRV_PCM_STREAM_PLAYBACK) continue; @@ -1441,10 +1442,10 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, } /* is there a valid BE rtd for this widget */ - be = dpcm_get_be(card, list->widgets[i], stream); + be = dpcm_get_be(card, widget, stream); if (!be) { dev_err(fe->dev, "ASoC: no BE found for %s\n", - list->widgets[i]->name); + widget->name); continue; } @@ -1460,7 +1461,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, err = dpcm_be_connect(fe, be, stream); if (err < 0) { dev_err(fe->dev, "ASoC: can't connect %s\n", - list->widgets[i]->name); + widget->name); break; } else if (err == 0) /* already connected */ continue; From e82ebffce3ec07584bcc2fc4c4d33a43fd9515f5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Feb 2020 12:14:26 +0900 Subject: [PATCH 061/415] ASoC: soc-pcm: don't use bit-OR'ed error Current soc-pcm is using bit-OR'ed error ret |= snd_soc_component_close(component, substream); ret |= snd_soc_component_hw_free(component, substream); The driver may return arbitrary error codes so they can conflict. The bit-OR'ed error works only if the return code is always consistent. This patch fixup it, and use *last* ret value. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/877e0vceyc.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7a490c05d4e99c..8d8ed4774e9ca8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -498,13 +498,16 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - int i, ret = 0; + int i, r, ret = 0; for_each_rtd_components(rtd, i, component) { if (component == last) break; - ret |= snd_soc_component_close(component, substream); + r = snd_soc_component_close(component, substream); + if (r < 0) + ret = r; /* use last ret */ + snd_soc_component_module_put_when_close(component); } @@ -798,13 +801,15 @@ static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - int i, ret = 0; + int i, r, ret = 0; for_each_rtd_components(rtd, i, component) { if (component == last) break; - ret |= snd_soc_component_hw_free(component, substream); + r = snd_soc_component_hw_free(component, substream); + if (r < 0) + ret = r; /* use last ret */ } return ret; From b56be800f1292c9b79c4f66571c701551bdf9e12 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Feb 2020 12:14:33 +0900 Subject: [PATCH 062/415] ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once Current soc_pcm_open() calls snd_soc_dai_startup() under loop. Thus, it needs to care about started/not-yet-started codec DAI. But, if soc-dai.c is handling it, soc-pcm.c don't need to care about it. This patch adds started flag to soc-dai.h, and simplify soc-pcm.c. This is one of prepare for cleanup soc-pcm-open() Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/875zgfcey5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 5 ++++- sound/soc/soc-dai.c | 11 +++++++++-- sound/soc/soc-pcm.c | 7 ++----- 3 files changed, 15 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index eaaeb00e9e84f6..04c23ac0dfffdf 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -324,7 +324,6 @@ struct snd_soc_dai { /* DAI runtime info */ unsigned int capture_active; /* stream usage count */ unsigned int playback_active; /* stream usage count */ - unsigned int probed:1; unsigned int active; @@ -348,6 +347,10 @@ struct snd_soc_dai { unsigned int rx_mask; struct list_head list; + + /* bit field */ + unsigned int probed:1; + unsigned int started:1; }; static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 51031e330179dd..73a829393652ac 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,17 +295,24 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (dai->driver->ops->startup) + if (!dai->started && + dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); + if (ret == 0) + dai->started = 1; + return ret; } void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->driver->ops->shutdown) + if (dai->started && + dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); + + dai->started = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8d8ed4774e9ca8..d53afb96b05bb7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -568,7 +568,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) { pr_err("ASoC: %s startup failed: %d\n", rtd->dai_link->name, ret); - goto machine_err; + goto codec_dai_err; } /* Dynamic PCM DAI links compat checks use dynamic capabilities */ @@ -637,11 +637,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) config_err: soc_rtd_shutdown(rtd, substream); -machine_err: - i = rtd->num_codecs; - codec_dai_err: - for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) + for_each_rtd_codec_dai(rtd, i, codec_dai) snd_soc_dai_shutdown(codec_dai, substream); component_err: From 9d789dc047e32fb0f85ff192f883a534017512a2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2020 17:33:36 +0200 Subject: [PATCH 063/415] ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback Some DMA engines can have big FIFOs which adds to the latency. The DMAengine framework can report the FIFO utilization in bytes. Use this information for the delay reporting. Signed-off-by: Peter Ujfalusi Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200210153336.10218-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/core/pcm_dmaengine.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index d8be7b48816283..6852bb670b4ef5 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -240,6 +240,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; struct dma_tx_state state; enum dma_status status; unsigned int buf_size; @@ -257,7 +258,7 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) state.in_flight_bytes); } - return bytes_to_frames(substream->runtime, pos); + return bytes_to_frames(runtime, pos); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); From 9478bd43a2eb5c72b599368513d10880b296d65f Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Tue, 11 Feb 2020 13:39:10 -0600 Subject: [PATCH 064/415] ALSA: core: Replace zero-length array with flexible-array member The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva Link: https://lore.kernel.org/r/20200211193910.GA4596@embeddedor Signed-off-by: Takashi Iwai --- sound/core/oss/rate.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 7cd09cef6961db..d381f4c967c902 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -47,7 +47,7 @@ struct rate_priv { unsigned int pos; rate_f func; snd_pcm_sframes_t old_src_frames, old_dst_frames; - struct rate_channel channels[0]; + struct rate_channel channels[]; }; static void rate_init(struct snd_pcm_plugin *plugin) From 6c8019d08e0e84ecad78f726c737dc6a49b58b57 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Tue, 11 Feb 2020 13:42:24 -0600 Subject: [PATCH 065/415] ALSA: usb-midi: Replace zero-length array with flexible-array member The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva Link: https://lore.kernel.org/r/20200211194224.GA9383@embeddedor Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 392e5fda680c7b..be5c597ed40c84 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -91,7 +91,7 @@ struct usb_ms_endpoint_descriptor { __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bNumEmbMIDIJack; - __u8 baAssocJackID[0]; + __u8 baAssocJackID[]; } __attribute__ ((packed)); struct snd_usb_midi_in_endpoint; From 76501954cb9ef5b3d614a075870cfce47fbbd6df Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Tue, 11 Feb 2020 13:44:03 -0600 Subject: [PATCH 066/415] ALSA: hda/ca0132 - Replace zero-length array with flexible-array member The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva Link: https://lore.kernel.org/r/20200211194403.GA10318@embeddedor Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index ded8bc07d755ac..a4a39f7d9ae1aa 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2698,7 +2698,7 @@ struct dsp_image_seg { u32 magic; u32 chip_addr; u32 count; - u32 data[0]; + u32 data[]; }; static const u32 g_magic_value = 0x4c46584d; From bb80b96422b443ddcb6dbb39261ea3e02a13661d Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Tue, 11 Feb 2020 14:07:39 -0600 Subject: [PATCH 067/415] ALSA: hda_codec: Replace zero-length array with flexible-array member The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva Link: https://lore.kernel.org/r/20200211200739.GA12948@embeddedor Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5dc42f932739f3..97a03685dd8bdc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -88,7 +88,7 @@ struct hda_conn_list { struct list_head list; int len; hda_nid_t nid; - hda_nid_t conns[0]; + hda_nid_t conns[]; }; /* look up the cached results */ From 82dabf599b221a712e951b9061c56669565552a9 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 12 Feb 2020 09:50:08 +0100 Subject: [PATCH 068/415] ASoC: sh: fsi: Restore devm_ioremap() alignment The alignment of the continuation of the devm_ioremap() call in fsi_probe() was broken. Join the lines, as all parameters can fit on a single line. Fixes: 4bdc0d676a643140 ("remove ioremap_nocache and devm_ioremap_nocache") Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/20200212085008.9652-1-geert+renesas@glider.be Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4b35ef402604ec..5ef4221be6c396 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1938,8 +1938,7 @@ static int fsi_probe(struct platform_device *pdev) if (!master) return -ENOMEM; - master->base = devm_ioremap(&pdev->dev, - res->start, resource_size(res)); + master->base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); if (!master->base) { dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); return -ENXIO; From 681c896ceb411ccd2ce0a88059d86ccf8d7a497e Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Tue, 11 Feb 2020 14:05:49 -0600 Subject: [PATCH 069/415] ASoC: wm0010: Replace zero-length array with flexible-array member The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20200211200549.GA12072@embeddedor Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 727d6703c905a2..fbcee21736e85e 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -43,7 +43,7 @@ struct dfw_binrec { u8 command; u32 length:24; u32 address; - uint8_t data[0]; + uint8_t data[]; } __packed; struct dfw_inforec { From 128f825aeab79ebff9679a84f49105eda85ecf2c Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Wed, 12 Feb 2020 13:55:16 +0800 Subject: [PATCH 070/415] ASoC: max98357a: move control of SD_MODE to DAPM Some machine may share the same I2S lines for multiple codecs. For example, mediatek/mt8183/mt8183-da7219-max98357 shares the same lines between max98357a and da7219. When writing audio data through the I2S lines, all codecs on the lines would try to generate sound if they accepts DO line. As a result, multiple codecs generate sound at a time. Moves control of SD_MODE to DAPM so that machine drivers have chances to manipulate DAPM widget to turn on/off max98357a. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200212124608.1.I73b26b5e319de173d05823e79f5861bf1826261c@changeid Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 36 +++++++++++++++++------------------- 1 file changed, 17 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 16313b973eaa41..74f20114297c62 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -5,6 +5,7 @@ */ #include +#include #include #include #include @@ -24,26 +25,24 @@ struct max98357a_priv { unsigned int sdmode_delay; }; -static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) +static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); if (!max98357a->sdmode) return 0; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - mdelay(max98357a->sdmode_delay); + if (event & SND_SOC_DAPM_POST_PMU) { + msleep(max98357a->sdmode_delay); gpiod_set_value(max98357a->sdmode, 1); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev_dbg(component->dev, "set sdmode to 1"); + } else if (event & SND_SOC_DAPM_PRE_PMD) { gpiod_set_value(max98357a->sdmode, 0); - break; + dev_dbg(component->dev, "set sdmode to 0"); } return 0; @@ -51,10 +50,14 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Speaker"), + SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0, + max98357a_sdmode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { - {"Speaker", NULL, "HiFi Playback"}, + {"SD_MODE", NULL, "HiFi Playback"}, + {"Speaker", NULL, "SD_MODE"}, }; static const struct snd_soc_component_driver max98357a_component_driver = { @@ -68,10 +71,6 @@ static const struct snd_soc_component_driver max98357a_component_driver = { .non_legacy_dai_naming = 1, }; -static const struct snd_soc_dai_ops max98357a_dai_ops = { - .trigger = max98357a_daiops_trigger, -}; - static struct snd_soc_dai_driver max98357a_dai_driver = { .name = "HiFi", .playback = { @@ -91,7 +90,6 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { .channels_min = 1, .channels_max = 2, }, - .ops = &max98357a_dai_ops, }; static int max98357a_platform_probe(struct platform_device *pdev) From 514de1c935d1670e0f162210f3794cf0be67c8a7 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Wed, 12 Feb 2020 13:55:17 +0800 Subject: [PATCH 071/415] ASoC: mediatek: mt8183-da7219: add speaker switch Da7219 and max98357a share the same I2S lines. When writing audio data to the I2S, both codecs generate sound. Da7219 already has a separate control "Headphone Switch". Adds a new control "Speakers Switch" for turning on/off max98357a. Userspace program can decide to turn on/off which codecs by different use cases. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200212124608.2.I5fa3fdca69dbb5d3dd5031c939b9b24095065a94@changeid Signed-off-by: Mark Brown --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c65493721e90d1..1626541cc0d68b 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -372,9 +372,28 @@ static struct snd_soc_codec_conf mt6358_codec_conf[] = { }, }; +static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), +}; + +static const +struct snd_soc_dapm_widget mt8183_da7219_max98357_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + +static const struct snd_soc_dapm_route mt8183_da7219_max98357_dapm_routes[] = { + {"Speakers", NULL, "Speaker"}, +}; + static struct snd_soc_card mt8183_da7219_max98357_card = { .name = "mt8183_da7219_max98357", .owner = THIS_MODULE, + .controls = mt8183_da7219_max98357_snd_controls, + .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls), + .dapm_widgets = mt8183_da7219_max98357_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets), + .dapm_routes = mt8183_da7219_max98357_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes), .dai_link = mt8183_da7219_max98357_dai_links, .num_links = ARRAY_SIZE(mt8183_da7219_max98357_dai_links), .aux_dev = &mt8183_da7219_max98357_headset_dev, From dd03907bf129b42e9e3203fdf405ea9873b28dd3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Feb 2020 12:14:37 +0900 Subject: [PATCH 072/415] ASoC: soc-pcm: call snd_soc_component_open/close() once Current soc_pcm_open() calls snd_soc_component_open() under loop. Thus, it needs to care about opened/not-yet-opened Component. But, if soc-component.c is handling it, soc-pcm.c don't need to care about it. This patch adds opened flag to soc-component.h, and simplify soc-pcm.c. This is one of prepare for cleanup soc-pcm-open() Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/874kvzcey1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 7 +++++-- sound/soc/soc-component.c | 35 ++++++++++++++++++++++++++++------- sound/soc/soc-pcm.c | 19 ++++++------------- 3 files changed, 39 insertions(+), 22 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 154d02fbbfed69..1866ecc8e94b25 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -147,8 +147,6 @@ struct snd_soc_component { unsigned int active; - unsigned int suspended:1; /* is in suspend PM state */ - struct list_head list; struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; @@ -182,6 +180,11 @@ struct snd_soc_component { struct dentry *debugfs_root; const char *debugfs_prefix; #endif + + /* bit field */ + unsigned int suspended:1; /* is in suspend PM state */ + unsigned int opened:1; + unsigned int module:1; }; #define for_each_component_dais(component, dai)\ diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 14e175cdeeb84b..ee00c09df5e791 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -297,34 +297,55 @@ EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); int snd_soc_component_module_get(struct snd_soc_component *component, int upon_open) { + if (component->module) + return 0; + if (component->driver->module_get_upon_open == !!upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; + component->module = 1; + return 0; } void snd_soc_component_module_put(struct snd_soc_component *component, int upon_open) { - if (component->driver->module_get_upon_open == !!upon_open) + if (component->module && + component->driver->module_get_upon_open == !!upon_open) module_put(component->dev->driver->owner); + + component->module = 0; } int snd_soc_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - if (component->driver->open) - return component->driver->open(component, substream); - return 0; + int ret = 0; + + if (!component->opened && + component->driver->open) + ret = component->driver->open(component, substream); + + if (ret == 0) + component->opened = 1; + + return ret; } int snd_soc_component_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - if (component->driver->close) - return component->driver->close(component, substream); - return 0; + int ret = 0; + + if (component->opened && + component->driver->close) + ret = component->driver->close(component, substream); + + component->opened = 0; + + return ret; } int snd_soc_component_prepare(struct snd_soc_component *component, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d53afb96b05bb7..ae94d8a869924c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -463,16 +463,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_max = min_not_zero(hw->rate_max, rate_max); } -static int soc_pcm_components_open(struct snd_pcm_substream *substream, - struct snd_soc_component **last) +static int soc_pcm_components_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; int i, ret = 0; for_each_rtd_components(rtd, i, component) { - *last = component; - ret = snd_soc_component_module_get_when_open(component); if (ret < 0) { dev_err(component->dev, @@ -489,21 +486,17 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, return ret; } } - *last = NULL; + return 0; } -static int soc_pcm_components_close(struct snd_pcm_substream *substream, - struct snd_soc_component *last) +static int soc_pcm_components_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; int i, r, ret = 0; for_each_rtd_components(rtd, i, component) { - if (component == last) - break; - r = snd_soc_component_close(component, substream); if (r < 0) ret = r; /* use last ret */ @@ -545,7 +538,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto out; } - ret = soc_pcm_components_open(substream, &component); + ret = soc_pcm_components_open(substream); if (ret < 0) goto component_err; @@ -642,7 +635,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_dai_shutdown(codec_dai, substream); component_err: - soc_pcm_components_close(substream, component); + soc_pcm_components_close(substream); snd_soc_dai_shutdown(cpu_dai, substream); out: @@ -696,7 +689,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) soc_rtd_shutdown(rtd, substream); - soc_pcm_components_close(substream, NULL); + soc_pcm_components_close(substream); snd_soc_dapm_stream_stop(rtd, substream->stream); From 62c86d1d5fd942c741791be94f670b99ffedfb5c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Feb 2020 12:14:41 +0900 Subject: [PATCH 073/415] ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() This patch moves soc_pcm_close() next to soc_pcm_open(). This is prepare for soc_pcm_open() cleanup. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/8736bjcexx.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 88 ++++++++++++++++++++++----------------------- 1 file changed, 44 insertions(+), 44 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ae94d8a869924c..8aa775e0eb0dce 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -507,6 +507,50 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream) return ret; } +/* + * Called by ALSA when a PCM substream is closed. Private data can be + * freed here. The cpu DAI, codec DAI, machine and components are also + * shutdown. + */ +static int soc_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i; + + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + snd_soc_runtime_deactivate(rtd, substream->stream); + + snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream); + + snd_soc_dai_shutdown(cpu_dai, substream); + + for_each_rtd_codec_dai(rtd, i, codec_dai) + snd_soc_dai_shutdown(codec_dai, substream); + + soc_rtd_shutdown(rtd, substream); + + soc_pcm_components_close(substream); + + snd_soc_dapm_stream_stop(rtd, substream->stream); + + mutex_unlock(&rtd->card->pcm_mutex); + + for_each_rtd_components(rtd, i, component) { + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + } + + for_each_rtd_components(rtd, i, component) + if (!component->active) + pinctrl_pm_select_sleep_state(component->dev); + + return 0; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -663,50 +707,6 @@ static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) */ } -/* - * Called by ALSA when a PCM substream is closed. Private data can be - * freed here. The cpu DAI, codec DAI, machine and components are also - * shutdown. - */ -static int soc_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int i; - - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); - - snd_soc_runtime_deactivate(rtd, substream->stream); - - snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream); - - snd_soc_dai_shutdown(cpu_dai, substream); - - for_each_rtd_codec_dai(rtd, i, codec_dai) - snd_soc_dai_shutdown(codec_dai, substream); - - soc_rtd_shutdown(rtd, substream); - - soc_pcm_components_close(substream); - - snd_soc_dapm_stream_stop(rtd, substream->stream); - - mutex_unlock(&rtd->card->pcm_mutex); - - for_each_rtd_components(rtd, i, component) { - pm_runtime_mark_last_busy(component->dev); - pm_runtime_put_autosuspend(component->dev); - } - - for_each_rtd_components(rtd, i, component) - if (!component->active) - pinctrl_pm_select_sleep_state(component->dev); - - return 0; -} - /* * Called by ALSA when the PCM substream is prepared, can set format, sample * rate, etc. This function is non atomic and can be called multiple times, From 5d9fa03e6c3514266fa94851ab1b6dd6e0595a13 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Feb 2020 12:14:45 +0900 Subject: [PATCH 074/415] ASoC: soc-pcm: tidyup soc_pcm_open() order soc_pcm_open() operation order is not good. At first, soc_pcm_open() operation order is 1) CPU DAI startup 2) Component open 3) Codec DAI startup 4) rtd startup But here, 2) will call try_module_get() if component has module_get_upon_open flags. This means 1) CPU DAI startup will be operated *before* its module was loaded. DAI should be called *after* Component. Second, soc_pcm_close() operation order is 1) CPU DAI shutdown 2) Codec DAI shutdown 3) rtd shutdown 4) Component close soc_pcm_open() and soc_pcm_close() are paired function, but, its operation order is unbalance. This patch tidyup soc_pcm_open() order to Component -> rtd -> DAI. This is one of prepare for cleanup soc-pcm-open() Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/871rr3cext.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 34 ++++++++++++++++------------------ 1 file changed, 16 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa775e0eb0dce..6630fadd6e098f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -574,25 +574,32 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + ret = soc_pcm_components_open(substream); + if (ret < 0) + goto component_err; + + ret = soc_rtd_startup(rtd, substream); + if (ret < 0) { + pr_err("ASoC: %s startup failed: %d\n", + rtd->dai_link->name, ret); + goto component_err; + } + /* startup the audio subsystem */ ret = snd_soc_dai_startup(cpu_dai, substream); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", cpu_dai->name, ret); - goto out; + goto cpu_dai_err; } - ret = soc_pcm_components_open(substream); - if (ret < 0) - goto component_err; - for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_startup(codec_dai, substream); if (ret < 0) { dev_err(codec_dai->dev, "ASoC: can't open codec %s: %d\n", codec_dai->name, ret); - goto codec_dai_err; + goto config_err; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -601,13 +608,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->rx_mask = 0; } - ret = soc_rtd_startup(rtd, substream); - if (ret < 0) { - pr_err("ASoC: %s startup failed: %d\n", - rtd->dai_link->name, ret); - goto codec_dai_err; - } - /* Dynamic PCM DAI links compat checks use dynamic capabilities */ if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) goto dynamic; @@ -672,17 +672,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - soc_rtd_shutdown(rtd, substream); - -codec_dai_err: for_each_rtd_codec_dai(rtd, i, codec_dai) snd_soc_dai_shutdown(codec_dai, substream); +cpu_dai_err: + snd_soc_dai_shutdown(cpu_dai, substream); + soc_rtd_shutdown(rtd, substream); component_err: soc_pcm_components_close(substream); - snd_soc_dai_shutdown(cpu_dai, substream); -out: mutex_unlock(&rtd->card->pcm_mutex); for_each_rtd_components(rtd, i, component) { From eadd54c75f1ef1566a6fe004787c028eb095f8b4 Mon Sep 17 00:00:00 2001 From: Dafna Hirschfeld Date: Mon, 27 Jan 2020 10:18:06 +0100 Subject: [PATCH 075/415] dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format. This was tested and verified with: make dt_binding_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml Signed-off-by: Dafna Hirschfeld Reviewed-by: Rob Herring Link: https://lore.kernel.org/r/20200127091806.11403-1-dafna.hirschfeld@collabora.com Signed-off-by: Mark Brown --- .../bindings/sound/google,cros-ec-codec.txt | 44 ------------- .../bindings/sound/google,cros-ec-codec.yaml | 62 +++++++++++++++++++ 2 files changed, 62 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt deleted file mode 100644 index 8ca52dcc557280..00000000000000 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt +++ /dev/null @@ -1,44 +0,0 @@ -Audio codec controlled by ChromeOS EC - -Google's ChromeOS EC codec is a digital mic codec provided by the -Embedded Controller (EC) and is controlled via a host-command interface. - -An EC codec node should only be found as a sub-node of the EC node (see -Documentation/devicetree/bindings/mfd/cros-ec.txt). - -Required properties: -- compatible: Must contain "google,cros-ec-codec" -- #sound-dai-cells: Should be 1. The cell specifies number of DAIs. - -Optional properties: -- reg: Pysical base address and length of shared memory region from EC. - It contains 3 unsigned 32-bit integer. The first 2 integers - combine to become an unsigned 64-bit physical address. The last - one integer is length of the shared memory. -- memory-region: Shared memory region to EC. A "shared-dma-pool". See - ../reserved-memory/reserved-memory.txt for details. - -Example: - -{ - ... - - reserved_mem: reserved_mem { - compatible = "shared-dma-pool"; - reg = <0 0x52800000 0 0x100000>; - no-map; - }; -} - -cros-ec@0 { - compatible = "google,cros-ec-spi"; - - ... - - cros_ec_codec: ec-codec { - compatible = "google,cros-ec-codec"; - #sound-dai-cells = <1>; - reg = <0x0 0x10500000 0x80000>; - memory-region = <&reserved_mem>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml new file mode 100644 index 00000000000000..94a85d0cbf433a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,cros-ec-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio codec controlled by ChromeOS EC + +maintainers: + - Cheng-Yi Chiang + +description: | + Google's ChromeOS EC codec is a digital mic codec provided by the + Embedded Controller (EC) and is controlled via a host-command interface. + An EC codec node should only be found as a sub-node of the EC node (see + Documentation/devicetree/bindings/mfd/cros-ec.txt). + +properties: + compatible: + const: google,cros-ec-codec + + "#sound-dai-cells": + const: 1 + + reg: + items: + - description: | + Physical base address and length of shared memory region from EC. + It contains 3 unsigned 32-bit integer. The first 2 integers + combine to become an unsigned 64-bit physical address. + The last one integer is the length of the shared memory. + + memory-region: + $ref: '/schemas/types.yaml#/definitions/phandle' + description: | + Shared memory region to EC. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + +required: + - compatible + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + reserved_mem: reserved_mem { + compatible = "shared-dma-pool"; + reg = <0 0x52800000 0 0x100000>; + no-map; + }; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + #address-cells = <2>; + #size-cells = <1>; + cros_ec_codec: ec-codec { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; + }; From ce0c97f8a2936d05951da7093b881c8eaaee72e0 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 12 Feb 2020 15:56:48 +0100 Subject: [PATCH 076/415] ASoC: Fix SND_SOC_ALL_CODECS imply SPI fallout Fixes for CONFIG_SPI=n: WARNING: unmet direct dependencies detected for REGMAP_SPI Depends on [n]: SPI [=n] Selected by [m]: - SND_SOC_ADAU1781_SPI [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] - SND_SOC_ADAU1977_SPI [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] ERROR: "spi_async" [...] undefined! ERROR: "spi_get_device_id" [...] undefined! ERROR: "__spi_register_driver" [...] undefined! ERROR: "spi_setup" [...] undefined! ERROR: "spi_sync" [...] undefined! ERROR: "spi_write_then_read" [...] undefined! Reported-by: Randy Dunlap Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Signed-off-by: Geert Uytterhoeven Acked-by: Randy Dunlap # build-tested Link: https://lore.kernel.org/r/20200212145650.4602-2-geert@linux-m68k.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 68 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 63 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7a14b1c416b55e..c2fb985de8c4e0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -327,12 +327,14 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1836 tristate + depends on SPI_MASTER config SND_SOC_AD193X tristate config SND_SOC_AD193X_SPI tristate + depends on SPI_MASTER select SND_SOC_AD193X config SND_SOC_AD193X_I2C @@ -390,6 +392,7 @@ config SND_SOC_ADAU1781_I2C config SND_SOC_ADAU1781_SPI tristate + depends on SPI_MASTER select SND_SOC_ADAU1781 select REGMAP_SPI @@ -398,6 +401,7 @@ config SND_SOC_ADAU1977 config SND_SOC_ADAU1977_SPI tristate + depends on SPI_MASTER select SND_SOC_ADAU1977 select REGMAP_SPI @@ -441,6 +445,7 @@ config SND_SOC_ADAV80X config SND_SOC_ADAV801 tristate + depends on SPI_MASTER select SND_SOC_ADAV80X config SND_SOC_ADAV803 @@ -498,6 +503,7 @@ config SND_SOC_ALC5623 config SND_SOC_ALC5632 tristate + depends on I2C config SND_SOC_BD28623 tristate "ROHM BD28623 CODEC" @@ -698,6 +704,7 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate + depends on I2C config SND_SOC_DA7213 tristate "Dialog DA7213 CODEC" @@ -705,15 +712,19 @@ config SND_SOC_DA7213 config SND_SOC_DA7218 tristate + depends on I2C config SND_SOC_DA7219 tristate + depends on I2C config SND_SOC_DA732X tristate + depends on I2C config SND_SOC_DA9055 tristate + depends on I2C config SND_SOC_DMIC tristate "Generic Digital Microphone CODEC" @@ -773,9 +784,11 @@ config SND_SOC_INNO_RK3036 config SND_SOC_ISABELLE tristate + depends on I2C config SND_SOC_LM49453 tristate + depends on I2C config SND_SOC_LOCHNAGAR_SC tristate "Lochnagar Sound Card" @@ -802,17 +815,20 @@ config SND_SOC_MAX98088 depends on I2C config SND_SOC_MAX98090 - tristate + tristate + depends on I2C config SND_SOC_MAX98095 - tristate + tristate + depends on I2C config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" depends on GPIOLIB config SND_SOC_MAX98371 - tristate + tristate + depends on I2C config SND_SOC_MAX98504 tristate "Maxim MAX98504 speaker amplifier" @@ -823,10 +839,12 @@ config SND_SOC_MAX9867 depends on I2C config SND_SOC_MAX98925 - tristate + tristate + depends on I2C config SND_SOC_MAX98926 tristate + depends on I2C config SND_SOC_MAX98927 tristate "Maxim Integrated MAX98927 Speaker Amplifier" @@ -838,6 +856,7 @@ config SND_SOC_MAX98373 config SND_SOC_MAX9850 tristate + depends on I2C config SND_SOC_MAX9860 tristate "Maxim MAX9860 Mono Audio Voice Codec" @@ -1016,26 +1035,32 @@ config SND_SOC_RT298 config SND_SOC_RT1011 tristate + depends on I2C config SND_SOC_RT1015 tristate + depends on I2C config SND_SOC_RT1305 tristate + depends on I2C config SND_SOC_RT1308 tristate + depends on I2C config SND_SOC_RT1308_SDW tristate "Realtek RT1308 Codec - SDW" - depends on SOUNDWIRE + depends on I2C && SOUNDWIRE select REGMAP_SOUNDWIRE config SND_SOC_RT5514 tristate + depends on I2C config SND_SOC_RT5514_SPI tristate + depends on SPI_MASTER config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors @@ -1051,30 +1076,39 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate + depends on I2C config SND_SOC_RT5645 tristate + depends on I2C config SND_SOC_RT5651 tristate + depends on I2C config SND_SOC_RT5659 tristate + depends on I2C config SND_SOC_RT5660 tristate + depends on I2C config SND_SOC_RT5663 tristate + depends on I2C config SND_SOC_RT5665 tristate + depends on I2C config SND_SOC_RT5668 tristate + depends on I2C config SND_SOC_RT5670 tristate + depends on I2C config SND_SOC_RT5677 tristate @@ -1087,6 +1121,7 @@ config SND_SOC_RT5677_SPI config SND_SOC_RT5682 tristate + depends on I2C config SND_SOC_RT700 tristate @@ -1154,6 +1189,7 @@ config SND_SOC_SSM2305 config SND_SOC_SSM2518 tristate + depends on I2C config SND_SOC_SSM2602 tristate @@ -1185,6 +1221,7 @@ config SND_SOC_STA350 config SND_SOC_STA529 tristate + depends on I2C config SND_SOC_STAC9766 tristate @@ -1282,6 +1319,7 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate + depends on I2C config SND_SOC_TS3A227E tristate "TI Headset/Mic detect and keypress chip" @@ -1348,18 +1386,23 @@ config SND_SOC_WL1273 config SND_SOC_WM0010 tristate + depends on SPI_MASTER config SND_SOC_WM1250_EV1 tristate + depends on I2C config SND_SOC_WM2000 tristate + depends on I2C config SND_SOC_WM2200 tristate + depends on I2C config SND_SOC_WM5100 tristate + depends on I2C config SND_SOC_WM5102 tristate @@ -1462,9 +1505,11 @@ config SND_SOC_WM8904 config SND_SOC_WM8940 tristate + depends on I2C config SND_SOC_WM8955 tristate + depends on I2C config SND_SOC_WM8960 tristate "Wolfson Microelectronics WM8960 CODEC" @@ -1472,6 +1517,7 @@ config SND_SOC_WM8960 config SND_SOC_WM8961 tristate + depends on I2C config SND_SOC_WM8962 tristate "Wolfson Microelectronics WM8962 CODEC" @@ -1479,6 +1525,7 @@ config SND_SOC_WM8962 config SND_SOC_WM8971 tristate + depends on I2C config SND_SOC_WM8974 tristate "Wolfson Microelectronics WM8974 codec" @@ -1490,6 +1537,7 @@ config SND_SOC_WM8978 config SND_SOC_WM8983 tristate + depends on I2C config SND_SOC_WM8985 tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver" @@ -1500,12 +1548,15 @@ config SND_SOC_WM8988 config SND_SOC_WM8990 tristate + depends on I2C config SND_SOC_WM8991 tristate + depends on I2C config SND_SOC_WM8993 tristate + depends on I2C config SND_SOC_WM8994 tristate @@ -1515,6 +1566,7 @@ config SND_SOC_WM8995 config SND_SOC_WM8996 tristate + depends on I2C config SND_SOC_WM8997 tristate @@ -1528,6 +1580,7 @@ config SND_SOC_WM9081 config SND_SOC_WM9090 tristate + depends on I2C config SND_SOC_WM9705 tristate @@ -1561,6 +1614,7 @@ config SND_SOC_ZX_AUD96P22 # Amp config SND_SOC_LM4857 tristate + depends on I2C config SND_SOC_MAX9759 tristate "Maxim MAX9759 speaker Amplifier" @@ -1568,15 +1622,18 @@ config SND_SOC_MAX9759 config SND_SOC_MAX9768 tristate + depends on I2C config SND_SOC_MAX9877 tristate + depends on I2C config SND_SOC_MC13783 tristate config SND_SOC_ML26124 tristate + depends on I2C config SND_SOC_MT6351 tristate "MediaTek MT6351 Codec" @@ -1614,6 +1671,7 @@ config SND_SOC_NAU8824 config SND_SOC_NAU8825 tristate + depends on I2C config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" From 1d0158f547e0dbefa9e18930e93f270ab0309707 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 12 Feb 2020 15:56:49 +0100 Subject: [PATCH 077/415] ASoC: Fix SND_SOC_ALL_CODECS imply I2C fallout MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes for CONFIG_I2C=n: WARNING: unmet direct dependencies detected for REGMAP_I2C Depends on [n]: I2C [=n] Selected by [m]: - SND_SOC_ADAU1781_I2C [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] - SND_SOC_ADAU1977_I2C [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] - SND_SOC_RT5677 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] sound/soc/codecs/...: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] drivers/base/regmap/regmap-i2c.c: In function ‘regmap_smbus_byte_reg_read’: drivers/base/regmap/regmap-i2c.c:25:8: error: implicit declaration of function ‘i2c_smbus_read_byte_data’; did you mean ‘i2c_set_adapdata’? [-Werror=implicit-function-declaration] Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/20200212145650.4602-3-geert@linux-m68k.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2fb985de8c4e0..3ef804d07deea4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -339,6 +339,7 @@ config SND_SOC_AD193X_SPI config SND_SOC_AD193X_I2C tristate + depends on I2C select SND_SOC_AD193X config SND_SOC_AD1980 @@ -353,6 +354,7 @@ config SND_SOC_ADAU_UTILS config SND_SOC_ADAU1373 tristate + depends on I2C select SND_SOC_ADAU_UTILS config SND_SOC_ADAU1701 @@ -387,6 +389,7 @@ config SND_SOC_ADAU1781 config SND_SOC_ADAU1781_I2C tristate + depends on I2C select SND_SOC_ADAU1781 select REGMAP_I2C @@ -407,6 +410,7 @@ config SND_SOC_ADAU1977_SPI config SND_SOC_ADAU1977_I2C tristate + depends on I2C select SND_SOC_ADAU1977 select REGMAP_I2C @@ -450,6 +454,7 @@ config SND_SOC_ADAV801 config SND_SOC_ADAV803 tristate + depends on I2C select SND_SOC_ADAV80X config SND_SOC_ADS117X @@ -471,6 +476,7 @@ config SND_SOC_AK4458 config SND_SOC_AK4535 tristate + depends on I2C config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" @@ -481,6 +487,7 @@ config SND_SOC_AK4613 config SND_SOC_AK4641 tristate + depends on I2C config SND_SOC_AK4642 tristate "AKM AK4642 CODEC" @@ -488,6 +495,7 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate + depends on I2C config SND_SOC_AK5386 tristate "AKM AK5638 CODEC" @@ -1112,6 +1120,7 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate + depends on I2C select REGMAP_I2C select REGMAP_IRQ From d8dd3f92a6ba11f9046df48765e6f12ad70a3946 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 12 Feb 2020 15:56:50 +0100 Subject: [PATCH 078/415] ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout Fixes for missing miscellaneous support: ERROR: "abx500_get_register_interruptible" [...] undefined! ERROR: "abx500_set_register_interruptible" [...] undefined! ERROR: "arizona_free_irq" [...] undefined! ERROR: "arizona_request_irq" [...] undefined! ERROR: "arizona_set_irq_wake" [...] undefined! ERROR: "mc13xxx_reg_rmw" [...] undefined! ERROR: "mc13xxx_reg_write" [...] undefined! ERROR: "snd_soc_free_ac97_component" [...] undefined! ERROR: "snd_soc_new_ac97_component" [...] undefined! Reported-by: Randy Dunlap Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/20200212145650.4602-4-geert@linux-m68k.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3ef804d07deea4..d957fd6980b10a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -319,6 +319,7 @@ config SND_SOC_WM_ADSP config SND_SOC_AB8500_CODEC tristate + depends on ABX500_CORE config SND_SOC_AC97_CODEC tristate "Build generic ASoC AC97 CODEC driver" @@ -343,8 +344,9 @@ config SND_SOC_AD193X_I2C select SND_SOC_AD193X config SND_SOC_AD1980 - select REGMAP_AC97 tristate + depends on SND_SOC_AC97_BUS + select REGMAP_AC97 config SND_SOC_AD73311 tristate @@ -646,6 +648,7 @@ config SND_SOC_CS47L15 config SND_SOC_CS47L24 tristate + depends on MFD_CS47L24 config SND_SOC_CS47L35 tristate @@ -1234,6 +1237,7 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate + depends on SND_SOC_AC97_BUS config SND_SOC_STI_SAS tristate "codec Audio support for STI SAS codec" @@ -1415,9 +1419,11 @@ config SND_SOC_WM5100 config SND_SOC_WM5102 tristate + depends on MFD_WM5102 config SND_SOC_WM5110 tristate + depends on MFD_WM5110 config SND_SOC_WM8350 tristate @@ -1579,9 +1585,11 @@ config SND_SOC_WM8996 config SND_SOC_WM8997 tristate + depends on MFD_WM8997 config SND_SOC_WM8998 tristate + depends on MFD_WM8998 config SND_SOC_WM9081 tristate @@ -1639,6 +1647,7 @@ config SND_SOC_MAX9877 config SND_SOC_MC13783 tristate + depends on MFD_MC13XXX config SND_SOC_ML26124 tristate From ec7ba9e1500b0af5bf30b4e56bfaaf3a88850bbf Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Thu, 13 Feb 2020 11:27:25 +0800 Subject: [PATCH 079/415] ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1 DL2 and UL1 are for BTSCO. Provides only 16-bit, mono, 8kHz and 16kHz to userspace. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200213112003.1.Ie5aedb9d34ebfc7f05ceb382bfe346c45331cd63@changeid Signed-off-by: Mark Brown --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 42 +++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 1626541cc0d68b..b52ffed882a748 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -116,6 +116,46 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static int +mt8183_da7219_max98357_bt_sco_startup( + struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 8000, 16000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const unsigned int channels[] = { + 1, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + return 0; +} + +static const struct snd_soc_ops mt8183_da7219_max98357_bt_sco_ops = { + .startup = mt8183_da7219_max98357_bt_sco_startup, +}; + /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), @@ -222,6 +262,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8183_da7219_max98357_bt_sco_ops, SND_SOC_DAILINK_REG(playback2), }, { @@ -240,6 +281,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8183_da7219_max98357_bt_sco_ops, SND_SOC_DAILINK_REG(capture1), }, { From 8726ee6148fe24e2b29d4a961ad95c4ff8025d1d Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Thu, 13 Feb 2020 11:27:26 +0800 Subject: [PATCH 080/415] ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed 1. Switch TDM GPIO pins according to playback on or off. 2. Pull TDM GPIO pins down when probed to avoid current leakage. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200213112003.2.I1d568b0c99742c6e755d051aadfd52e4be3cc0a5@changeid Signed-off-by: Mark Brown --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 104 +++++++++++++++++- 1 file changed, 98 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index b52ffed882a748..d7685916a5cb39 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -18,6 +18,22 @@ static struct snd_soc_jack headset_jack; +enum PINCTRL_PIN_STATE { + PIN_STATE_DEFAULT = 0, + PIN_TDM_OUT_ON, + PIN_TDM_OUT_OFF, + PIN_STATE_MAX +}; + +static const char * const mt8183_pin_str[PIN_STATE_MAX] = { + "default", "aud_tdm_out_on", "aud_tdm_out_off", +}; + +struct mt8183_da7219_max98357_priv { + struct pinctrl *pinctrl; + struct pinctrl_state *pin_states[PIN_STATE_MAX]; +}; + static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -244,6 +260,47 @@ SND_SOC_DAILINK_DEFS(tdm, DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); +static int mt8183_da7219_tdm_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON])) + return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]); + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_ON]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + + return ret; +} + +static void mt8183_da7219_tdm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) + return; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); +} + +static struct snd_soc_ops mt8183_da7219_tdm_ops = { + .startup = mt8183_da7219_tdm_startup, + .shutdown = mt8183_da7219_tdm_shutdown, +}; + static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { /* FE */ { @@ -395,6 +452,8 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_da7219_tdm_ops, SND_SOC_DAILINK_REG(tdm), }, }; @@ -470,7 +529,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct snd_soc_card *card = &mt8183_da7219_max98357_card; struct device_node *platform_node; struct snd_soc_dai_link *dai_link; - struct pinctrl *default_pins; + struct mt8183_da7219_max98357_priv *priv; int ret, i; card->dev = &pdev->dev; @@ -504,12 +563,45 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return ret; } - default_pins = - devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); - if (IS_ERR(default_pins)) { - dev_err(&pdev->dev, "%s set pins failed\n", + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + priv->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(priv->pinctrl)) { + dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n", __func__); - return PTR_ERR(default_pins); + return PTR_ERR(priv->pinctrl); + } + + for (i = 0; i < PIN_STATE_MAX; i++) { + priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, + mt8183_pin_str[i]); + if (IS_ERR(priv->pin_states[i])) { + ret = PTR_ERR(priv->pin_states[i]); + dev_info(&pdev->dev, "%s Can't find pin state %s %d\n", + __func__, mt8183_pin_str[i], ret); + } + } + + if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); + } + + if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_DEFAULT]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); } return ret; From 195a6431710543ace009ff8e32dcf1e087506199 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Thu, 13 Feb 2020 11:27:27 +0800 Subject: [PATCH 081/415] ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out Supports TDM out and 8ch I2S out. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200213112003.3.I30f0b8c87d5ec2a0e5f1b0fabf0a8ccef374f5ea@changeid Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index d7685916a5cb39..c7f766f24e440c 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -450,6 +450,9 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { { .name = "TDM", .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_IF | + SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, .ignore_suspend = 1, .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, From ff0035e4c22371a29e4e0d4a07cdce5726fe50aa Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Thu, 13 Feb 2020 11:27:28 +0800 Subject: [PATCH 082/415] ASoC: mediatek: mt8183-da7219: apply some refactors 1. Moves headset jack to card-specific storage. 2. Removes trailing blank line. 3. Moves card registration to the end of probe. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200213112003.4.Ia542007f51d3de753a9e0a83135ee074581dbf71@changeid Signed-off-by: Mark Brown --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c7f766f24e440c..c0c85972cfb749 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -16,8 +16,6 @@ #include "../../codecs/da7219-aad.h" #include "../../codecs/da7219.h" -static struct snd_soc_jack headset_jack; - enum PINCTRL_PIN_STATE { PIN_STATE_DEFAULT = 0, PIN_TDM_OUT_ON, @@ -32,6 +30,7 @@ static const char * const mt8183_pin_str[PIN_STATE_MAX] = { struct mt8183_da7219_max98357_priv { struct pinctrl *pinctrl; struct pinctrl_state *pin_states[PIN_STATE_MAX]; + struct snd_soc_jack headset_jack; }; static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, @@ -510,6 +509,8 @@ static int mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) { int ret; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(component->card); /* Enable Headset and 4 Buttons Jack detection */ ret = snd_soc_card_jack_new(&mt8183_da7219_max98357_card, @@ -517,12 +518,12 @@ mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, - &headset_jack, + &priv->headset_jack, NULL, 0); if (ret) return ret; - da7219_aad_jack_det(component, &headset_jack); + da7219_aad_jack_det(component, &priv->headset_jack); return ret; } @@ -559,13 +560,6 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return -EINVAL; } - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", - __func__, ret); - return ret; - } - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; @@ -607,7 +601,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) __func__, ret); } - return ret; + return devm_snd_soc_register_card(&pdev->dev, card); } #ifdef CONFIG_OF @@ -634,4 +628,3 @@ MODULE_DESCRIPTION("MT8183-DA7219-MAX98357 ALSA SoC machine driver"); MODULE_AUTHOR("Shunli Wang "); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("mt8183_da7219_max98357 soc card"); - From 0c48a65394ab6d2b4afde3fbe840dbb05a52d929 Mon Sep 17 00:00:00 2001 From: "derek.fang" Date: Thu, 13 Feb 2020 15:05:10 +0800 Subject: [PATCH 083/415] ASoC: rt5682: Enable PLL2 function Enable RT5682 PLL2 function to implement the more complex frequency conversion. Signed-off-by: derek.fang Link: https://lore.kernel.org/r/1581577510-1807-1-git-send-email-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 211 ++++++++++++++++++++++++++++++-------- sound/soc/codecs/rt5682.h | 39 ++++++- 2 files changed, 203 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 82a6366201314e..9fbb3862f8d70d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -64,9 +64,9 @@ struct rt5682_priv { int bclk[RT5682_AIFS]; int master[RT5682_AIFS]; - int pll_src; - int pll_in; - int pll_out; + int pll_src[RT5682_PLLS]; + int pll_in[RT5682_PLLS]; + int pll_out[RT5682_PLLS]; int jack_type; }; @@ -75,6 +75,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, + {RT5682_PLL2_INTERNAL, 0x8266}, }; static const struct reg_default rt5682_reg[] = { @@ -222,7 +223,7 @@ static const struct reg_default rt5682_reg[] = { {0x0148, 0x0000}, {0x0149, 0x0000}, {0x0150, 0x79a1}, - {0x0151, 0x0000}, + {0x0156, 0xaaaa}, {0x0160, 0x4ec0}, {0x0161, 0x0080}, {0x0162, 0x0200}, @@ -928,10 +929,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component, RT5682_PWR_VREF2 | RT5682_PWR_MB, RT5682_PWR_VREF2 | RT5682_PWR_MB); snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); usleep_range(15000, 20000); snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, RT5682_PWR_CBJ); @@ -1298,6 +1299,21 @@ static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, return 0; } +static int is_sys_clk_from_pll2(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5682_GLB_CLK); + val &= RT5682_SCLK_SRC_MASK; + if (val == RT5682_SCLK_SRC_PLL2) + return 1; + else + return 0; +} + static int is_using_asrc(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_widget *sink) { @@ -1612,9 +1628,11 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, - 0, NULL, 0), + 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, + NULL, 0), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -1796,7 +1814,11 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { /*PLL*/ {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"ADC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2}, + {"ADC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2}, {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2}, + {"DAC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2}, /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, @@ -2053,8 +2075,10 @@ static int rt5682_hw_params(struct snd_pcm_substream *substream, RT5682_I2S1_DL_MASK, len_1); if (rt5682->master[RT5682_AIF1]) { snd_soc_component_update_bits(component, - RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK, - pre_div << RT5682_I2S_M_DIV_SFT); + RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK | + RT5682_I2S_CLK_SRC_MASK, + pre_div << RT5682_I2S_M_DIV_SFT | + (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT); } if (params_channels(params) == 1) /* mono mode */ snd_soc_component_update_bits(component, @@ -2227,61 +2251,157 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, unsigned int freq_out) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - struct rl6231_pll_code pll_code; + struct rl6231_pll_code pll_code, pll2f_code, pll2b_code; + unsigned int pll2_fout1; int ret; - if (source == rt5682->pll_src && freq_in == rt5682->pll_in && - freq_out == rt5682->pll_out) + if (source == rt5682->pll_src[pll_id] && + freq_in == rt5682->pll_in[pll_id] && + freq_out == rt5682->pll_out[pll_id]) return 0; if (!freq_in || !freq_out) { dev_dbg(component->dev, "PLL disabled\n"); - rt5682->pll_in = 0; - rt5682->pll_out = 0; + rt5682->pll_in[pll_id] = 0; + rt5682->pll_out[pll_id] = 0; snd_soc_component_update_bits(component, RT5682_GLB_CLK, RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK); return 0; } - switch (source) { - case RT5682_PLL1_S_MCLK: - snd_soc_component_update_bits(component, RT5682_GLB_CLK, - RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK); - break; - case RT5682_PLL1_S_BCLK1: - snd_soc_component_update_bits(component, RT5682_GLB_CLK, - RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1); - break; - default: - dev_err(component->dev, "Unknown PLL Source %d\n", source); - return -EINVAL; - } + if (pll_id == RT5682_PLL2) { + switch (source) { + case RT5682_PLL2_S_MCLK: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL2_SRC_MASK, + RT5682_PLL2_SRC_MCLK); + break; + default: + dev_err(component->dev, "Unknown PLL2 Source %d\n", + source); + return -EINVAL; + } - ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); - if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); - return ret; + /** + * PLL2 concatenates 2 PLL units. + * We suggest the Fout of the front PLL is 3.84MHz. + */ + pll2_fout1 = 3840000; + ret = rl6231_pll_calc(freq_in, pll2_fout1, &pll2f_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + freq_in); + return ret; + } + dev_dbg(component->dev, "PLL2F: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n", + freq_in, pll2_fout1, + pll2f_code.m_bp, + (pll2f_code.m_bp ? 0 : pll2f_code.m_code), + pll2f_code.n_code, pll2f_code.k_code); + + ret = rl6231_pll_calc(pll2_fout1, freq_out, &pll2b_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + pll2_fout1); + return ret; + } + dev_dbg(component->dev, "PLL2B: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n", + pll2_fout1, freq_out, + pll2b_code.m_bp, + (pll2b_code.m_bp ? 0 : pll2b_code.m_code), + pll2b_code.n_code, pll2b_code.k_code); + + snd_soc_component_write(component, RT5682_PLL2_CTRL_1, + pll2f_code.k_code << RT5682_PLL2F_K_SFT | + pll2b_code.k_code << RT5682_PLL2B_K_SFT | + pll2b_code.m_code); + snd_soc_component_write(component, RT5682_PLL2_CTRL_2, + pll2f_code.m_code << RT5682_PLL2F_M_SFT | + pll2b_code.n_code); + snd_soc_component_write(component, RT5682_PLL2_CTRL_3, + pll2f_code.n_code << RT5682_PLL2F_N_SFT); + snd_soc_component_update_bits(component, RT5682_PLL2_CTRL_4, + RT5682_PLL2B_M_BP_MASK | RT5682_PLL2F_M_BP_MASK | 0xf, + (pll2b_code.m_bp ? 1 : 0) << RT5682_PLL2B_M_BP_SFT | + (pll2f_code.m_bp ? 1 : 0) << RT5682_PLL2F_M_BP_SFT | + 0xf); + } else { + switch (source) { + case RT5682_PLL1_S_MCLK: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK, + RT5682_PLL1_SRC_MCLK); + break; + case RT5682_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK, + RT5682_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL1 Source %d\n", + source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5682_PLL_CTRL_1, + pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5682_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | + pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); } - dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", - pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), - pll_code.n_code, pll_code.k_code); + rt5682->pll_in[pll_id] = freq_in; + rt5682->pll_out[pll_id] = freq_out; + rt5682->pll_src[pll_id] = source; + + return 0; +} + +static int rt5682_set_bclk1_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - snd_soc_component_write(component, RT5682_PLL_CTRL_1, - pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); - snd_soc_component_write(component, RT5682_PLL_CTRL_2, - (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | - pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); + rt5682->bclk[dai->id] = ratio; - rt5682->pll_in = freq_in; - rt5682->pll_out = freq_out; - rt5682->pll_src = source; + switch (ratio) { + case 256: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_256); + break; + case 128: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_128); + break; + case 64: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_32); + break; + default: + dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio); + return -EINVAL; + } return 0; } -static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +static int rt5682_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio) { struct snd_soc_component *component = dai->component; struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); @@ -2300,7 +2420,7 @@ static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) RT5682_I2S2_BCLK_MS2_32); break; default: - dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio); return -EINVAL; } @@ -2389,12 +2509,13 @@ static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = { .hw_params = rt5682_hw_params, .set_fmt = rt5682_set_dai_fmt, .set_tdm_slot = rt5682_set_tdm_slot, + .set_bclk_ratio = rt5682_set_bclk1_ratio, }; static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { .hw_params = rt5682_hw_params, .set_fmt = rt5682_set_dai_fmt, - .set_bclk_ratio = rt5682_set_bclk_ratio, + .set_bclk_ratio = rt5682_set_bclk2_ratio, }; static struct snd_soc_dai_driver rt5682_dai[] = { diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 4d3a8c41546e7b..465c99b7f906d7 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -177,7 +177,7 @@ #define RT5682_TEST_MODE_CTRL_4 0x0148 #define RT5682_TEST_MODE_CTRL_5 0x0149 #define RT5682_PLL1_INTERNAL 0x0150 -#define RT5682_PLL2_INTERNAL 0x0151 +#define RT5682_PLL2_INTERNAL 0x0156 #define RT5682_STO_NG2_CTRL_1 0x0160 #define RT5682_STO_NG2_CTRL_2 0x0161 #define RT5682_STO_NG2_CTRL_3 0x0162 @@ -738,7 +738,7 @@ #define RT5682_ADC_OSR_D_24 (0x7 << 12) #define RT5682_ADC_OSR_D_32 (0x8 << 12) #define RT5682_ADC_OSR_D_48 (0x9 << 12) -#define RT5682_I2S_M_DIV_MASK (0xf << 12) +#define RT5682_I2S_M_DIV_MASK (0xf << 8) #define RT5682_I2S_M_DIV_SFT 8 #define RT5682_I2S_M_D_1 (0x0 << 8) #define RT5682_I2S_M_D_2 (0x1 << 8) @@ -820,6 +820,12 @@ #define RT5682_TDM_DF_PCM_B (0x3 << 11) #define RT5682_TDM_DF_PCM_A_N (0x6 << 11) #define RT5682_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682_TDM_BCLK_MS1_MASK (0x3 << 9) +#define RT5682_TDM_BCLK_MS1_SFT 9 +#define RT5682_TDM_BCLK_MS1_32 (0x0 << 9) +#define RT5682_TDM_BCLK_MS1_64 (0x1 << 9) +#define RT5682_TDM_BCLK_MS1_128 (0x2 << 9) +#define RT5682_TDM_BCLK_MS1_256 (0x3 << 9) #define RT5682_TDM_CL_MASK (0x3 << 4) #define RT5682_TDM_CL_16 (0x0 << 4) #define RT5682_TDM_CL_20 (0x1 << 4) @@ -1049,6 +1055,28 @@ #define RT5682_PWR_CLK1M_PD (0x0 << 8) #define RT5682_PWR_CLK1M_PU (0x1 << 8) +/* PLL2 M/N/K Code Control 1 (0x009b) */ +#define RT5682_PLL2F_K_MASK (0x1f << 8) +#define RT5682_PLL2F_K_SFT 8 +#define RT5682_PLL2B_K_MASK (0xf << 4) +#define RT5682_PLL2B_K_SFT 4 +#define RT5682_PLL2B_M_MASK (0xf << 0) + +/* PLL2 M/N/K Code Control 2 (0x009c) */ +#define RT5682_PLL2F_M_MASK (0x3f << 8) +#define RT5682_PLL2F_M_SFT 8 +#define RT5682_PLL2B_N_MASK (0x3f << 0) + +/* PLL2 M/N/K Code Control 2 (0x009d) */ +#define RT5682_PLL2F_N_MASK (0x7f << 8) +#define RT5682_PLL2F_N_SFT 8 + +/* PLL2 M/N/K Code Control 2 (0x009e) */ +#define RT5682_PLL2B_M_BP_MASK (0x1 << 11) +#define RT5682_PLL2B_M_BP_SFT 11 +#define RT5682_PLL2F_M_BP_MASK (0x1 << 7) +#define RT5682_PLL2F_M_BP_SFT 7 + /* RC Clock Control (0x009f) */ #define RT5682_POW_IRQ (0x1 << 15) #define RT5682_POW_JDH (0x1 << 14) @@ -1315,6 +1343,13 @@ enum { RT5682_PLL1_S_MCLK, RT5682_PLL1_S_BCLK1, RT5682_PLL1_S_RCCLK, + RT5682_PLL2_S_MCLK, +}; + +enum { + RT5682_PLL1, + RT5682_PLL2, + RT5682_PLLS, }; enum { From 1dfa5a5ab34560fd9647083f623d19705be2e706 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:51 +0100 Subject: [PATCH 084/415] ASoC: core: allow a dt node to provide several components At the moment, querying the dai_name will stop of the first component matching the dt node. This does not allow a device (single dt node) to provide several ASoC components which could then be used through DT. This change let the search go on if the xlate function of the component returns an error, giving the possibility to another component to match and return the dai_name. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-2-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068d809c349a2f..03b87427faa774 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3102,6 +3102,14 @@ int snd_soc_get_dai_name(struct of_phandle_args *args, *dai_name = dai->driver->name; if (!*dai_name) *dai_name = pos->name; + } else if (ret) { + /* + * if another error than ENOTSUPP is returned go on and + * check if another component is provided with the same + * node. This may happen if a device provides several + * components + */ + continue; } break; From 9c29fd9bdf92900dc0cc5c2d8f58951a7bdc0f41 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:52 +0100 Subject: [PATCH 085/415] ASoC: meson: g12a: extract codec-to-codec utils The hdmi routing mechanism used on g12a hdmi is also used: * other Amlogic SoC types * for the internal DAC path Each of these codec glues are slightly different but the idea behind it remains the same. This change extract some helper functions from the g12a-tohdmitx driver to make them available for other Amlogic codecs. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-3-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 4 + sound/soc/meson/Makefile | 2 + sound/soc/meson/g12a-tohdmitx.c | 219 ++++++----------------------- sound/soc/meson/meson-codec-glue.c | 149 ++++++++++++++++++++ sound/soc/meson/meson-codec-glue.h | 32 +++++ 5 files changed, 230 insertions(+), 176 deletions(-) create mode 100644 sound/soc/meson/meson-codec-glue.c create mode 100644 sound/soc/meson/meson-codec-glue.h diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 2e3676147ceafa..ee6d53949d4536 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -85,9 +85,13 @@ config SND_MESON_AXG_PDM Select Y or M to add support for PDM input embedded in the Amlogic AXG SoC family +config SND_MESON_CODEC_GLUE + tristate + config SND_MESON_G12A_TOHDMITX tristate "Amlogic G12A To HDMI TX Control Support" select REGMAP_MMIO + select SND_MESON_CODEC_GLUE imply SND_SOC_HDMI_CODEC help Select Y or M to add support for HDMI audio on the g12a SoC diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 1a8b1470ed843e..529a807b3f376f 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -11,6 +11,7 @@ snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifin-objs := axg-spdifin.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o snd-soc-meson-axg-pdm-objs := axg-pdm.o +snd-soc-meson-codec-glue-objs := meson-codec-glue.o snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o @@ -24,4 +25,5 @@ obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o +obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 9cfbd343a00c86..f8853f2fba0873 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -12,112 +12,51 @@ #include #include +#include "meson-codec-glue.h" #define G12A_TOHDMITX_DRV_NAME "g12a-tohdmitx" #define TOHDMITX_CTRL0 0x0 #define CTRL0_ENABLE_SHIFT 31 -#define CTRL0_I2S_DAT_SEL GENMASK(13, 12) +#define CTRL0_I2S_DAT_SEL_SHIFT 12 +#define CTRL0_I2S_DAT_SEL (0x3 << CTRL0_I2S_DAT_SEL_SHIFT) #define CTRL0_I2S_LRCLK_SEL GENMASK(9, 8) #define CTRL0_I2S_BLK_CAP_INV BIT(7) #define CTRL0_I2S_BCLK_O_INV BIT(6) #define CTRL0_I2S_BCLK_SEL GENMASK(5, 4) #define CTRL0_SPDIF_CLK_CAP_INV BIT(3) #define CTRL0_SPDIF_CLK_O_INV BIT(2) -#define CTRL0_SPDIF_SEL BIT(1) +#define CTRL0_SPDIF_SEL_SHIFT 1 +#define CTRL0_SPDIF_SEL (0x1 << CTRL0_SPDIF_SEL_SHIFT) #define CTRL0_SPDIF_CLK_SEL BIT(0) -struct g12a_tohdmitx_input { - struct snd_soc_pcm_stream params; - unsigned int fmt; -}; - -static struct snd_soc_dapm_widget * -g12a_tohdmitx_get_input(struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_path *p = NULL; - struct snd_soc_dapm_widget *in; - - snd_soc_dapm_widget_for_each_source_path(w, p) { - if (!p->connect) - continue; - - /* Check that we still are in the same component */ - if (snd_soc_dapm_to_component(w->dapm) != - snd_soc_dapm_to_component(p->source->dapm)) - continue; - - if (p->source->id == snd_soc_dapm_dai_in) - return p->source; - - in = g12a_tohdmitx_get_input(p->source); - if (in) - return in; - } - - return NULL; -} - -static struct g12a_tohdmitx_input * -g12a_tohdmitx_get_input_data(struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_widget *in = - g12a_tohdmitx_get_input(w); - struct snd_soc_dai *dai; - - if (WARN_ON(!in)) - return NULL; - - dai = in->priv; - - return dai->playback_dma_data; -} - static const char * const g12a_tohdmitx_i2s_mux_texts[] = { "I2S A", "I2S B", "I2S C", }; -static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_i2s_mux_enum, - g12a_tohdmitx_i2s_mux_texts); - -static int g12a_tohdmitx_get_input_val(struct snd_soc_component *component, - unsigned int mask) -{ - unsigned int val; - - snd_soc_component_read(component, TOHDMITX_CTRL0, &val); - return (val & mask) >> __ffs(mask); -} - -static int g12a_tohdmitx_i2s_mux_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_dapm_kcontrol_component(kcontrol); - - ucontrol->value.enumerated.item[0] = - g12a_tohdmitx_get_input_val(component, CTRL0_I2S_DAT_SEL); - - return 0; -} - static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux = ucontrol->value.enumerated.item[0]; - unsigned int val = g12a_tohdmitx_get_input_val(component, - CTRL0_I2S_DAT_SEL); + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL0_I2S_DAT_SEL, + FIELD_PREP(CTRL0_I2S_DAT_SEL, + mux)); + + if (!changed) + return 0; /* Force disconnect of the mux while updating */ - if (val != mux) - snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); - snd_soc_component_update_bits(component, TOHDMITX_CTRL0, + snd_soc_component_update_bits(component, e->reg, CTRL0_I2S_DAT_SEL | CTRL0_I2S_LRCLK_SEL | CTRL0_I2S_BCLK_SEL, @@ -130,30 +69,19 @@ static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, return 0; } +static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_i2s_mux_enum, TOHDMITX_CTRL0, + CTRL0_I2S_DAT_SEL_SHIFT, + g12a_tohdmitx_i2s_mux_texts); + static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux = SOC_DAPM_ENUM_EXT("I2S Source", g12a_tohdmitx_i2s_mux_enum, - g12a_tohdmitx_i2s_mux_get_enum, + snd_soc_dapm_get_enum_double, g12a_tohdmitx_i2s_mux_put_enum); static const char * const g12a_tohdmitx_spdif_mux_texts[] = { "SPDIF A", "SPDIF B", }; -static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_spdif_mux_enum, - g12a_tohdmitx_spdif_mux_texts); - -static int g12a_tohdmitx_spdif_mux_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_dapm_kcontrol_component(kcontrol); - - ucontrol->value.enumerated.item[0] = - g12a_tohdmitx_get_input_val(component, CTRL0_SPDIF_SEL); - - return 0; -} - static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -162,13 +90,18 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux = ucontrol->value.enumerated.item[0]; - unsigned int val = g12a_tohdmitx_get_input_val(component, - CTRL0_SPDIF_SEL); + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, TOHDMITX_CTRL0, + CTRL0_SPDIF_SEL, + FIELD_PREP(CTRL0_SPDIF_SEL, mux)); + + if (!changed) + return 0; /* Force disconnect of the mux while updating */ - if (val != mux) - snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); snd_soc_component_update_bits(component, TOHDMITX_CTRL0, CTRL0_SPDIF_SEL | @@ -181,9 +114,13 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, return 0; } +static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_spdif_mux_enum, TOHDMITX_CTRL0, + CTRL0_SPDIF_SEL_SHIFT, + g12a_tohdmitx_spdif_mux_texts); + static const struct snd_kcontrol_new g12a_tohdmitx_spdif_mux = SOC_DAPM_ENUM_EXT("SPDIF Source", g12a_tohdmitx_spdif_mux_enum, - g12a_tohdmitx_spdif_mux_get_enum, + snd_soc_dapm_get_enum_double, g12a_tohdmitx_spdif_mux_put_enum); static const struct snd_kcontrol_new g12a_tohdmitx_out_enable = @@ -201,83 +138,13 @@ static const struct snd_soc_dapm_widget g12a_tohdmitx_widgets[] = { &g12a_tohdmitx_out_enable), }; -static int g12a_tohdmitx_input_probe(struct snd_soc_dai *dai) -{ - struct g12a_tohdmitx_input *data; - - data = kzalloc(sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; - - dai->playback_dma_data = data; - return 0; -} - -static int g12a_tohdmitx_input_remove(struct snd_soc_dai *dai) -{ - kfree(dai->playback_dma_data); - return 0; -} - -static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct g12a_tohdmitx_input *data = dai->playback_dma_data; - - data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params)); - data->params.rate_min = params_rate(params); - data->params.rate_max = params_rate(params); - data->params.formats = 1 << params_format(params); - data->params.channels_min = params_channels(params); - data->params.channels_max = params_channels(params); - data->params.sig_bits = dai->driver->playback.sig_bits; - - return 0; -} - - -static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct g12a_tohdmitx_input *data = dai->playback_dma_data; - - /* Save the source stream format for the downstream link */ - data->fmt = fmt; - return 0; -} - -static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct g12a_tohdmitx_input *in_data = - g12a_tohdmitx_get_input_data(dai->capture_widget); - - if (!in_data) - return -ENODEV; - - if (WARN_ON(!rtd->dai_link->params)) { - dev_warn(dai->dev, "codec2codec link expected\n"); - return -EINVAL; - } - - /* Replace link params with the input params */ - rtd->dai_link->params = &in_data->params; - - if (!in_data->fmt) - return 0; - - return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt); -} - static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = { - .hw_params = g12a_tohdmitx_input_hw_params, - .set_fmt = g12a_tohdmitx_input_set_fmt, + .hw_params = meson_codec_glue_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, }; static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = { - .startup = g12a_tohdmitx_output_startup, + .startup = meson_codec_glue_output_startup, }; #define TOHDMITX_SPDIF_FORMATS \ @@ -304,8 +171,8 @@ static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = { .id = (xid), \ .playback = TOHDMITX_STREAM(xname, "Playback", xfmt, xchmax), \ .ops = &g12a_tohdmitx_input_ops, \ - .probe = g12a_tohdmitx_input_probe, \ - .remove = g12a_tohdmitx_input_remove, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ } #define TOHDMITX_OUT(xname, xid, xfmt, xchmax) { \ diff --git a/sound/soc/meson/meson-codec-glue.c b/sound/soc/meson/meson-codec-glue.c new file mode 100644 index 00000000000000..97bbc967e1769b --- /dev/null +++ b/sound/soc/meson/meson-codec-glue.c @@ -0,0 +1,149 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2019 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +#include "meson-codec-glue.h" + +static struct snd_soc_dapm_widget * +meson_codec_glue_get_input(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_widget *in; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + /* Check that we still are in the same component */ + if (snd_soc_dapm_to_component(w->dapm) != + snd_soc_dapm_to_component(p->source->dapm)) + continue; + + if (p->source->id == snd_soc_dapm_dai_in) + return p->source; + + in = meson_codec_glue_get_input(p->source); + if (in) + return in; + } + + return NULL; +} + +static void meson_codec_glue_input_set_data(struct snd_soc_dai *dai, + struct meson_codec_glue_input *data) +{ + dai->playback_dma_data = data; +} + +struct meson_codec_glue_input * +meson_codec_glue_input_get_data(struct snd_soc_dai *dai) +{ + return dai->playback_dma_data; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_get_data); + +static struct meson_codec_glue_input * +meson_codec_glue_output_get_input_data(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget *in = + meson_codec_glue_get_input(w); + struct snd_soc_dai *dai; + + if (WARN_ON(!in)) + return NULL; + + dai = in->priv; + + return meson_codec_glue_input_get_data(dai); +} + +int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params)); + data->params.rate_min = params_rate(params); + data->params.rate_max = params_rate(params); + data->params.formats = 1 << params_format(params); + data->params.channels_min = params_channels(params); + data->params.channels_max = params_channels(params); + data->params.sig_bits = dai->driver->playback.sig_bits; + + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_hw_params); + +int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + /* Save the source stream format for the downstream link */ + data->fmt = fmt; + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_set_fmt); + +int meson_codec_glue_output_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct meson_codec_glue_input *in_data = + meson_codec_glue_output_get_input_data(dai->capture_widget); + + if (!in_data) + return -ENODEV; + + if (WARN_ON(!rtd->dai_link->params)) { + dev_warn(dai->dev, "codec2codec link expected\n"); + return -EINVAL; + } + + /* Replace link params with the input params */ + rtd->dai_link->params = &in_data->params; + + if (!in_data->fmt) + return 0; + + return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt); +} +EXPORT_SYMBOL_GPL(meson_codec_glue_output_startup); + +int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data; + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + meson_codec_glue_input_set_data(dai, data); + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_probe); + +int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + kfree(data); + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_remove); + +MODULE_AUTHOR("Jerome Brunet "); +MODULE_DESCRIPTION("Amlogic Codec Glue Helpers"); +MODULE_LICENSE("GPL v2"); + diff --git a/sound/soc/meson/meson-codec-glue.h b/sound/soc/meson/meson-codec-glue.h new file mode 100644 index 00000000000000..07f99446c0c6bf --- /dev/null +++ b/sound/soc/meson/meson-codec-glue.h @@ -0,0 +1,32 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_CODEC_GLUE_H +#define _MESON_CODEC_GLUE_H + +#include + +struct meson_codec_glue_input { + struct snd_soc_pcm_stream params; + unsigned int fmt; +}; + +/* Input helpers */ +struct meson_codec_glue_input * +meson_codec_glue_input_get_data(struct snd_soc_dai *dai); +int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt); +int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai); +int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai); + +/* Output helpers */ +int meson_codec_glue_output_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +#endif /* _MESON_CODEC_GLUE_H */ From 06b72824386795bf6f0a6ac0f0cfef6b7f0165c1 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:53 +0100 Subject: [PATCH 086/415] ASoC: meson: aiu: add audio output dt-bindings Add the dt-bindings and documentation of the AIU audio controller. This component provides most of the audio outputs found on the Amlogic Gx SoC family. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-4-jbrunet@baylibre.com Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,aiu.yaml | 111 ++++++++++++++++++ include/dt-bindings/sound/meson-aiu.h | 18 +++ 2 files changed, 129 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,aiu.yaml create mode 100644 include/dt-bindings/sound/meson-aiu.h diff --git a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml new file mode 100644 index 00000000000000..3ef7632dcb5992 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml @@ -0,0 +1,111 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,aiu.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AIU audio output controller + +maintainers: + - Jerome Brunet + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 2 + + compatible: + items: + - enum: + - amlogic,aiu-gxbb + - amlogic,aiu-gxl + - const: + amlogic,aiu + + clocks: + items: + - description: AIU peripheral clock + - description: I2S peripheral clock + - description: I2S output clock + - description: I2S master clock + - description: I2S mixer clock + - description: SPDIF peripheral clock + - description: SPDIF output clock + - description: SPDIF master clock + - description: SPDIF master clock multiplexer + + clock-names: + items: + - const: pclk + - const: i2s_pclk + - const: i2s_aoclk + - const: i2s_mclk + - const: i2s_mixer + - const: spdif_pclk + - const: spdif_aoclk + - const: spdif_mclk + - const: spdif_mclk_sel + + interrupts: + items: + - description: I2S interrupt line + - description: SPDIF interrupt line + + interrupt-names: + items: + - const: i2s + - const: spdif + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - clocks + - clock-names + - interrupts + - interrupt-names + - reg + - resets + +examples: + - | + #include + #include + #include + #include + + aiu: audio-controller@5400 { + compatible = "amlogic,aiu-gxl", "amlogic,aiu"; + #sound-dai-cells = <2>; + reg = <0x0 0x5400 0x0 0x2ac>; + interrupts = , + ; + interrupt-names = "i2s", "spdif"; + clocks = <&clkc CLKID_AIU_GLUE>, + <&clkc CLKID_I2S_OUT>, + <&clkc CLKID_AOCLK_GATE>, + <&clkc CLKID_CTS_AMCLK>, + <&clkc CLKID_MIXER_IFACE>, + <&clkc CLKID_IEC958>, + <&clkc CLKID_IEC958_GATE>, + <&clkc CLKID_CTS_MCLK_I958>, + <&clkc CLKID_CTS_I958>; + clock-names = "pclk", + "i2s_pclk", + "i2s_aoclk", + "i2s_mclk", + "i2s_mixer", + "spdif_pclk", + "spdif_aoclk", + "spdif_mclk", + "spdif_mclk_sel"; + resets = <&reset RESET_AIU>; + }; + diff --git a/include/dt-bindings/sound/meson-aiu.h b/include/dt-bindings/sound/meson-aiu.h new file mode 100644 index 00000000000000..1051b8af298b8d --- /dev/null +++ b/include/dt-bindings/sound/meson-aiu.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_MESON_AIU_H +#define __DT_MESON_AIU_H + +#define AIU_CPU 0 +#define AIU_HDMI 1 +#define AIU_ACODEC 2 + +#define CPU_I2S_FIFO 0 +#define CPU_SPDIF_FIFO 1 +#define CPU_I2S_ENCODER 2 +#define CPU_SPDIF_ENCODER 3 + +#define CTRL_I2S 0 +#define CTRL_PCM 1 +#define CTRL_OUT 2 + +#endif /* __DT_MESON_AIU_H */ From 6ae9ca9ce986bffe71fd0fbf9595de8500891b52 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:54 +0100 Subject: [PATCH 087/415] ASoC: meson: aiu: add i2s and spdif support Add support for the i2s and spdif audio outputs (AIU) found in the amlogic Gx SoC family Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-5-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 7 + sound/soc/meson/Makefile | 7 + sound/soc/meson/aiu-encoder-i2s.c | 324 ++++++++++++++++++++++++++ sound/soc/meson/aiu-encoder-spdif.c | 209 +++++++++++++++++ sound/soc/meson/aiu-fifo-i2s.c | 153 ++++++++++++ sound/soc/meson/aiu-fifo-spdif.c | 186 +++++++++++++++ sound/soc/meson/aiu-fifo.c | 223 ++++++++++++++++++ sound/soc/meson/aiu-fifo.h | 50 ++++ sound/soc/meson/aiu.c | 348 ++++++++++++++++++++++++++++ sound/soc/meson/aiu.h | 82 +++++++ 10 files changed, 1589 insertions(+) create mode 100644 sound/soc/meson/aiu-encoder-i2s.c create mode 100644 sound/soc/meson/aiu-encoder-spdif.c create mode 100644 sound/soc/meson/aiu-fifo-i2s.c create mode 100644 sound/soc/meson/aiu-fifo-spdif.c create mode 100644 sound/soc/meson/aiu-fifo.c create mode 100644 sound/soc/meson/aiu-fifo.h create mode 100644 sound/soc/meson/aiu.c create mode 100644 sound/soc/meson/aiu.h diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index ee6d53949d4536..ca269dedfc7fdc 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -2,6 +2,13 @@ menu "ASoC support for Amlogic platforms" depends on ARCH_MESON || COMPILE_TEST +config SND_MESON_AIU + tristate "Amlogic AIU" + select SND_PCM_IEC958 + help + Select Y or M to add support for the Audio output subsystem found + in the Amlogic GX SoC family + config SND_MESON_AXG_FIFO tristate select REGMAP_MMIO diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 529a807b3f376f..a7b79d717288f3 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -1,5 +1,11 @@ # SPDX-License-Identifier: (GPL-2.0 OR MIT) +snd-soc-meson-aiu-objs := aiu.o +snd-soc-meson-aiu-objs += aiu-encoder-i2s.o +snd-soc-meson-aiu-objs += aiu-encoder-spdif.o +snd-soc-meson-aiu-objs += aiu-fifo.o +snd-soc-meson-aiu-objs += aiu-fifo-i2s.o +snd-soc-meson-aiu-objs += aiu-fifo-spdif.o snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o @@ -14,6 +20,7 @@ snd-soc-meson-axg-pdm-objs := axg-pdm.o snd-soc-meson-codec-glue-objs := meson-codec-glue.o snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o +obj-$(CONFIG_SND_MESON_AIU) += snd-soc-meson-aiu.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c new file mode 100644 index 00000000000000..13bf029086a9df --- /dev/null +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -0,0 +1,324 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "aiu.h" + +#define AIU_I2S_SOURCE_DESC_MODE_8CH BIT(0) +#define AIU_I2S_SOURCE_DESC_MODE_24BIT BIT(5) +#define AIU_I2S_SOURCE_DESC_MODE_32BIT BIT(9) +#define AIU_I2S_SOURCE_DESC_MODE_SPLIT BIT(11) +#define AIU_RST_SOFT_I2S_FAST BIT(0) + +#define AIU_I2S_DAC_CFG_MSB_FIRST BIT(2) +#define AIU_I2S_MISC_HOLD_EN BIT(2) +#define AIU_CLK_CTRL_I2S_DIV_EN BIT(0) +#define AIU_CLK_CTRL_I2S_DIV GENMASK(3, 2) +#define AIU_CLK_CTRL_AOCLK_INVERT BIT(6) +#define AIU_CLK_CTRL_LRCLK_INVERT BIT(7) +#define AIU_CLK_CTRL_LRCLK_SKEW GENMASK(9, 8) +#define AIU_CLK_CTRL_MORE_HDMI_AMCLK BIT(6) +#define AIU_CLK_CTRL_MORE_I2S_DIV GENMASK(5, 0) +#define AIU_CODEC_DAC_LRCLK_CTRL_DIV GENMASK(11, 0) + +struct aiu_encoder_i2s { + struct clk *aoclk; + struct clk *mclk; + struct clk *mixer; + struct clk *pclk; +}; + +static void aiu_encoder_i2s_divider_enable(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV_EN, + enable ? AIU_CLK_CTRL_I2S_DIV_EN : 0); +} + +static void aiu_encoder_i2s_hold(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_I2S_MISC, + AIU_I2S_MISC_HOLD_EN, + enable ? AIU_I2S_MISC_HOLD_EN : 0); +} + +static int aiu_encoder_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + aiu_encoder_i2s_hold(component, false); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + aiu_encoder_i2s_hold(component, true); + return 0; + + default: + return -EINVAL; + } +} + +static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + /* Always operate in split (classic interleaved) mode */ + unsigned int desc = AIU_I2S_SOURCE_DESC_MODE_SPLIT; + unsigned int val; + + /* Reset required to update the pipeline */ + snd_soc_component_write(component, AIU_RST_SOFT, AIU_RST_SOFT_I2S_FAST); + snd_soc_component_read(component, AIU_I2S_SYNC, &val); + + switch (params_physical_width(params)) { + case 16: /* Nothing to do */ + break; + + case 32: + desc |= (AIU_I2S_SOURCE_DESC_MODE_24BIT | + AIU_I2S_SOURCE_DESC_MODE_32BIT); + break; + + default: + return -EINVAL; + } + + switch (params_channels(params)) { + case 2: /* Nothing to do */ + break; + case 8: + desc |= AIU_I2S_SOURCE_DESC_MODE_8CH; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_I2S_SOURCE_DESC, + AIU_I2S_SOURCE_DESC_MODE_8CH | + AIU_I2S_SOURCE_DESC_MODE_24BIT | + AIU_I2S_SOURCE_DESC_MODE_32BIT | + AIU_I2S_SOURCE_DESC_MODE_SPLIT, + desc); + + return 0; +} + +static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(component); + unsigned int srate = params_rate(params); + unsigned int fs, bs; + + /* Get the oversampling factor */ + fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); + + if (fs % 64) + return -EINVAL; + + /* Send data MSB first */ + snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, + AIU_I2S_DAC_CFG_MSB_FIRST, + AIU_I2S_DAC_CFG_MSB_FIRST); + + /* Set bclk to lrlck ratio */ + snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, + AIU_CODEC_DAC_LRCLK_CTRL_DIV, + FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, + 64 - 1)); + + /* Use CLK_MORE for mclk to bclk divider */ + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV, 0); + + /* + * NOTE: this HW is odd. + * In most configuration, the i2s divider is 'mclk / blck'. + * However, in 16 bits - 8ch mode, this factor needs to be + * increased by 50% to get the correct output rate. + * No idea why ! + */ + bs = fs / 64; + if (params_width(params) == 16 && params_channels(params) == 8) { + if (bs % 2) { + dev_err(component->dev, + "Cannot increase i2s divider by 50%%\n"); + return -EINVAL; + } + bs += bs / 2; + } + + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, + AIU_CLK_CTRL_MORE_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, + bs - 1)); + + /* Make sure amclk is used for HDMI i2s as well */ + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, + AIU_CLK_CTRL_MORE_HDMI_AMCLK, + AIU_CLK_CTRL_MORE_HDMI_AMCLK); + + return 0; +} + +static int aiu_encoder_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + int ret; + + /* Disable the clock while changing the settings */ + aiu_encoder_i2s_divider_enable(component, false); + + ret = aiu_encoder_i2s_setup_desc(component, params); + if (ret) { + dev_err(dai->dev, "setting i2s desc failed\n"); + return ret; + } + + ret = aiu_encoder_i2s_set_clocks(component, params); + if (ret) { + dev_err(dai->dev, "setting i2s clocks failed\n"); + return ret; + } + + aiu_encoder_i2s_divider_enable(component, true); + + return 0; +} + +static int aiu_encoder_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + aiu_encoder_i2s_divider_enable(component, false); + + return 0; +} + +static int aiu_encoder_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + unsigned int inv = fmt & SND_SOC_DAIFMT_INV_MASK; + unsigned int val = 0; + unsigned int skew; + + /* Only CPU Master / Codec Slave supported ATM */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + if (inv == SND_SOC_DAIFMT_NB_IF || + inv == SND_SOC_DAIFMT_IB_IF) + val |= AIU_CLK_CTRL_LRCLK_INVERT; + + if (inv == SND_SOC_DAIFMT_IB_NF || + inv == SND_SOC_DAIFMT_IB_IF) + val |= AIU_CLK_CTRL_AOCLK_INVERT; + + /* Signal skew */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Invert sample clock for i2s */ + val ^= AIU_CLK_CTRL_LRCLK_INVERT; + skew = 1; + break; + case SND_SOC_DAIFMT_LEFT_J: + skew = 0; + break; + default: + return -EINVAL; + } + + val |= FIELD_PREP(AIU_CLK_CTRL_LRCLK_SKEW, skew); + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_LRCLK_INVERT | + AIU_CLK_CTRL_AOCLK_INVERT | + AIU_CLK_CTRL_LRCLK_SKEW, + val); + + return 0; +} + +static int aiu_encoder_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + int ret; + + if (WARN_ON(clk_id != 0)) + return -EINVAL; + + if (dir == SND_SOC_CLOCK_IN) + return 0; + + ret = clk_set_rate(aiu->i2s.clks[MCLK].clk, freq); + if (ret) + dev_err(dai->dev, "Failed to set sysclk to %uHz", freq); + + return ret; +} + +static const unsigned int hw_channels[] = {2, 8}; +static const struct snd_pcm_hw_constraint_list hw_channel_constraints = { + .list = hw_channels, + .count = ARRAY_SIZE(hw_channels), + .mask = 0, +}; + +static int aiu_encoder_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + int ret; + + /* Make sure the encoder gets either 2 or 8 channels */ + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &hw_channel_constraints); + if (ret) { + dev_err(dai->dev, "adding channels constraints failed\n"); + return ret; + } + + ret = clk_bulk_prepare_enable(aiu->i2s.clk_num, aiu->i2s.clks); + if (ret) + dev_err(dai->dev, "failed to enable i2s clocks\n"); + + return ret; +} + +static void aiu_encoder_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + + clk_bulk_disable_unprepare(aiu->i2s.clk_num, aiu->i2s.clks); +} + +const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops = { + .trigger = aiu_encoder_i2s_trigger, + .hw_params = aiu_encoder_i2s_hw_params, + .hw_free = aiu_encoder_i2s_hw_free, + .set_fmt = aiu_encoder_i2s_set_fmt, + .set_sysclk = aiu_encoder_i2s_set_sysclk, + .startup = aiu_encoder_i2s_startup, + .shutdown = aiu_encoder_i2s_shutdown, +}; + diff --git a/sound/soc/meson/aiu-encoder-spdif.c b/sound/soc/meson/aiu-encoder-spdif.c new file mode 100644 index 00000000000000..de850913975f17 --- /dev/null +++ b/sound/soc/meson/aiu-encoder-spdif.c @@ -0,0 +1,209 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include + +#include "aiu.h" + +#define AIU_958_MISC_NON_PCM BIT(0) +#define AIU_958_MISC_MODE_16BITS BIT(1) +#define AIU_958_MISC_16BITS_ALIGN GENMASK(6, 5) +#define AIU_958_MISC_MODE_32BITS BIT(7) +#define AIU_958_MISC_U_FROM_STREAM BIT(12) +#define AIU_958_MISC_FORCE_LR BIT(13) +#define AIU_958_CTRL_HOLD_EN BIT(0) +#define AIU_CLK_CTRL_958_DIV_EN BIT(1) +#define AIU_CLK_CTRL_958_DIV GENMASK(5, 4) +#define AIU_CLK_CTRL_958_DIV_MORE BIT(12) + +#define AIU_CS_WORD_LEN 4 +#define AIU_958_INTERNAL_DIV 2 + +static void +aiu_encoder_spdif_divider_enable(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_958_DIV_EN, + enable ? AIU_CLK_CTRL_958_DIV_EN : 0); +} + +static void aiu_encoder_spdif_hold(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_958_CTRL, + AIU_958_CTRL_HOLD_EN, + enable ? AIU_958_CTRL_HOLD_EN : 0); +} + +static int +aiu_encoder_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + aiu_encoder_spdif_hold(component, false); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + aiu_encoder_spdif_hold(component, true); + return 0; + + default: + return -EINVAL; + } +} + +static int aiu_encoder_spdif_setup_cs_word(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + u8 cs[AIU_CS_WORD_LEN]; + unsigned int val; + int ret; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, + AIU_CS_WORD_LEN); + if (ret < 0) + return ret; + + /* Write the 1st half word */ + val = cs[1] | cs[0] << 8; + snd_soc_component_write(component, AIU_958_CHSTAT_L0, val); + snd_soc_component_write(component, AIU_958_CHSTAT_R0, val); + + /* Write the 2nd half word */ + val = cs[3] | cs[2] << 8; + snd_soc_component_write(component, AIU_958_CHSTAT_L1, val); + snd_soc_component_write(component, AIU_958_CHSTAT_R1, val); + + return 0; +} + +static int aiu_encoder_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu *aiu = snd_soc_component_get_drvdata(component); + unsigned int val = 0, mrate; + int ret; + + /* Disable the clock while changing the settings */ + aiu_encoder_spdif_divider_enable(component, false); + + switch (params_physical_width(params)) { + case 16: + val |= AIU_958_MISC_MODE_16BITS; + val |= FIELD_PREP(AIU_958_MISC_16BITS_ALIGN, 2); + break; + case 32: + val |= AIU_958_MISC_MODE_32BITS; + break; + default: + dev_err(dai->dev, "Unsupport physical width\n"); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_958_MISC, + AIU_958_MISC_NON_PCM | + AIU_958_MISC_MODE_16BITS | + AIU_958_MISC_16BITS_ALIGN | + AIU_958_MISC_MODE_32BITS | + AIU_958_MISC_FORCE_LR | + AIU_958_MISC_U_FROM_STREAM, + val); + + /* Set the stream channel status word */ + ret = aiu_encoder_spdif_setup_cs_word(component, params); + if (ret) { + dev_err(dai->dev, "failed to set channel status word\n"); + return ret; + } + + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_958_DIV | + AIU_CLK_CTRL_958_DIV_MORE, + FIELD_PREP(AIU_CLK_CTRL_958_DIV, + __ffs(AIU_958_INTERNAL_DIV))); + + /* 2 * 32bits per subframe * 2 channels = 128 */ + mrate = params_rate(params) * 128 * AIU_958_INTERNAL_DIV; + ret = clk_set_rate(aiu->spdif.clks[MCLK].clk, mrate); + if (ret) { + dev_err(dai->dev, "failed to set mclk rate\n"); + return ret; + } + + aiu_encoder_spdif_divider_enable(component, true); + + return 0; +} + +static int aiu_encoder_spdif_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + aiu_encoder_spdif_divider_enable(component, false); + + return 0; +} + +static int aiu_encoder_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + int ret; + + /* + * NOTE: Make sure the spdif block is on its own divider. + * + * The spdif can be clocked by the i2s master clock or its own + * clock. We should (in theory) change the source depending on the + * origin of the data. + * + * However, considering the clocking scheme used on these platforms, + * the master clocks will pick the same PLL source when they are + * playing from the same FIFO. The clock should be in sync so, it + * should not be necessary to reparent the spdif master clock. + */ + ret = clk_set_parent(aiu->spdif.clks[MCLK].clk, + aiu->spdif_mclk); + if (ret) + return ret; + + ret = clk_bulk_prepare_enable(aiu->spdif.clk_num, aiu->spdif.clks); + if (ret) + dev_err(dai->dev, "failed to enable spdif clocks\n"); + + return ret; +} + +static void aiu_encoder_spdif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + + clk_bulk_disable_unprepare(aiu->spdif.clk_num, aiu->spdif.clks); +} + +const struct snd_soc_dai_ops aiu_encoder_spdif_dai_ops = { + .trigger = aiu_encoder_spdif_trigger, + .hw_params = aiu_encoder_spdif_hw_params, + .hw_free = aiu_encoder_spdif_hw_free, + .startup = aiu_encoder_spdif_startup, + .shutdown = aiu_encoder_spdif_shutdown, +}; diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c new file mode 100644 index 00000000000000..9a5271ce80fe38 --- /dev/null +++ b/sound/soc/meson/aiu-fifo-i2s.c @@ -0,0 +1,153 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "aiu.h" +#include "aiu-fifo.h" + +#define AIU_I2S_SOURCE_DESC_MODE_8CH BIT(0) +#define AIU_I2S_SOURCE_DESC_MODE_24BIT BIT(5) +#define AIU_I2S_SOURCE_DESC_MODE_32BIT BIT(9) +#define AIU_I2S_SOURCE_DESC_MODE_SPLIT BIT(11) +#define AIU_MEM_I2S_MASKS_IRQ_BLOCK GENMASK(31, 16) +#define AIU_MEM_I2S_CONTROL_MODE_16BIT BIT(6) +#define AIU_MEM_I2S_BUF_CNTL_INIT BIT(0) +#define AIU_RST_SOFT_I2S_FAST BIT(0) + +#define AIU_FIFO_I2S_BLOCK 256 + +static struct snd_pcm_hardware fifo_i2s_pcm = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = AIU_FORMATS, + .rate_min = 5512, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, + .period_bytes_min = AIU_FIFO_I2S_BLOCK, + .period_bytes_max = AIU_FIFO_I2S_BLOCK * USHRT_MAX, + .periods_min = 2, + .periods_max = UINT_MAX, + + /* No real justification for this */ + .buffer_bytes_max = 1 * 1024 * 1024, +}; + +static int aiu_fifo_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + unsigned int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + snd_soc_component_write(component, AIU_RST_SOFT, + AIU_RST_SOFT_I2S_FAST); + snd_soc_component_read(component, AIU_I2S_SYNC, &val); + break; + } + + return aiu_fifo_trigger(substream, cmd, dai); +} + +static int aiu_fifo_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + int ret; + + ret = aiu_fifo_prepare(substream, dai); + if (ret) + return ret; + + snd_soc_component_update_bits(component, + AIU_MEM_I2S_BUF_CNTL, + AIU_MEM_I2S_BUF_CNTL_INIT, + AIU_MEM_I2S_BUF_CNTL_INIT); + snd_soc_component_update_bits(component, + AIU_MEM_I2S_BUF_CNTL, + AIU_MEM_I2S_BUF_CNTL_INIT, 0); + + return 0; +} + +static int aiu_fifo_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu_fifo *fifo = dai->playback_dma_data; + unsigned int val; + int ret; + + ret = aiu_fifo_hw_params(substream, params, dai); + if (ret) + return ret; + + switch (params_physical_width(params)) { + case 16: + val = AIU_MEM_I2S_CONTROL_MODE_16BIT; + break; + case 32: + val = 0; + break; + default: + dev_err(dai->dev, "Unsupported physical width %u\n", + params_physical_width(params)); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_MEM_I2S_CONTROL, + AIU_MEM_I2S_CONTROL_MODE_16BIT, + val); + + /* Setup the irq periodicity */ + val = params_period_bytes(params) / fifo->fifo_block; + val = FIELD_PREP(AIU_MEM_I2S_MASKS_IRQ_BLOCK, val); + snd_soc_component_update_bits(component, AIU_MEM_I2S_MASKS, + AIU_MEM_I2S_MASKS_IRQ_BLOCK, val); + + return 0; +} + +const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops = { + .trigger = aiu_fifo_i2s_trigger, + .prepare = aiu_fifo_i2s_prepare, + .hw_params = aiu_fifo_i2s_hw_params, + .hw_free = aiu_fifo_hw_free, + .startup = aiu_fifo_startup, + .shutdown = aiu_fifo_shutdown, +}; + +int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu *aiu = snd_soc_component_get_drvdata(component); + struct aiu_fifo *fifo; + int ret; + + ret = aiu_fifo_dai_probe(dai); + if (ret) + return ret; + + fifo = dai->playback_dma_data; + + fifo->pcm = &fifo_i2s_pcm; + fifo->mem_offset = AIU_MEM_I2S_START; + fifo->fifo_block = AIU_FIFO_I2S_BLOCK; + fifo->pclk = aiu->i2s.clks[PCLK].clk; + fifo->irq = aiu->i2s.irq; + + return 0; +} diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c new file mode 100644 index 00000000000000..44eb6faacf44d0 --- /dev/null +++ b/sound/soc/meson/aiu-fifo-spdif.c @@ -0,0 +1,186 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +#include "aiu.h" +#include "aiu-fifo.h" + +#define AIU_IEC958_DCU_FF_CTRL_EN BIT(0) +#define AIU_IEC958_DCU_FF_CTRL_AUTO_DISABLE BIT(1) +#define AIU_IEC958_DCU_FF_CTRL_IRQ_MODE GENMASK(3, 2) +#define AIU_IEC958_DCU_FF_CTRL_IRQ_OUT_THD BIT(2) +#define AIU_IEC958_DCU_FF_CTRL_IRQ_FRAME_READ BIT(3) +#define AIU_IEC958_DCU_FF_CTRL_SYNC_HEAD_EN BIT(4) +#define AIU_IEC958_DCU_FF_CTRL_BYTE_SEEK BIT(5) +#define AIU_IEC958_DCU_FF_CTRL_CONTINUE BIT(6) +#define AIU_MEM_IEC958_CONTROL_ENDIAN GENMASK(5, 3) +#define AIU_MEM_IEC958_CONTROL_RD_DDR BIT(6) +#define AIU_MEM_IEC958_CONTROL_MODE_16BIT BIT(7) +#define AIU_MEM_IEC958_CONTROL_MODE_LINEAR BIT(8) +#define AIU_MEM_IEC958_BUF_CNTL_INIT BIT(0) + +#define AIU_FIFO_SPDIF_BLOCK 8 + +static struct snd_pcm_hardware fifo_spdif_pcm = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = AIU_FORMATS, + .rate_min = 5512, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = AIU_FIFO_SPDIF_BLOCK, + .period_bytes_max = AIU_FIFO_SPDIF_BLOCK * USHRT_MAX, + .periods_min = 2, + .periods_max = UINT_MAX, + + /* No real justification for this */ + .buffer_bytes_max = 1 * 1024 * 1024, +}; + +static void fifo_spdif_dcu_enable(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_IEC958_DCU_FF_CTRL, + AIU_IEC958_DCU_FF_CTRL_EN, + enable ? AIU_IEC958_DCU_FF_CTRL_EN : 0); +} + +static int fifo_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + int ret; + + ret = aiu_fifo_trigger(substream, cmd, dai); + if (ret) + return ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fifo_spdif_dcu_enable(component, true); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + fifo_spdif_dcu_enable(component, false); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fifo_spdif_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + int ret; + + ret = aiu_fifo_prepare(substream, dai); + if (ret) + return ret; + + snd_soc_component_update_bits(component, + AIU_MEM_IEC958_BUF_CNTL, + AIU_MEM_IEC958_BUF_CNTL_INIT, + AIU_MEM_IEC958_BUF_CNTL_INIT); + snd_soc_component_update_bits(component, + AIU_MEM_IEC958_BUF_CNTL, + AIU_MEM_IEC958_BUF_CNTL_INIT, 0); + + return 0; +} + +static int fifo_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + unsigned int val; + int ret; + + ret = aiu_fifo_hw_params(substream, params, dai); + if (ret) + return ret; + + val = AIU_MEM_IEC958_CONTROL_RD_DDR | + AIU_MEM_IEC958_CONTROL_MODE_LINEAR; + + switch (params_physical_width(params)) { + case 16: + val |= AIU_MEM_IEC958_CONTROL_MODE_16BIT; + break; + case 32: + break; + default: + dev_err(dai->dev, "Unsupported physical width %u\n", + params_physical_width(params)); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_MEM_IEC958_CONTROL, + AIU_MEM_IEC958_CONTROL_ENDIAN | + AIU_MEM_IEC958_CONTROL_RD_DDR | + AIU_MEM_IEC958_CONTROL_MODE_LINEAR | + AIU_MEM_IEC958_CONTROL_MODE_16BIT, + val); + + /* Number bytes read by the FIFO between each IRQ */ + snd_soc_component_write(component, AIU_IEC958_BPF, + params_period_bytes(params)); + + /* + * AUTO_DISABLE and SYNC_HEAD are enabled by default but + * this should be disabled in PCM (uncompressed) mode + */ + snd_soc_component_update_bits(component, AIU_IEC958_DCU_FF_CTRL, + AIU_IEC958_DCU_FF_CTRL_AUTO_DISABLE | + AIU_IEC958_DCU_FF_CTRL_IRQ_MODE | + AIU_IEC958_DCU_FF_CTRL_SYNC_HEAD_EN, + AIU_IEC958_DCU_FF_CTRL_IRQ_FRAME_READ); + + return 0; +} + +const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops = { + .trigger = fifo_spdif_trigger, + .prepare = fifo_spdif_prepare, + .hw_params = fifo_spdif_hw_params, + .hw_free = aiu_fifo_hw_free, + .startup = aiu_fifo_startup, + .shutdown = aiu_fifo_shutdown, +}; + +int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu *aiu = snd_soc_component_get_drvdata(component); + struct aiu_fifo *fifo; + int ret; + + ret = aiu_fifo_dai_probe(dai); + if (ret) + return ret; + + fifo = dai->playback_dma_data; + + fifo->pcm = &fifo_spdif_pcm; + fifo->mem_offset = AIU_MEM_IEC958_START; + fifo->fifo_block = 1; + fifo->pclk = aiu->spdif.clks[PCLK].clk; + fifo->irq = aiu->spdif.irq; + + return 0; +} diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c new file mode 100644 index 00000000000000..da8c098e875029 --- /dev/null +++ b/sound/soc/meson/aiu-fifo.c @@ -0,0 +1,223 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "aiu-fifo.h" + +#define AIU_MEM_START 0x00 +#define AIU_MEM_RD 0x04 +#define AIU_MEM_END 0x08 +#define AIU_MEM_MASKS 0x0c +#define AIU_MEM_MASK_CH_RD GENMASK(7, 0) +#define AIU_MEM_MASK_CH_MEM GENMASK(15, 8) +#define AIU_MEM_CONTROL 0x10 +#define AIU_MEM_CONTROL_INIT BIT(0) +#define AIU_MEM_CONTROL_FILL_EN BIT(1) +#define AIU_MEM_CONTROL_EMPTY_EN BIT(2) + +static struct snd_soc_dai *aiu_fifo_dai(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + + return rtd->cpu_dai; +} + +snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_dai *dai = aiu_fifo_dai(substream); + struct aiu_fifo *fifo = dai->playback_dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int addr; + + snd_soc_component_read(component, fifo->mem_offset + AIU_MEM_RD, + &addr); + + return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr); +} + +static void aiu_fifo_enable(struct snd_soc_dai *dai, bool enable) +{ + struct snd_soc_component *component = dai->component; + struct aiu_fifo *fifo = dai->playback_dma_data; + unsigned int en_mask = (AIU_MEM_CONTROL_FILL_EN | + AIU_MEM_CONTROL_EMPTY_EN); + + snd_soc_component_update_bits(component, + fifo->mem_offset + AIU_MEM_CONTROL, + en_mask, enable ? en_mask : 0); +} + +int aiu_fifo_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + aiu_fifo_enable(dai, true); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + aiu_fifo_enable(dai, false); + break; + default: + return -EINVAL; + } + + return 0; +} + +int aiu_fifo_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu_fifo *fifo = dai->playback_dma_data; + + snd_soc_component_update_bits(component, + fifo->mem_offset + AIU_MEM_CONTROL, + AIU_MEM_CONTROL_INIT, + AIU_MEM_CONTROL_INIT); + snd_soc_component_update_bits(component, + fifo->mem_offset + AIU_MEM_CONTROL, + AIU_MEM_CONTROL_INIT, 0); + return 0; +} + +int aiu_fifo_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_component *component = dai->component; + struct aiu_fifo *fifo = dai->playback_dma_data; + dma_addr_t end; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + /* Setup the fifo boundaries */ + end = runtime->dma_addr + runtime->dma_bytes - fifo->fifo_block; + snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_START, + runtime->dma_addr); + snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_RD, + runtime->dma_addr); + snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_END, + end); + + /* Setup the fifo to read all the memory - no skip */ + snd_soc_component_update_bits(component, + fifo->mem_offset + AIU_MEM_MASKS, + AIU_MEM_MASK_CH_RD | AIU_MEM_MASK_CH_MEM, + FIELD_PREP(AIU_MEM_MASK_CH_RD, 0xff) | + FIELD_PREP(AIU_MEM_MASK_CH_MEM, 0xff)); + + return 0; +} + +int aiu_fifo_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_lib_free_pages(substream); +} + +static irqreturn_t aiu_fifo_isr(int irq, void *dev_id) +{ + struct snd_pcm_substream *playback = dev_id; + + snd_pcm_period_elapsed(playback); + + return IRQ_HANDLED; +} + +int aiu_fifo_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu_fifo *fifo = dai->playback_dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, fifo->pcm); + + /* + * Make sure the buffer and period size are multiple of the fifo burst + * size + */ + ret = snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + fifo->fifo_block); + if (ret) + return ret; + + ret = snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + fifo->fifo_block); + if (ret) + return ret; + + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + ret = request_irq(fifo->irq, aiu_fifo_isr, 0, dev_name(dai->dev), + substream); + if (ret) + clk_disable_unprepare(fifo->pclk); + + return ret; +} + +void aiu_fifo_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu_fifo *fifo = dai->playback_dma_data; + + free_irq(fifo->irq, substream); + clk_disable_unprepare(fifo->pclk); +} + +int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + struct snd_pcm_substream *substream = + rtd->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct snd_card *card = rtd->card->snd_card; + struct aiu_fifo *fifo = dai->playback_dma_data; + size_t size = fifo->pcm->buffer_bytes_max; + + snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); + + return 0; +} + +int aiu_fifo_dai_probe(struct snd_soc_dai *dai) +{ + struct aiu_fifo *fifo; + + fifo = kzalloc(sizeof(*fifo), GFP_KERNEL); + if (!fifo) + return -ENOMEM; + + dai->playback_dma_data = fifo; + + return 0; +} + +int aiu_fifo_dai_remove(struct snd_soc_dai *dai) +{ + kfree(dai->playback_dma_data); + + return 0; +} + diff --git a/sound/soc/meson/aiu-fifo.h b/sound/soc/meson/aiu-fifo.h new file mode 100644 index 00000000000000..42ce266677cc86 --- /dev/null +++ b/sound/soc/meson/aiu-fifo.h @@ -0,0 +1,50 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ +/* + * Copyright (c) 2020 BayLibre, SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_AIU_FIFO_H +#define _MESON_AIU_FIFO_H + +struct snd_pcm_hardware; +struct snd_soc_component_driver; +struct snd_soc_dai_driver; +struct clk; +struct snd_pcm_ops; +struct snd_pcm_substream; +struct snd_soc_dai; +struct snd_pcm_hw_params; +struct platform_device; + +struct aiu_fifo { + struct snd_pcm_hardware *pcm; + unsigned int mem_offset; + unsigned int fifo_block; + struct clk *pclk; + int irq; +}; + +int aiu_fifo_dai_probe(struct snd_soc_dai *dai); +int aiu_fifo_dai_remove(struct snd_soc_dai *dai); + +snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream); + +int aiu_fifo_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai); +int aiu_fifo_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +int aiu_fifo_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +int aiu_fifo_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +int aiu_fifo_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +void aiu_fifo_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); + +#endif /* _MESON_AIU_FIFO_H */ diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c new file mode 100644 index 00000000000000..a62aced9b68717 --- /dev/null +++ b/sound/soc/meson/aiu.c @@ -0,0 +1,348 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "aiu.h" +#include "aiu-fifo.h" + +#define AIU_I2S_MISC_958_SRC_SHIFT 3 + +static const char * const aiu_spdif_encode_sel_texts[] = { + "SPDIF", "I2S", +}; + +static SOC_ENUM_SINGLE_DECL(aiu_spdif_encode_sel_enum, AIU_I2S_MISC, + AIU_I2S_MISC_958_SRC_SHIFT, + aiu_spdif_encode_sel_texts); + +static const struct snd_kcontrol_new aiu_spdif_encode_mux = + SOC_DAPM_ENUM("SPDIF Buffer Src", aiu_spdif_encode_sel_enum); + +static const struct snd_soc_dapm_widget aiu_cpu_dapm_widgets[] = { + SND_SOC_DAPM_MUX("SPDIF SRC SEL", SND_SOC_NOPM, 0, 0, + &aiu_spdif_encode_mux), +}; + +static const struct snd_soc_dapm_route aiu_cpu_dapm_routes[] = { + { "I2S Encoder Playback", NULL, "I2S FIFO Playback" }, + { "SPDIF SRC SEL", "SPDIF", "SPDIF FIFO Playback" }, + { "SPDIF SRC SEL", "I2S", "I2S FIFO Playback" }, + { "SPDIF Encoder Playback", NULL, "SPDIF SRC SEL" }, +}; + +int aiu_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name, + unsigned int component_id) +{ + struct snd_soc_dai *dai; + int id; + + if (args->args_count != 2) + return -EINVAL; + + if (args->args[0] != component_id) + return -EINVAL; + + id = args->args[1]; + + if (id < 0 || id >= component->num_dai) + return -EINVAL; + + for_each_component_dais(component, dai) { + if (id == 0) + break; + id--; + } + + *dai_name = dai->driver->name; + + return 0; +} + +static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name) +{ + return aiu_of_xlate_dai_name(component, args, dai_name, AIU_CPU); +} + +static int aiu_cpu_component_probe(struct snd_soc_component *component) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(component); + + /* Required for the SPDIF Source control operation */ + return clk_prepare_enable(aiu->i2s.clks[PCLK].clk); +} + +static void aiu_cpu_component_remove(struct snd_soc_component *component) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(component); + + clk_disable_unprepare(aiu->i2s.clks[PCLK].clk); +} + +static const struct snd_soc_component_driver aiu_cpu_component = { + .name = "AIU CPU", + .dapm_widgets = aiu_cpu_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aiu_cpu_dapm_widgets), + .dapm_routes = aiu_cpu_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aiu_cpu_dapm_routes), + .of_xlate_dai_name = aiu_cpu_of_xlate_dai_name, + .pointer = aiu_fifo_pointer, + .probe = aiu_cpu_component_probe, + .remove = aiu_cpu_component_remove, +}; + +static struct snd_soc_dai_driver aiu_cpu_dai_drv[] = { + [CPU_I2S_FIFO] = { + .name = "I2S FIFO", + .playback = { + .stream_name = "I2S FIFO Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = AIU_FORMATS, + }, + .ops = &aiu_fifo_i2s_dai_ops, + .pcm_new = aiu_fifo_pcm_new, + .probe = aiu_fifo_i2s_dai_probe, + .remove = aiu_fifo_dai_remove, + }, + [CPU_SPDIF_FIFO] = { + .name = "SPDIF FIFO", + .playback = { + .stream_name = "SPDIF FIFO Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = AIU_FORMATS, + }, + .ops = &aiu_fifo_spdif_dai_ops, + .pcm_new = aiu_fifo_pcm_new, + .probe = aiu_fifo_spdif_dai_probe, + .remove = aiu_fifo_dai_remove, + }, + [CPU_I2S_ENCODER] = { + .name = "I2S Encoder", + .playback = { + .stream_name = "I2S Encoder Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = AIU_FORMATS, + }, + .ops = &aiu_encoder_i2s_dai_ops, + }, + [CPU_SPDIF_ENCODER] = { + .name = "SPDIF Encoder", + .playback = { + .stream_name = "SPDIF Encoder Playback", + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), + .formats = AIU_FORMATS, + }, + .ops = &aiu_encoder_spdif_dai_ops, + } +}; + +static const struct regmap_config aiu_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0x2ac, +}; + +static int aiu_clk_bulk_get(struct device *dev, + const char * const *ids, + unsigned int num, + struct aiu_interface *interface) +{ + struct clk_bulk_data *clks; + int i, ret; + + clks = devm_kcalloc(dev, num, sizeof(clks), GFP_KERNEL); + if (!clks) + return -ENOMEM; + + for (i = 0; i < num; i++) + clks[i].id = ids[i]; + + ret = devm_clk_bulk_get(dev, num, clks); + if (ret < 0) + return ret; + + interface->clks = clks; + interface->clk_num = num; + return 0; +} + +static const char * const aiu_i2s_ids[] = { + [PCLK] = "i2s_pclk", + [AOCLK] = "i2s_aoclk", + [MCLK] = "i2s_mclk", + [MIXER] = "i2s_mixer", +}; + +static const char * const aiu_spdif_ids[] = { + [PCLK] = "spdif_pclk", + [AOCLK] = "spdif_aoclk", + [MCLK] = "spdif_mclk_sel" +}; + +static int aiu_clk_get(struct device *dev) +{ + struct aiu *aiu = dev_get_drvdata(dev); + int ret; + + aiu->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(aiu->pclk)) { + if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER) + dev_err(dev, "Can't get the aiu pclk\n"); + return PTR_ERR(aiu->pclk); + } + + aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); + if (IS_ERR(aiu->spdif_mclk)) { + if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER) + dev_err(dev, "Can't get the aiu spdif master clock\n"); + return PTR_ERR(aiu->spdif_mclk); + } + + ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids), + &aiu->i2s); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "Can't get the i2s clocks\n"); + return ret; + } + + ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids), + &aiu->spdif); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "Can't get the spdif clocks\n"); + return ret; + } + + ret = clk_prepare_enable(aiu->pclk); + if (ret) { + dev_err(dev, "peripheral clock enable failed\n"); + return ret; + } + + ret = devm_add_action_or_reset(dev, + (void(*)(void *))clk_disable_unprepare, + aiu->pclk); + if (ret) + dev_err(dev, "failed to add reset action on pclk"); + + return ret; +} + +static int aiu_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + void __iomem *regs; + struct regmap *map; + struct aiu *aiu; + int ret; + + aiu = devm_kzalloc(dev, sizeof(*aiu), GFP_KERNEL); + if (!aiu) + return -ENOMEM; + platform_set_drvdata(pdev, aiu); + + ret = device_reset(dev); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "Failed to reset device\n"); + return ret; + } + + regs = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + map = devm_regmap_init_mmio(dev, regs, &aiu_regmap_cfg); + if (IS_ERR(map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(map)); + return PTR_ERR(map); + } + + aiu->i2s.irq = platform_get_irq_byname(pdev, "i2s"); + if (aiu->i2s.irq < 0) { + dev_err(dev, "Can't get i2s irq\n"); + return aiu->i2s.irq; + } + + aiu->spdif.irq = platform_get_irq_byname(pdev, "spdif"); + if (aiu->spdif.irq < 0) { + dev_err(dev, "Can't get spdif irq\n"); + return aiu->spdif.irq; + } + + ret = aiu_clk_get(dev); + if (ret) + return ret; + + /* Register the cpu component of the aiu */ + ret = snd_soc_register_component(dev, &aiu_cpu_component, + aiu_cpu_dai_drv, + ARRAY_SIZE(aiu_cpu_dai_drv)); + if (ret) + dev_err(dev, "Failed to register cpu component\n"); + + return ret; +} + +static int aiu_remove(struct platform_device *pdev) +{ + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static const struct of_device_id aiu_of_match[] = { + { .compatible = "amlogic,aiu-gxbb", }, + { .compatible = "amlogic,aiu-gxl", }, + {} +}; +MODULE_DEVICE_TABLE(of, aiu_of_match); + +static struct platform_driver aiu_pdrv = { + .probe = aiu_probe, + .remove = aiu_remove, + .driver = { + .name = "meson-aiu", + .of_match_table = aiu_of_match, + }, +}; +module_platform_driver(aiu_pdrv); + +MODULE_DESCRIPTION("Meson AIU Driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h new file mode 100644 index 00000000000000..a3488027b9d5c0 --- /dev/null +++ b/sound/soc/meson/aiu.h @@ -0,0 +1,82 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ +/* + * Copyright (c) 2018 BayLibre, SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_AIU_H +#define _MESON_AIU_H + +struct clk; +struct clk_bulk_data; +struct device; +struct of_phandle_args; +struct snd_soc_component_driver; +struct snd_soc_dai; +struct snd_soc_dai_driver; +struct snd_soc_dai_ops; + +enum aiu_clk_ids { + PCLK = 0, + AOCLK, + MCLK, + MIXER +}; + +struct aiu_interface { + struct clk_bulk_data *clks; + unsigned int clk_num; + unsigned int irq; +}; + +struct aiu { + struct clk *pclk; + struct clk *spdif_mclk; + struct aiu_interface i2s; + struct aiu_interface spdif; +}; + +#define AIU_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +int aiu_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name, + unsigned int component_id); + +int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai); +int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai); + +extern const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops; +extern const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops; +extern const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops; +extern const struct snd_soc_dai_ops aiu_encoder_spdif_dai_ops; + +#define AIU_IEC958_BPF 0x000 +#define AIU_958_MISC 0x010 +#define AIU_IEC958_DCU_FF_CTRL 0x01c +#define AIU_958_CHSTAT_L0 0x020 +#define AIU_958_CHSTAT_L1 0x024 +#define AIU_958_CTRL 0x028 +#define AIU_I2S_SOURCE_DESC 0x034 +#define AIU_I2S_DAC_CFG 0x040 +#define AIU_I2S_SYNC 0x044 +#define AIU_I2S_MISC 0x048 +#define AIU_RST_SOFT 0x054 +#define AIU_CLK_CTRL 0x058 +#define AIU_CLK_CTRL_MORE 0x064 +#define AIU_CODEC_DAC_LRCLK_CTRL 0x0a0 +#define AIU_HDMI_CLK_DATA_CTRL 0x0a8 +#define AIU_ACODEC_CTRL 0x0b0 +#define AIU_958_CHSTAT_R0 0x0c0 +#define AIU_958_CHSTAT_R1 0x0c4 +#define AIU_MEM_I2S_START 0x180 +#define AIU_MEM_I2S_MASKS 0x18c +#define AIU_MEM_I2S_CONTROL 0x190 +#define AIU_MEM_IEC958_START 0x194 +#define AIU_MEM_IEC958_CONTROL 0x1a4 +#define AIU_MEM_I2S_BUF_CNTL 0x1d8 +#define AIU_MEM_IEC958_BUF_CNTL 0x1fc + +#endif /* _MESON_AIU_H */ From b82b734c0e9a75e1b956214ac523a8eb590f51f3 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:55 +0100 Subject: [PATCH 088/415] ASoC: meson: aiu: add hdmi codec control support Add the codec to codec component which handles the routing between the audio producers (PCM and I2S) and the synopsys hdmi controller on the amlogic GX SoC family Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-6-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 2 + sound/soc/meson/Makefile | 1 + sound/soc/meson/aiu-codec-ctrl.c | 152 +++++++++++++++++++++++++++++++ sound/soc/meson/aiu.c | 34 ++++++- sound/soc/meson/aiu.h | 8 ++ 5 files changed, 196 insertions(+), 1 deletion(-) create mode 100644 sound/soc/meson/aiu-codec-ctrl.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index ca269dedfc7fdc..19de97ae4ce973 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -4,7 +4,9 @@ menu "ASoC support for Amlogic platforms" config SND_MESON_AIU tristate "Amlogic AIU" + select SND_MESON_CODEC_GLUE select SND_PCM_IEC958 + imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI help Select Y or M to add support for the Audio output subsystem found in the Amlogic GX SoC family diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index a7b79d717288f3..3b21f648e3223a 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -1,6 +1,7 @@ # SPDX-License-Identifier: (GPL-2.0 OR MIT) snd-soc-meson-aiu-objs := aiu.o +snd-soc-meson-aiu-objs += aiu-codec-ctrl.o snd-soc-meson-aiu-objs += aiu-encoder-i2s.o snd-soc-meson-aiu-objs += aiu-encoder-spdif.o snd-soc-meson-aiu-objs += aiu-fifo.o diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c new file mode 100644 index 00000000000000..8646a953e3b33f --- /dev/null +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -0,0 +1,152 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +#include +#include "aiu.h" +#include "meson-codec-glue.h" + +#define CTRL_CLK_SEL GENMASK(1, 0) +#define CTRL_DATA_SEL_SHIFT 4 +#define CTRL_DATA_SEL (0x3 << CTRL_DATA_SEL_SHIFT) + +static const char * const aiu_codec_ctrl_mux_texts[] = { + "DISABLED", "PCM", "I2S", +}; + +static int aiu_codec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL_DATA_SEL, + FIELD_PREP(CTRL_DATA_SEL, mux)); + + if (!changed) + return 0; + + /* Force disconnect of the mux while updating */ + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + /* Reset the source first */ + snd_soc_component_update_bits(component, e->reg, + CTRL_CLK_SEL | + CTRL_DATA_SEL, + FIELD_PREP(CTRL_CLK_SEL, 0) | + FIELD_PREP(CTRL_DATA_SEL, 0)); + + /* Set the appropriate source */ + snd_soc_component_update_bits(component, e->reg, + CTRL_CLK_SEL | + CTRL_DATA_SEL, + FIELD_PREP(CTRL_CLK_SEL, mux) | + FIELD_PREP(CTRL_DATA_SEL, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + +static SOC_ENUM_SINGLE_DECL(aiu_hdmi_ctrl_mux_enum, AIU_HDMI_CLK_DATA_CTRL, + CTRL_DATA_SEL_SHIFT, + aiu_codec_ctrl_mux_texts); + +static const struct snd_kcontrol_new aiu_hdmi_ctrl_mux = + SOC_DAPM_ENUM_EXT("HDMI Source", aiu_hdmi_ctrl_mux_enum, + snd_soc_dapm_get_enum_double, + aiu_codec_ctrl_mux_put_enum); + +static const struct snd_soc_dapm_widget aiu_hdmi_ctrl_widgets[] = { + SND_SOC_DAPM_MUX("HDMI CTRL SRC", SND_SOC_NOPM, 0, 0, + &aiu_hdmi_ctrl_mux), +}; + +static const struct snd_soc_dai_ops aiu_codec_ctrl_input_ops = { + .hw_params = meson_codec_glue_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, +}; + +static const struct snd_soc_dai_ops aiu_codec_ctrl_output_ops = { + .startup = meson_codec_glue_output_startup, +}; + +#define AIU_CODEC_CTRL_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define AIU_CODEC_CTRL_STREAM(xname, xsuffix) \ +{ \ + .stream_name = xname " " xsuffix, \ + .channels_min = 1, \ + .channels_max = 8, \ + .rate_min = 5512, \ + .rate_max = 192000, \ + .formats = AIU_CODEC_CTRL_FORMATS, \ +} + +#define AIU_CODEC_CTRL_INPUT(xname) { \ + .name = "CODEC CTRL " xname, \ + .playback = AIU_CODEC_CTRL_STREAM(xname, "Playback"), \ + .ops = &aiu_codec_ctrl_input_ops, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ +} + +#define AIU_CODEC_CTRL_OUTPUT(xname) { \ + .name = "CODEC CTRL " xname, \ + .capture = AIU_CODEC_CTRL_STREAM(xname, "Capture"), \ + .ops = &aiu_codec_ctrl_output_ops, \ +} + +static struct snd_soc_dai_driver aiu_hdmi_ctrl_dai_drv[] = { + [CTRL_I2S] = AIU_CODEC_CTRL_INPUT("HDMI I2S IN"), + [CTRL_PCM] = AIU_CODEC_CTRL_INPUT("HDMI PCM IN"), + [CTRL_OUT] = AIU_CODEC_CTRL_OUTPUT("HDMI OUT"), +}; + +static const struct snd_soc_dapm_route aiu_hdmi_ctrl_routes[] = { + { "HDMI CTRL SRC", "I2S", "HDMI I2S IN Playback" }, + { "HDMI CTRL SRC", "PCM", "HDMI PCM IN Playback" }, + { "HDMI OUT Capture", NULL, "HDMI CTRL SRC" }, +}; + +static int aiu_hdmi_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name) +{ + return aiu_of_xlate_dai_name(component, args, dai_name, AIU_HDMI); +} + +static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = { + .name = "AIU HDMI Codec Control", + .dapm_widgets = aiu_hdmi_ctrl_widgets, + .num_dapm_widgets = ARRAY_SIZE(aiu_hdmi_ctrl_widgets), + .dapm_routes = aiu_hdmi_ctrl_routes, + .num_dapm_routes = ARRAY_SIZE(aiu_hdmi_ctrl_routes), + .of_xlate_dai_name = aiu_hdmi_of_xlate_dai_name, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +int aiu_hdmi_ctrl_register_component(struct device *dev) +{ + return aiu_add_component(dev, &aiu_hdmi_ctrl_component, + aiu_hdmi_ctrl_dai_drv, + ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv), + "hdmi"); +} + diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index a62aced9b68717..b765dfb7072695 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -71,6 +71,26 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, return 0; } +int aiu_add_component(struct device *dev, + const struct snd_soc_component_driver *component_driver, + struct snd_soc_dai_driver *dai_drv, + int num_dai, + const char *debugfs_prefix) +{ + struct snd_soc_component *component; + + component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL); + if (!component) + return -ENOMEM; + +#ifdef CONFIG_DEBUG_FS + component->debugfs_prefix = debugfs_prefix; +#endif + + return snd_soc_add_component(dev, component, component_driver, + dai_drv, num_dai); +} + static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name) @@ -313,9 +333,21 @@ static int aiu_probe(struct platform_device *pdev) ret = snd_soc_register_component(dev, &aiu_cpu_component, aiu_cpu_dai_drv, ARRAY_SIZE(aiu_cpu_dai_drv)); - if (ret) + if (ret) { dev_err(dev, "Failed to register cpu component\n"); + return ret; + } + /* Register the hdmi codec control component */ + ret = aiu_hdmi_ctrl_register_component(dev); + if (ret) { + dev_err(dev, "Failed to register hdmi control component\n"); + goto err; + } + + return 0; +err: + snd_soc_unregister_component(dev); return ret; } diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index a3488027b9d5c0..9242ab1ab64b10 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -45,6 +45,14 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, const char **dai_name, unsigned int component_id); +int aiu_add_component(struct device *dev, + const struct snd_soc_component_driver *component_driver, + struct snd_soc_dai_driver *dai_drv, + int num_dai, + const char *debugfs_prefix); + +int aiu_hdmi_ctrl_register_component(struct device *dev); + int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai); int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai); From 65816025d46169973d308d83fbcf5c3981ed5621 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:56 +0100 Subject: [PATCH 089/415] ASoC: meson: aiu: add internal dac codec control support Add the codec to codec component which handles the routing between the audio producers and the internal audio DAC found on the amlogic GXL SoC family Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-7-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Makefile | 1 + sound/soc/meson/aiu-acodec-ctrl.c | 205 ++++++++++++++++++++++++++++++ sound/soc/meson/aiu.c | 10 ++ sound/soc/meson/aiu.h | 1 + 4 files changed, 217 insertions(+) create mode 100644 sound/soc/meson/aiu-acodec-ctrl.c diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 3b21f648e3223a..80f9113701b3a0 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -1,6 +1,7 @@ # SPDX-License-Identifier: (GPL-2.0 OR MIT) snd-soc-meson-aiu-objs := aiu.o +snd-soc-meson-aiu-objs += aiu-acodec-ctrl.o snd-soc-meson-aiu-objs += aiu-codec-ctrl.o snd-soc-meson-aiu-objs += aiu-encoder-i2s.o snd-soc-meson-aiu-objs += aiu-encoder-spdif.o diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c new file mode 100644 index 00000000000000..12d8a4d351a1f7 --- /dev/null +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -0,0 +1,205 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +#include +#include "aiu.h" +#include "meson-codec-glue.h" + +#define CTRL_DIN_EN 15 +#define CTRL_CLK_INV BIT(14) +#define CTRL_LRCLK_INV BIT(13) +#define CTRL_I2S_IN_BCLK_SRC BIT(11) +#define CTRL_DIN_LRCLK_SRC_SHIFT 6 +#define CTRL_DIN_LRCLK_SRC (0x3 << CTRL_DIN_LRCLK_SRC_SHIFT) +#define CTRL_BCLK_MCLK_SRC GENMASK(5, 4) +#define CTRL_DIN_SKEW GENMASK(3, 2) +#define CTRL_I2S_OUT_LANE_SRC 0 + +#define AIU_ACODEC_OUT_CHMAX 2 + +static const char * const aiu_acodec_ctrl_mux_texts[] = { + "DISABLED", "I2S", "PCM", +}; + +static int aiu_acodec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL_DIN_LRCLK_SRC, + FIELD_PREP(CTRL_DIN_LRCLK_SRC, + mux)); + + if (!changed) + return 0; + + /* Force disconnect of the mux while updating */ + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + snd_soc_component_update_bits(component, e->reg, + CTRL_DIN_LRCLK_SRC | + CTRL_BCLK_MCLK_SRC, + FIELD_PREP(CTRL_DIN_LRCLK_SRC, mux) | + FIELD_PREP(CTRL_BCLK_MCLK_SRC, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + +static SOC_ENUM_SINGLE_DECL(aiu_acodec_ctrl_mux_enum, AIU_ACODEC_CTRL, + CTRL_DIN_LRCLK_SRC_SHIFT, + aiu_acodec_ctrl_mux_texts); + +static const struct snd_kcontrol_new aiu_acodec_ctrl_mux = + SOC_DAPM_ENUM_EXT("ACodec Source", aiu_acodec_ctrl_mux_enum, + snd_soc_dapm_get_enum_double, + aiu_acodec_ctrl_mux_put_enum); + +static const struct snd_kcontrol_new aiu_acodec_ctrl_out_enable = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", AIU_ACODEC_CTRL, + CTRL_DIN_EN, 1, 0); + +static const struct snd_soc_dapm_widget aiu_acodec_ctrl_widgets[] = { + SND_SOC_DAPM_MUX("ACODEC SRC", SND_SOC_NOPM, 0, 0, + &aiu_acodec_ctrl_mux), + SND_SOC_DAPM_SWITCH("ACODEC OUT EN", SND_SOC_NOPM, 0, 0, + &aiu_acodec_ctrl_out_enable), +}; + +static int aiu_acodec_ctrl_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data; + int ret; + + ret = meson_codec_glue_input_hw_params(substream, params, dai); + if (ret) + return ret; + + /* The glue will provide 1 lane out of the 4 to the output */ + data = meson_codec_glue_input_get_data(dai); + data->params.channels_min = min_t(unsigned int, AIU_ACODEC_OUT_CHMAX, + data->params.channels_min); + data->params.channels_max = min_t(unsigned int, AIU_ACODEC_OUT_CHMAX, + data->params.channels_max); + + return 0; +} + +static const struct snd_soc_dai_ops aiu_acodec_ctrl_input_ops = { + .hw_params = aiu_acodec_ctrl_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, +}; + +static const struct snd_soc_dai_ops aiu_acodec_ctrl_output_ops = { + .startup = meson_codec_glue_output_startup, +}; + +#define AIU_ACODEC_CTRL_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define AIU_ACODEC_STREAM(xname, xsuffix, xchmax) \ +{ \ + .stream_name = xname " " xsuffix, \ + .channels_min = 1, \ + .channels_max = (xchmax), \ + .rate_min = 5512, \ + .rate_max = 192000, \ + .formats = AIU_ACODEC_CTRL_FORMATS, \ +} + +#define AIU_ACODEC_INPUT(xname) { \ + .name = "ACODEC CTRL " xname, \ + .name = xname, \ + .playback = AIU_ACODEC_STREAM(xname, "Playback", 8), \ + .ops = &aiu_acodec_ctrl_input_ops, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ +} + +#define AIU_ACODEC_OUTPUT(xname) { \ + .name = "ACODEC CTRL " xname, \ + .capture = AIU_ACODEC_STREAM(xname, "Capture", AIU_ACODEC_OUT_CHMAX), \ + .ops = &aiu_acodec_ctrl_output_ops, \ +} + +static struct snd_soc_dai_driver aiu_acodec_ctrl_dai_drv[] = { + [CTRL_I2S] = AIU_ACODEC_INPUT("ACODEC I2S IN"), + [CTRL_PCM] = AIU_ACODEC_INPUT("ACODEC PCM IN"), + [CTRL_OUT] = AIU_ACODEC_OUTPUT("ACODEC OUT"), +}; + +static const struct snd_soc_dapm_route aiu_acodec_ctrl_routes[] = { + { "ACODEC SRC", "I2S", "ACODEC I2S IN Playback" }, + { "ACODEC SRC", "PCM", "ACODEC PCM IN Playback" }, + { "ACODEC OUT EN", "Switch", "ACODEC SRC" }, + { "ACODEC OUT Capture", NULL, "ACODEC OUT EN" }, +}; + +static const struct snd_kcontrol_new aiu_acodec_ctrl_controls[] = { + SOC_SINGLE("ACODEC I2S Lane Select", AIU_ACODEC_CTRL, + CTRL_I2S_OUT_LANE_SRC, 3, 0), +}; + +static int aiu_acodec_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name) +{ + return aiu_of_xlate_dai_name(component, args, dai_name, AIU_ACODEC); +} + +static int aiu_acodec_ctrl_component_probe(struct snd_soc_component *component) +{ + /* + * NOTE: Din Skew setting + * According to the documentation, the following update adds one delay + * to the din line. Without this, the output saturates. This happens + * regardless of the link format (i2s or left_j) so it is not clear what + * it actually does but it seems to be required + */ + snd_soc_component_update_bits(component, AIU_ACODEC_CTRL, + CTRL_DIN_SKEW, + FIELD_PREP(CTRL_DIN_SKEW, 2)); + + return 0; +} + +static const struct snd_soc_component_driver aiu_acodec_ctrl_component = { + .name = "AIU Internal DAC Codec Control", + .probe = aiu_acodec_ctrl_component_probe, + .controls = aiu_acodec_ctrl_controls, + .num_controls = ARRAY_SIZE(aiu_acodec_ctrl_controls), + .dapm_widgets = aiu_acodec_ctrl_widgets, + .num_dapm_widgets = ARRAY_SIZE(aiu_acodec_ctrl_widgets), + .dapm_routes = aiu_acodec_ctrl_routes, + .num_dapm_routes = ARRAY_SIZE(aiu_acodec_ctrl_routes), + .of_xlate_dai_name = aiu_acodec_of_xlate_dai_name, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +int aiu_acodec_ctrl_register_component(struct device *dev) +{ + return aiu_add_component(dev, &aiu_acodec_ctrl_component, + aiu_acodec_ctrl_dai_drv, + ARRAY_SIZE(aiu_acodec_ctrl_dai_drv), + "acodec"); +} diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index b765dfb7072695..5c4845a23a34f5 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -345,6 +345,16 @@ static int aiu_probe(struct platform_device *pdev) goto err; } + /* Register the internal dac control component on gxl */ + if (of_device_is_compatible(dev->of_node, "amlogic,aiu-gxl")) { + ret = aiu_acodec_ctrl_register_component(dev); + if (ret) { + dev_err(dev, + "Failed to register acodec control component\n"); + goto err; + } + } + return 0; err: snd_soc_unregister_component(dev); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 9242ab1ab64b10..a65a576e34005b 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -52,6 +52,7 @@ int aiu_add_component(struct device *dev, const char *debugfs_prefix); int aiu_hdmi_ctrl_register_component(struct device *dev); +int aiu_acodec_ctrl_register_component(struct device *dev); int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai); int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai); From aa9c3b7273a58b5d9b2c1161b76b5fc8ea8c159b Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:57 +0100 Subject: [PATCH 090/415] ASoC: meson: axg: extract sound card utils This prepares the addition of the GX SoC family sound card driver. The GX sound card, while slightly different, will be similar to the AXG one. The purpose of this change is to share the utils common to both sound card driver. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-8-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 4 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 403 ++--------------------------- sound/soc/meson/meson-card-utils.c | 385 +++++++++++++++++++++++++++ sound/soc/meson/meson-card.h | 55 ++++ 5 files changed, 473 insertions(+), 376 deletions(-) create mode 100644 sound/soc/meson/meson-card-utils.c create mode 100644 sound/soc/meson/meson-card.h diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 19de97ae4ce973..347fa78e309a4c 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -59,6 +59,7 @@ config SND_MESON_AXG_TDMOUT config SND_MESON_AXG_SOUND_CARD tristate "Amlogic AXG Sound Card Support" select SND_MESON_AXG_TDM_INTERFACE + select SND_MESON_CARD_UTILS imply SND_MESON_AXG_FRDDR imply SND_MESON_AXG_TODDR imply SND_MESON_AXG_TDMIN @@ -94,6 +95,9 @@ config SND_MESON_AXG_PDM Select Y or M to add support for PDM input embedded in the Amlogic AXG SoC family +config SND_MESON_CARD_UTILS + tristate + config SND_MESON_CODEC_GLUE tristate diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 80f9113701b3a0..bef2b72fd7a730 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -19,6 +19,7 @@ snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifin-objs := axg-spdifin.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o snd-soc-meson-axg-pdm-objs := axg-pdm.o +snd-soc-meson-card-utils-objs := meson-card-utils.o snd-soc-meson-codec-glue-objs := meson-codec-glue.o snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o @@ -34,5 +35,6 @@ obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o +obj-$(CONFIG_SND_MESON_CARD_UTILS) += snd-soc-meson-card-utils.o obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 1f698adde506c9..372dc696cc8e9b 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -9,11 +9,7 @@ #include #include "axg-tdm.h" - -struct axg_card { - struct snd_soc_card card; - void **link_data; -}; +#include "meson-card.h" struct axg_dai_link_tdm_mask { u32 tx; @@ -41,161 +37,15 @@ static const struct snd_soc_pcm_stream codec_params = { .channels_max = 8, }; -#define PREFIX "amlogic," - -static int axg_card_reallocate_links(struct axg_card *priv, - unsigned int num_links) -{ - struct snd_soc_dai_link *links; - void **ldata; - - links = krealloc(priv->card.dai_link, - num_links * sizeof(*priv->card.dai_link), - GFP_KERNEL | __GFP_ZERO); - ldata = krealloc(priv->link_data, - num_links * sizeof(*priv->link_data), - GFP_KERNEL | __GFP_ZERO); - - if (!links || !ldata) { - dev_err(priv->card.dev, "failed to allocate links\n"); - return -ENOMEM; - } - - priv->card.dai_link = links; - priv->link_data = ldata; - priv->card.num_links = num_links; - return 0; -} - -static int axg_card_parse_dai(struct snd_soc_card *card, - struct device_node *node, - struct device_node **dai_of_node, - const char **dai_name) -{ - struct of_phandle_args args; - int ret; - - if (!dai_name || !dai_of_node || !node) - return -EINVAL; - - ret = of_parse_phandle_with_args(node, "sound-dai", - "#sound-dai-cells", 0, &args); - if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(card->dev, "can't parse dai %d\n", ret); - return ret; - } - *dai_of_node = args.np; - - return snd_soc_get_dai_name(&args, dai_name); -} - -static int axg_card_set_link_name(struct snd_soc_card *card, - struct snd_soc_dai_link *link, - struct device_node *node, - const char *prefix) -{ - char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", - prefix, node->full_name); - if (!name) - return -ENOMEM; - - link->name = name; - link->stream_name = name; - - return 0; -} - -static void axg_card_clean_references(struct axg_card *priv) -{ - struct snd_soc_card *card = &priv->card; - struct snd_soc_dai_link *link; - struct snd_soc_dai_link_component *codec; - struct snd_soc_aux_dev *aux; - int i, j; - - if (card->dai_link) { - for_each_card_prelinks(card, i, link) { - if (link->cpus) - of_node_put(link->cpus->of_node); - for_each_link_codecs(link, j, codec) - of_node_put(codec->of_node); - } - } - - if (card->aux_dev) { - for_each_card_pre_auxs(card, i, aux) - of_node_put(aux->dlc.of_node); - } - - kfree(card->dai_link); - kfree(priv->link_data); -} - -static int axg_card_add_aux_devices(struct snd_soc_card *card) -{ - struct device_node *node = card->dev->of_node; - struct snd_soc_aux_dev *aux; - int num, i; - - num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); - if (num == -ENOENT) { - /* - * It is ok to have no auxiliary devices but for this card it - * is a strange situtation. Let's warn the about it. - */ - dev_warn(card->dev, "card has no auxiliary devices\n"); - return 0; - } else if (num < 0) { - dev_err(card->dev, "error getting auxiliary devices: %d\n", - num); - return num; - } - - aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); - if (!aux) - return -ENOMEM; - card->aux_dev = aux; - card->num_aux_devs = num; - - for_each_card_pre_auxs(card, i, aux) { - aux->dlc.of_node = - of_parse_phandle(node, "audio-aux-devs", i); - if (!aux->dlc.of_node) - return -EINVAL; - } - - return 0; -} - static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; - struct snd_soc_dai *codec_dai; - unsigned int mclk; - int ret, i; - - if (be->mclk_fs) { - mclk = params_rate(params) * be->mclk_fs; - - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) - return ret; - } - - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret && ret != -ENOTSUPP) - return ret; - } - return 0; + return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs); } static const struct snd_soc_ops axg_card_tdm_be_ops = { @@ -204,7 +54,7 @@ static const struct snd_soc_ops axg_card_tdm_be_ops = { static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; struct snd_soc_dai *codec_dai; @@ -234,7 +84,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) { - struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; int ret; @@ -253,14 +103,14 @@ static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) static int axg_card_add_tdm_loopback(struct snd_soc_card *card, int *index) { - struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct meson_card *priv = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *pad = &card->dai_link[*index]; struct snd_soc_dai_link *lb; struct snd_soc_dai_link_component *dlc; int ret; /* extend links */ - ret = axg_card_reallocate_links(priv, card->num_links + 1); + ret = meson_card_reallocate_links(card, card->num_links + 1); if (ret) return ret; @@ -304,32 +154,6 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, return 0; } -static unsigned int axg_card_parse_daifmt(struct device_node *node, - struct device_node *cpu_node) -{ - struct device_node *bitclkmaster = NULL; - struct device_node *framemaster = NULL; - unsigned int daifmt; - - daifmt = snd_soc_of_parse_daifmt(node, PREFIX, - &bitclkmaster, &framemaster); - daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; - - /* If no master is provided, default to cpu master */ - if (!bitclkmaster || bitclkmaster == cpu_node) { - daifmt |= (!framemaster || framemaster == cpu_node) ? - SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM; - } else { - daifmt |= (!framemaster || framemaster == cpu_node) ? - SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM; - } - - of_node_put(bitclkmaster); - of_node_put(framemaster); - - return daifmt; -} - static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card, struct snd_soc_dai_link *link, struct device_node *node, @@ -424,7 +248,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card, struct device_node *node, int *index) { - struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct meson_card *priv = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *link = &card->dai_link[*index]; struct axg_dai_link_tdm_data *be; int ret; @@ -438,7 +262,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card, /* Setup tdm link */ link->ops = &axg_card_tdm_be_ops; link->init = axg_card_tdm_dai_init; - link->dai_fmt = axg_card_parse_daifmt(node, link->cpus->of_node); + link->dai_fmt = meson_card_parse_daifmt(node, link->cpus->of_node); of_property_read_u32(node, "mclk-fs", &be->mclk_fs); @@ -462,97 +286,24 @@ static int axg_card_parse_tdm(struct snd_soc_card *card, return 0; } -static int axg_card_set_be_link(struct snd_soc_card *card, - struct snd_soc_dai_link *link, - struct device_node *node) -{ - struct snd_soc_dai_link_component *codec; - struct device_node *np; - int ret, num_codecs; - - link->no_pcm = 1; - link->dpcm_playback = 1; - link->dpcm_capture = 1; - - num_codecs = of_get_child_count(node); - if (!num_codecs) { - dev_err(card->dev, "be link %s has no codec\n", - node->full_name); - return -EINVAL; - } - - codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL); - if (!codec) - return -ENOMEM; - - link->codecs = codec; - link->num_codecs = num_codecs; - - for_each_child_of_node(node, np) { - ret = axg_card_parse_dai(card, np, &codec->of_node, - &codec->dai_name); - if (ret) { - of_node_put(np); - return ret; - } - - codec++; - } - - ret = axg_card_set_link_name(card, link, node, "be"); - if (ret) - dev_err(card->dev, "error setting %pOFn link name\n", np); - - return ret; -} - -static int axg_card_set_fe_link(struct snd_soc_card *card, - struct snd_soc_dai_link *link, - struct device_node *node, - bool is_playback) -{ - struct snd_soc_dai_link_component *codec; - - codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL); - if (!codec) - return -ENOMEM; - - link->codecs = codec; - link->num_codecs = 1; - - link->dynamic = 1; - link->dpcm_merged_format = 1; - link->dpcm_merged_chan = 1; - link->dpcm_merged_rate = 1; - link->codecs->dai_name = "snd-soc-dummy-dai"; - link->codecs->name = "snd-soc-dummy"; - - if (is_playback) - link->dpcm_playback = 1; - else - link->dpcm_capture = 1; - - return axg_card_set_link_name(card, link, node, "fe"); -} - static int axg_card_cpu_is_capture_fe(struct device_node *np) { - return of_device_is_compatible(np, PREFIX "axg-toddr"); + return of_device_is_compatible(np, DT_PREFIX "axg-toddr"); } static int axg_card_cpu_is_playback_fe(struct device_node *np) { - return of_device_is_compatible(np, PREFIX "axg-frddr"); + return of_device_is_compatible(np, DT_PREFIX "axg-frddr"); } static int axg_card_cpu_is_tdm_iface(struct device_node *np) { - return of_device_is_compatible(np, PREFIX "axg-tdm-iface"); + return of_device_is_compatible(np, DT_PREFIX "axg-tdm-iface"); } static int axg_card_cpu_is_codec(struct device_node *np) { - return of_device_is_compatible(np, PREFIX "g12a-tohdmitx"); + return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx"); } static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, @@ -569,17 +320,17 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, dai_link->cpus = cpu; dai_link->num_cpus = 1; - ret = axg_card_parse_dai(card, np, &dai_link->cpus->of_node, - &dai_link->cpus->dai_name); + ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node, + &dai_link->cpus->dai_name); if (ret) return ret; if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node)) - ret = axg_card_set_fe_link(card, dai_link, np, true); + ret = meson_card_set_fe_link(card, dai_link, np, true); else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node)) - ret = axg_card_set_fe_link(card, dai_link, np, false); + ret = meson_card_set_fe_link(card, dai_link, np, false); else - ret = axg_card_set_be_link(card, dai_link, np); + ret = meson_card_set_be_link(card, dai_link, np); if (ret) return ret; @@ -592,121 +343,21 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, return ret; } -static int axg_card_add_links(struct snd_soc_card *card) -{ - struct axg_card *priv = snd_soc_card_get_drvdata(card); - struct device_node *node = card->dev->of_node; - struct device_node *np; - int num, i, ret; - - num = of_get_child_count(node); - if (!num) { - dev_err(card->dev, "card has no links\n"); - return -EINVAL; - } - - ret = axg_card_reallocate_links(priv, num); - if (ret) - return ret; - - i = 0; - for_each_child_of_node(node, np) { - ret = axg_card_add_link(card, np, &i); - if (ret) { - of_node_put(np); - return ret; - } - - i++; - } - - return 0; -} - -static int axg_card_parse_of_optional(struct snd_soc_card *card, - const char *propname, - int (*func)(struct snd_soc_card *c, - const char *p)) -{ - /* If property is not provided, don't fail ... */ - if (!of_property_read_bool(card->dev->of_node, propname)) - return 0; - - /* ... but do fail if it is provided and the parsing fails */ - return func(card, propname); -} +static const struct meson_card_match_data axg_card_match_data = { + .add_link = axg_card_add_link, +}; static const struct of_device_id axg_card_of_match[] = { - { .compatible = "amlogic,axg-sound-card", }, - {} + { + .compatible = "amlogic,axg-sound-card", + .data = &axg_card_match_data, + }, {} }; MODULE_DEVICE_TABLE(of, axg_card_of_match); -static int axg_card_probe(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - struct axg_card *priv; - int ret; - - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - platform_set_drvdata(pdev, priv); - snd_soc_card_set_drvdata(&priv->card, priv); - - priv->card.owner = THIS_MODULE; - priv->card.dev = dev; - - ret = snd_soc_of_parse_card_name(&priv->card, "model"); - if (ret < 0) - return ret; - - ret = axg_card_parse_of_optional(&priv->card, "audio-routing", - snd_soc_of_parse_audio_routing); - if (ret) { - dev_err(dev, "error while parsing routing\n"); - return ret; - } - - ret = axg_card_parse_of_optional(&priv->card, "audio-widgets", - snd_soc_of_parse_audio_simple_widgets); - if (ret) { - dev_err(dev, "error while parsing widgets\n"); - return ret; - } - - ret = axg_card_add_links(&priv->card); - if (ret) - goto out_err; - - ret = axg_card_add_aux_devices(&priv->card); - if (ret) - goto out_err; - - ret = devm_snd_soc_register_card(dev, &priv->card); - if (ret) - goto out_err; - - return 0; - -out_err: - axg_card_clean_references(priv); - return ret; -} - -static int axg_card_remove(struct platform_device *pdev) -{ - struct axg_card *priv = platform_get_drvdata(pdev); - - axg_card_clean_references(priv); - - return 0; -} - static struct platform_driver axg_card_pdrv = { - .probe = axg_card_probe, - .remove = axg_card_remove, + .probe = meson_card_probe, + .remove = meson_card_remove, .driver = { .name = "axg-sound-card", .of_match_table = axg_card_of_match, diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c new file mode 100644 index 00000000000000..a70d244ef88b6b --- /dev/null +++ b/sound/soc/meson/meson-card-utils.c @@ -0,0 +1,385 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include + +#include "meson-card.h" + +int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + unsigned int mclk_fs) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + unsigned int mclk; + int ret, i; + + if (!mclk_fs) + return 0; + + mclk = params_rate(params) * mclk_fs; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + return ret; + + return 0; +} +EXPORT_SYMBOL_GPL(meson_card_i2s_set_sysclk); + +int meson_card_reallocate_links(struct snd_soc_card *card, + unsigned int num_links) +{ + struct meson_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *links; + void **ldata; + + links = krealloc(priv->card.dai_link, + num_links * sizeof(*priv->card.dai_link), + GFP_KERNEL | __GFP_ZERO); + ldata = krealloc(priv->link_data, + num_links * sizeof(*priv->link_data), + GFP_KERNEL | __GFP_ZERO); + + if (!links || !ldata) { + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } + + priv->card.dai_link = links; + priv->link_data = ldata; + priv->card.num_links = num_links; + return 0; +} +EXPORT_SYMBOL_GPL(meson_card_reallocate_links); + +int meson_card_parse_dai(struct snd_soc_card *card, + struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name) +{ + struct of_phandle_args args; + int ret; + + if (!dai_name || !dai_of_node || !node) + return -EINVAL; + + ret = of_parse_phandle_with_args(node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "can't parse dai %d\n", ret); + return ret; + } + *dai_of_node = args.np; + + return snd_soc_get_dai_name(&args, dai_name); +} +EXPORT_SYMBOL_GPL(meson_card_parse_dai); + +static int meson_card_set_link_name(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + const char *prefix) +{ + char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", + prefix, node->full_name); + if (!name) + return -ENOMEM; + + link->name = name; + link->stream_name = name; + + return 0; +} + +unsigned int meson_card_parse_daifmt(struct device_node *node, + struct device_node *cpu_node) +{ + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + unsigned int daifmt; + + daifmt = snd_soc_of_parse_daifmt(node, DT_PREFIX, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + + /* If no master is provided, default to cpu master */ + if (!bitclkmaster || bitclkmaster == cpu_node) { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM; + } else { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + + return daifmt; +} +EXPORT_SYMBOL_GPL(meson_card_parse_daifmt); + +int meson_card_set_be_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node) +{ + struct snd_soc_dai_link_component *codec; + struct device_node *np; + int ret, num_codecs; + + link->no_pcm = 1; + link->dpcm_playback = 1; + link->dpcm_capture = 1; + + num_codecs = of_get_child_count(node); + if (!num_codecs) { + dev_err(card->dev, "be link %s has no codec\n", + node->full_name); + return -EINVAL; + } + + codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = num_codecs; + + for_each_child_of_node(node, np) { + ret = meson_card_parse_dai(card, np, &codec->of_node, + &codec->dai_name); + if (ret) { + of_node_put(np); + return ret; + } + + codec++; + } + + ret = meson_card_set_link_name(card, link, node, "be"); + if (ret) + dev_err(card->dev, "error setting %pOFn link name\n", np); + + return ret; +} +EXPORT_SYMBOL_GPL(meson_card_set_be_link); + +int meson_card_set_fe_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + bool is_playback) +{ + struct snd_soc_dai_link_component *codec; + + codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = 1; + + link->dynamic = 1; + link->dpcm_merged_format = 1; + link->dpcm_merged_chan = 1; + link->dpcm_merged_rate = 1; + link->codecs->dai_name = "snd-soc-dummy-dai"; + link->codecs->name = "snd-soc-dummy"; + + if (is_playback) + link->dpcm_playback = 1; + else + link->dpcm_capture = 1; + + return meson_card_set_link_name(card, link, node, "fe"); +} +EXPORT_SYMBOL_GPL(meson_card_set_fe_link); + +static int meson_card_add_links(struct snd_soc_card *card) +{ + struct meson_card *priv = snd_soc_card_get_drvdata(card); + struct device_node *node = card->dev->of_node; + struct device_node *np; + int num, i, ret; + + num = of_get_child_count(node); + if (!num) { + dev_err(card->dev, "card has no links\n"); + return -EINVAL; + } + + ret = meson_card_reallocate_links(card, num); + if (ret) + return ret; + + i = 0; + for_each_child_of_node(node, np) { + ret = priv->match_data->add_link(card, np, &i); + if (ret) { + of_node_put(np); + return ret; + } + + i++; + } + + return 0; +} + +static int meson_card_parse_of_optional(struct snd_soc_card *card, + const char *propname, + int (*func)(struct snd_soc_card *c, + const char *p)) +{ + /* If property is not provided, don't fail ... */ + if (!of_property_read_bool(card->dev->of_node, propname)) + return 0; + + /* ... but do fail if it is provided and the parsing fails */ + return func(card, propname); +} + +static int meson_card_add_aux_devices(struct snd_soc_card *card) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); + if (num == -ENOENT) { + return 0; + } else if (num < 0) { + dev_err(card->dev, "error getting auxiliary devices: %d\n", + num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for_each_card_pre_auxs(card, i, aux) { + aux->dlc.of_node = + of_parse_phandle(node, "audio-aux-devs", i); + if (!aux->dlc.of_node) + return -EINVAL; + } + + return 0; +} + +static void meson_card_clean_references(struct meson_card *priv) +{ + struct snd_soc_card *card = &priv->card; + struct snd_soc_dai_link *link; + struct snd_soc_dai_link_component *codec; + struct snd_soc_aux_dev *aux; + int i, j; + + if (card->dai_link) { + for_each_card_prelinks(card, i, link) { + if (link->cpus) + of_node_put(link->cpus->of_node); + for_each_link_codecs(link, j, codec) + of_node_put(codec->of_node); + } + } + + if (card->aux_dev) { + for_each_card_pre_auxs(card, i, aux) + of_node_put(aux->dlc.of_node); + } + + kfree(card->dai_link); + kfree(priv->link_data); +} + +int meson_card_probe(struct platform_device *pdev) +{ + const struct meson_card_match_data *data; + struct device *dev = &pdev->dev; + struct meson_card *priv; + int ret; + + data = of_device_get_match_data(dev); + if (!data) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + priv->card.owner = THIS_MODULE; + priv->card.dev = dev; + priv->match_data = data; + + ret = snd_soc_of_parse_card_name(&priv->card, "model"); + if (ret < 0) + return ret; + + ret = meson_card_parse_of_optional(&priv->card, "audio-routing", + snd_soc_of_parse_audio_routing); + if (ret) { + dev_err(dev, "error while parsing routing\n"); + return ret; + } + + ret = meson_card_parse_of_optional(&priv->card, "audio-widgets", + snd_soc_of_parse_audio_simple_widgets); + if (ret) { + dev_err(dev, "error while parsing widgets\n"); + return ret; + } + + ret = meson_card_add_links(&priv->card); + if (ret) + goto out_err; + + ret = meson_card_add_aux_devices(&priv->card); + if (ret) + goto out_err; + + ret = devm_snd_soc_register_card(dev, &priv->card); + if (ret) + goto out_err; + + return 0; + +out_err: + meson_card_clean_references(priv); + return ret; +} +EXPORT_SYMBOL_GPL(meson_card_probe); + +int meson_card_remove(struct platform_device *pdev) +{ + struct meson_card *priv = platform_get_drvdata(pdev); + + meson_card_clean_references(priv); + + return 0; +} +EXPORT_SYMBOL_GPL(meson_card_remove); + +MODULE_DESCRIPTION("Amlogic Sound Card Utils"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/meson-card.h b/sound/soc/meson/meson-card.h new file mode 100644 index 00000000000000..74314071c80db4 --- /dev/null +++ b/sound/soc/meson/meson-card.h @@ -0,0 +1,55 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2020 BayLibre, SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_SND_CARD_H +#define _MESON_SND_CARD_H + +struct device_node; +struct platform_device; + +struct snd_soc_card; +struct snd_pcm_substream; +struct snd_pcm_hw_params; + +#define DT_PREFIX "amlogic," + +struct meson_card_match_data { + int (*add_link)(struct snd_soc_card *card, + struct device_node *node, + int *index); +}; + +struct meson_card { + const struct meson_card_match_data *match_data; + struct snd_soc_card card; + void **link_data; +}; + +unsigned int meson_card_parse_daifmt(struct device_node *node, + struct device_node *cpu_node); + +int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + unsigned int mclk_fs); + +int meson_card_reallocate_links(struct snd_soc_card *card, + unsigned int num_links); +int meson_card_parse_dai(struct snd_soc_card *card, + struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name); +int meson_card_set_be_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node); +int meson_card_set_fe_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + bool is_playback); + +int meson_card_probe(struct platform_device *pdev); +int meson_card_remove(struct platform_device *pdev); + +#endif /* _MESON_SND_CARD_H */ From fd00366b8e4125d29e32d49053a702ddf77430f6 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:58 +0100 Subject: [PATCH 091/415] ASoC: meson: gx: add sound card dt-binding documentation Add the dt-binding documentation of sound card supporting the amlogic GX SoC family Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-9-jbrunet@baylibre.com Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,gx-sound-card.yaml | 113 ++++++++++++++++++ 1 file changed, 113 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml new file mode 100644 index 00000000000000..fb374c659be1a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -0,0 +1,113 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,gx-sound-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic GX sound card + +maintainers: + - Jerome Brunet + +properties: + compatible: + items: + - const: amlogic,gx-sound-card + + audio-aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: list of auxiliary devices + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + minItems: 2 + description: |- + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + minItems: 2 + description: |- + A list off component DAPM widget. Each entry is a pair of strings, + the first being the widget type, the second being the widget name + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +patternProperties: + "^dai-link-[0-9]+$": + type: object + description: |- + dai-link child nodes: + Container for dai-link level properties and the CODEC sub-nodes. + There should be at least one (and probably more) subnode of this type + + properties: + dai-format: + $ref: /schemas/types.yaml#/definitions/string + enum: [ i2s, left-j, dsp_a ] + + mclk-fs: + $ref: /schemas/types.yaml#/definitions/uint32 + description: |- + Multiplication factor between the frame rate and master clock + rate + + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the CPU DAI + + patternProperties: + "^codec-[0-9]+$": + type: object + description: |- + Codecs: + dai-link representing backend links should have at least one subnode. + One subnode for each codec of the dai-link. dai-link representing + frontend links have no codec, therefore have no subnodes + + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the codec DAI + + required: + - sound-dai + + required: + - sound-dai + +required: + - model + - dai-link-0 + +examples: + - | + sound { + compatible = "amlogic,gx-sound-card"; + model = "GXL-ACME-S905X-FOO"; + audio-aux-devs = <&>; + audio-routing = "I2S ENCODER I2S IN", "I2S FIFO Playback"; + + dai-link-0 { + sound-dai = <&i2s_fifo>; + }; + + dai-link-1 { + sound-dai = <&i2s_encoder>; + dai-format = "i2s"; + mclk-fs = <256>; + + codec-0 { + sound-dai = <&codec0>; + }; + + codec-1 { + sound-dai = <&codec1>; + }; + }; + }; + From e37a0c313a0f8ba0b8de9c30db98fbc77bd8d446 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 13 Feb 2020 16:51:59 +0100 Subject: [PATCH 092/415] ASoC: meson: gx: add sound card support Add support for the sound card used on the amlogic GX SoC family Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200213155159.3235792-10-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 7 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/gx-card.c | 141 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 150 insertions(+) create mode 100644 sound/soc/meson/gx-card.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 347fa78e309a4c..22d2af75b59e37 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -101,6 +101,13 @@ config SND_MESON_CARD_UTILS config SND_MESON_CODEC_GLUE tristate +config SND_MESON_GX_SOUND_CARD + tristate "Amlogic GX Sound Card Support" + select SND_MESON_CARD_UTILS + imply SND_MESON_AIU + help + Select Y or M to add support for the GXBB/GXL SoC sound card + config SND_MESON_G12A_TOHDMITX tristate "Amlogic G12A To HDMI TX Control Support" select REGMAP_MMIO diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index bef2b72fd7a730..f9c90c39149819 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -21,6 +21,7 @@ snd-soc-meson-axg-spdifout-objs := axg-spdifout.o snd-soc-meson-axg-pdm-objs := axg-pdm.o snd-soc-meson-card-utils-objs := meson-card-utils.o snd-soc-meson-codec-glue-objs := meson-codec-glue.o +snd-soc-meson-gx-sound-card-objs := gx-card.o snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o obj-$(CONFIG_SND_MESON_AIU) += snd-soc-meson-aiu.o @@ -37,4 +38,5 @@ obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o obj-$(CONFIG_SND_MESON_CARD_UTILS) += snd-soc-meson-card-utils.o obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o +obj-$(CONFIG_SND_MESON_GX_SOUND_CARD) += snd-soc-meson-gx-sound-card.o obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c new file mode 100644 index 00000000000000..7b01dcb73e5e1c --- /dev/null +++ b/sound/soc/meson/gx-card.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +#include "meson-card.h" + +struct gx_dai_link_i2s_data { + unsigned int mclk_fs; +}; + +/* + * Base params for the codec to codec links + * Those will be over-written by the CPU side of the link + */ +static const struct snd_soc_pcm_stream codec_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 5525, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 8, +}; + +static int gx_card_i2s_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct gx_dai_link_i2s_data *be = + (struct gx_dai_link_i2s_data *)priv->link_data[rtd->num]; + + return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs); +} + +static const struct snd_soc_ops gx_card_i2s_be_ops = { + .hw_params = gx_card_i2s_be_hw_params, +}; + +static int gx_card_parse_i2s(struct snd_soc_card *card, + struct device_node *node, + int *index) +{ + struct meson_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *link = &card->dai_link[*index]; + struct gx_dai_link_i2s_data *be; + + /* Allocate i2s link parameters */ + be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL); + if (!be) + return -ENOMEM; + priv->link_data[*index] = be; + + /* Setup i2s link */ + link->ops = &gx_card_i2s_be_ops; + link->dai_fmt = meson_card_parse_daifmt(node, link->cpus->of_node); + + of_property_read_u32(node, "mclk-fs", &be->mclk_fs); + + return 0; +} + +static int gx_card_cpu_identify(struct snd_soc_dai_link_component *c, + char *match) +{ + if (of_device_is_compatible(c->of_node, DT_PREFIX "aiu")) { + if (strstr(c->dai_name, match)) + return 1; + } + + /* dai not matched */ + return 0; +} + +static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np, + int *index) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[*index]; + struct snd_soc_dai_link_component *cpu; + int ret; + + cpu = devm_kzalloc(card->dev, sizeof(*cpu), GFP_KERNEL); + if (!cpu) + return -ENOMEM; + + dai_link->cpus = cpu; + dai_link->num_cpus = 1; + + ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node, + &dai_link->cpus->dai_name); + if (ret) + return ret; + + if (gx_card_cpu_identify(dai_link->cpus, "FIFO")) + ret = meson_card_set_fe_link(card, dai_link, np, true); + else + ret = meson_card_set_be_link(card, dai_link, np); + + if (ret) + return ret; + + /* Check if the cpu is the i2s encoder and parse i2s data */ + if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder")) + ret = gx_card_parse_i2s(card, np, index); + + /* Or apply codec to codec params if necessary */ + else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) + dai_link->params = &codec_params; + + return ret; +} + +static const struct meson_card_match_data gx_card_match_data = { + .add_link = gx_card_add_link, +}; + +static const struct of_device_id gx_card_of_match[] = { + { + .compatible = "amlogic,gx-sound-card", + .data = &gx_card_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, gx_card_of_match); + +static struct platform_driver gx_card_pdrv = { + .probe = meson_card_probe, + .remove = meson_card_remove, + .driver = { + .name = "gx-sound-card", + .of_match_table = gx_card_of_match, + }, +}; +module_platform_driver(gx_card_pdrv); + +MODULE_DESCRIPTION("Amlogic GX ALSA machine driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); From e9a0ef0b5ddcbc0d56c65aefc0f18d16e6f71207 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Feb 2020 15:49:28 +0100 Subject: [PATCH 093/415] ALSA: usb-audio: Don't create a mixer element with bogus volume range Some USB-audio descriptors provide a bogus volume range (e.g. volume min and max are identical), which confuses user-space. This patch makes the driver skipping such a control element. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206221 Link: https://lore.kernel.org/r/20200214144928.23628-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d659fdb475e2bc..96e20488804d1c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1666,6 +1666,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, /* get min/max values */ get_min_max_with_quirks(cval, 0, kctl); + /* skip a bogus volume range */ + if (cval->max <= cval->min) { + usb_audio_dbg(mixer->chip, + "[%d] FU [%s] skipped due to invalid volume\n", + cval->head.id, kctl->id.name); + snd_ctl_free_one(kctl); + return; + } + + if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); if (cval->dBmin < cval->dBmax || !cval->initialized) { From b2354e4009a773c00054b964d937e1b81cb92078 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 14 Feb 2020 14:47:04 +0100 Subject: [PATCH 094/415] ASoC: core: ensure component names are unique Make sure each ASoC component is registered with a unique name. The component is derived from the device name. If a device registers more than one component, the component names will be the same. This usually brings up a warning about the debugfs directory creation of the component since directory already exists. In such case, start numbering the component of the device so the names don't collide anymore. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200214134704.342501-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 03b87427faa774..6a58a8f6e3c4e9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2446,6 +2446,33 @@ static int snd_soc_register_dais(struct snd_soc_component *component, return ret; } +static char *snd_soc_component_unique_name(struct device *dev, + struct snd_soc_component *component) +{ + struct snd_soc_component *pos; + int count = 0; + char *name, *unique; + + name = fmt_single_name(dev, &component->id); + if (!name) + return name; + + /* Count the number of components registred by the device */ + for_each_component(pos) { + if (dev == pos->dev) + count++; + } + + /* Keep naming as it is for the 1st component */ + if (!count) + return name; + + unique = devm_kasprintf(dev, GFP_KERNEL, "%s-%d", name, count); + devm_kfree(dev, name); + + return unique; +} + static int snd_soc_component_initialize(struct snd_soc_component *component, const struct snd_soc_component_driver *driver, struct device *dev) { @@ -2454,7 +2481,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->card_list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = snd_soc_component_unique_name(dev, component); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; From 146f66975bafbcfab349901c9f9c9f521ac96cbb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Feb 2020 18:16:43 +0100 Subject: [PATCH 095/415] ALSA: pcm: oss: Unlock mutex temporarily for sleeping at read/write ALSA PCM OSS layer calls the generic __snd_pcm_lib_xfer() helper for the actual transfer of the audio data. The xfer helper may sleep long for waiting for the enough space becoming empty for read/write, and it does unlock/relock for the substream lock. This works fine, so far, but a slight problem specific to OSS layer is that OSS layer wraps yet more mutex (runtime->oss.params_lock) over __snd_pcm_lib_xfer() call; so this mutex is still locked during a possible long sleep, and it prevents the whole ioctl and other actions applied to the given stream. This patch adds the temporarily unlock and relock of the mutex around __snd_pcm_lib_xfer() call in the OSS layer to be more friendly to the concurrent accesses. The long mutex protection itself shouldn't be a real issue for the normal systems, and its influence appears only on strange things like fuzzers. Link: https://lore.kernel.org/r/20200214171643.26212-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 707eb2a9d50ccc..930def8201f411 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1217,8 +1217,10 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const if (ret < 0) break; } + mutex_unlock(&runtime->oss.params_lock); ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true, frames, in_kernel); + mutex_lock(&runtime->oss.params_lock); if (ret != -EPIPE && ret != -ESTRPIPE) break; /* test, if we can't store new data, because the stream */ @@ -1254,8 +1256,10 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p ret = snd_pcm_oss_capture_position_fixup(substream, &delay); if (ret < 0) break; + mutex_unlock(&runtime->oss.params_lock); ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true, frames, in_kernel); + mutex_lock(&runtime->oss.params_lock); if (ret == -EPIPE) { if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); From 51c366e38aaa6b298ba1e6ceef0f2c3de1180b29 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 14 Feb 2020 14:13:46 +0100 Subject: [PATCH 096/415] ASoC: meson: aiu: remove unused encoder structure Remove an unused structure definition which slipped through the initial driver submission. Fixes: 6ae9ca9ce986 ("ASoC: meson: aiu: add i2s and spdif support") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200214131350.337968-2-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-encoder-i2s.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index 13bf029086a9df..4900e38e7e494e 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -28,13 +28,6 @@ #define AIU_CLK_CTRL_MORE_I2S_DIV GENMASK(5, 0) #define AIU_CODEC_DAC_LRCLK_CTRL_DIV GENMASK(11, 0) -struct aiu_encoder_i2s { - struct clk *aoclk; - struct clk *mclk; - struct clk *mixer; - struct clk *pclk; -}; - static void aiu_encoder_i2s_divider_enable(struct snd_soc_component *component, bool enable) { From 269f00171273e47eebc915cc6ee8ceececa37a3a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 14 Feb 2020 14:13:47 +0100 Subject: [PATCH 097/415] ASoC: meson: aiu: fix clk bulk size allocation Fix the size of allocated memory for the clock bulk data Fixes: 6ae9ca9ce986 ("ASoC: meson: aiu: add i2s and spdif support") Reported-by: kbuild test robot Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200214131350.337968-3-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 5c4845a23a34f5..de678a9d5cab6d 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -203,7 +203,7 @@ static int aiu_clk_bulk_get(struct device *dev, struct clk_bulk_data *clks; int i, ret; - clks = devm_kcalloc(dev, num, sizeof(clks), GFP_KERNEL); + clks = devm_kcalloc(dev, num, sizeof(*clks), GFP_KERNEL); if (!clks) return -ENOMEM; From 6e700f0672199f773ad645c2b7e886c1d2e2046e Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 14 Feb 2020 14:13:48 +0100 Subject: [PATCH 098/415] ASoC: meson: aiu: fix irq registration The aiu stored the irq in an unsigned integer which may have discarded an error returned by platform_get_irq_byname(). This is incorrect and should have been a signed integer. Also drop the irq error traces from the probe function as this is already done by platform_get_irq_byname(). Fixes: 6ae9ca9ce986 ("ASoC: meson: aiu: add i2s and spdif support") Reported-by: kbuild test robot Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200214131350.337968-4-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu.c | 8 ++------ sound/soc/meson/aiu.h | 2 +- 2 files changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index de678a9d5cab6d..34b40b8b82997f 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -314,16 +314,12 @@ static int aiu_probe(struct platform_device *pdev) } aiu->i2s.irq = platform_get_irq_byname(pdev, "i2s"); - if (aiu->i2s.irq < 0) { - dev_err(dev, "Can't get i2s irq\n"); + if (aiu->i2s.irq < 0) return aiu->i2s.irq; - } aiu->spdif.irq = platform_get_irq_byname(pdev, "spdif"); - if (aiu->spdif.irq < 0) { - dev_err(dev, "Can't get spdif irq\n"); + if (aiu->spdif.irq < 0) return aiu->spdif.irq; - } ret = aiu_clk_get(dev); if (ret) diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index a65a576e34005b..097c26de7b7c60 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -26,7 +26,7 @@ enum aiu_clk_ids { struct aiu_interface { struct clk_bulk_data *clks; unsigned int clk_num; - unsigned int irq; + int irq; }; struct aiu { From 74a56f2a4a9ec72ef1daceeb2dda8b41370c1419 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 14 Feb 2020 14:13:49 +0100 Subject: [PATCH 099/415] ASoC: meson: aiu: fix acodec dai input name init Remove the double initialization of the dai input name as reported by sparse. Fixes: 65816025d461 ("ASoC: meson: aiu: add internal dac codec control support") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200214131350.337968-5-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-acodec-ctrl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index 12d8a4d351a1f7..b8e88b1a4fc87e 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -128,7 +128,6 @@ static const struct snd_soc_dai_ops aiu_acodec_ctrl_output_ops = { #define AIU_ACODEC_INPUT(xname) { \ .name = "ACODEC CTRL " xname, \ - .name = xname, \ .playback = AIU_ACODEC_STREAM(xname, "Playback", 8), \ .ops = &aiu_acodec_ctrl_input_ops, \ .probe = meson_codec_glue_input_dai_probe, \ From 3cd23f021e2e5f3350125abcb39f12430df87d06 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 14 Feb 2020 14:13:50 +0100 Subject: [PATCH 100/415] ASoC: meson: codec-glue: fix pcm format cast warning Clarify the cast of snd_pcm_format_t and fix the sparse warning: restricted snd_pcm_format_t degrades to integer Fixes: 9c29fd9bdf92 ("ASoC: meson: g12a: extract codec-to-codec utils") Reported-by: kbuild test robot Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200214131350.337968-6-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/meson-codec-glue.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/meson-codec-glue.c b/sound/soc/meson/meson-codec-glue.c index 97bbc967e1769b..524a3347233738 100644 --- a/sound/soc/meson/meson-codec-glue.c +++ b/sound/soc/meson/meson-codec-glue.c @@ -74,7 +74,7 @@ int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream, data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params)); data->params.rate_min = params_rate(params); data->params.rate_max = params_rate(params); - data->params.formats = 1 << params_format(params); + data->params.formats = 1ULL << (__force int) params_format(params); data->params.channels_min = params_channels(params); data->params.channels_max = params_channels(params); data->params.sig_bits = dai->driver->playback.sig_bits; From 8dc5efe3d17cd572328ac4f1ebde629c83317f54 Mon Sep 17 00:00:00 2001 From: Nick Kossifidis Date: Sat, 15 Feb 2020 03:23:35 +0200 Subject: [PATCH 101/415] ALSA: usb-audio: Add support for Presonus Studio 1810c This patch adds support for Presonus Studio 1810c, a usb interface that's UAC2 compliant with a few quirks and a few extra hw-specific controls. I've tested all 3 altsettings and the added switch controls and they work as expected. More infos on the card: https://www.presonus.com/products/Studio-1810c Note that this work is based on packet inspection with usbmon. I just wanted to get this card to work for using it on our open-source radio station: https://github.com/UoC-Radio v2 address issues reported by Takashi: * Properly get/set enum type controls * Prevent race condition on switch_get/set * Various control naming changes * Various coding style fixes v3 improve readability of sample rate filtering and some other minor changes. Signed-off-by: Nick Kossifidis Link: https://lore.kernel.org/r/5e47481a.1c69fb81.befb3.8dac@mx.google.com Signed-off-by: Takashi Iwai --- sound/usb/Makefile | 1 + sound/usb/format.c | 37 +++ sound/usb/mixer_quirks.c | 5 + sound/usb/mixer_s1810c.c | 595 +++++++++++++++++++++++++++++++++++++++ sound/usb/mixer_s1810c.h | 7 + sound/usb/quirks.c | 36 +++ 6 files changed, 681 insertions(+) create mode 100644 sound/usb/mixer_s1810c.c create mode 100644 sound/usb/mixer_s1810c.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 78edd7d2f41861..56031026b1137d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \ mixer_scarlett.o \ mixer_scarlett_gen2.o \ mixer_us16x08.o \ + mixer_s1810c.o \ pcm.o \ power.o \ proc.o \ diff --git a/sound/usb/format.c b/sound/usb/format.c index 9260136e4c9bb7..51e9fca1f3dc89 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -226,6 +226,36 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } + +/* + * Presonus Studio 1810c supports a limited set of sampling + * rates per altsetting but reports the full set each time. + * If we don't filter out the unsupported rates and attempt + * to configure the card, it will hang refusing to do any + * further audio I/O until a hard reset is performed. + * + * The list of supported rates per altsetting (set of available + * I/O channels) is described in the owner's manual, section 2.2. + */ +static bool s1810c_valid_sample_rate(struct audioformat *fp, + unsigned int rate) +{ + switch (fp->altsetting) { + case 1: + /* All ADAT ports available */ + return rate <= 48000; + case 2: + /* Half of ADAT ports available */ + return (rate == 88200 || rate == 96000); + case 3: + /* Analog I/O only (no S/PDIF nor ADAT) */ + return rate >= 176400; + default: + return false; + } + return false; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -262,6 +292,12 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, } for (rate = min; rate <= max; rate += res) { + + /* Filter out invalid rates on Presonus Studio 1810c */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + !s1810c_valid_sample_rate(fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) @@ -276,6 +312,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, break; } +skip_rate: /* avoid endless loop */ if (res == 0) break; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index c237e24f08d9bb..02b036b2aefbd6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -34,6 +34,7 @@ #include "mixer_scarlett.h" #include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" +#include "mixer_s1810c.h" #include "helper.h" struct std_mono_table { @@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fd4): /* RME */ err = snd_rme_controls_create(mixer); break; + + case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + err = snd_sc1810_init_mixer(mixer); + break; } return err; diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c new file mode 100644 index 00000000000000..816879a07f82f0 --- /dev/null +++ b/sound/usb/mixer_s1810c.c @@ -0,0 +1,595 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis + * + * Based on reverse engineering of the communication protocol + * between the windows driver / Univeral Control (UC) program + * and the device, through usbmon. + * + * For now this bypasses the mixer, with all channels split, + * so that the software can mix with greater flexibility. + * It also adds controls for the 4 buttons on the front of + * the device. + */ + +#include +#include +#include +#include +#include + +#include "usbaudio.h" +#include "mixer.h" +#include "mixer_quirks.h" +#include "helper.h" +#include "mixer_s1810c.h" + +#define SC1810C_CMD_REQ 160 +#define SC1810C_CMD_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT) +#define SC1810C_CMD_F1 0x50617269 +#define SC1810C_CMD_F2 0x14 + +/* + * DISCLAIMER: These are just guesses based on the + * dumps I got. + * + * It seems like a selects between + * device (0), mixer (0x64) and output (0x65) + * + * For mixer (0x64): + * * b selects an input channel (see below). + * * c selects an output channel pair (see below). + * * d selects left (0) or right (1) of that pair. + * * e 0-> disconnect, 0x01000000-> connect, + * 0x0109-> used for stereo-linking channels, + * e is also used for setting volume levels + * in which case b is also set so I guess + * this way it is possible to set the volume + * level from the specified input to the + * specified output. + * + * IN Channels: + * 0 - 7 Mic/Inst/Line (Analog inputs) + * 8 - 9 S/PDIF + * 10 - 17 ADAT + * 18 - 35 DAW (Inputs from the host) + * + * OUT Channels (pairs): + * 0 -> Main out + * 1 -> Line1/2 + * 2 -> Line3/4 + * 3 -> S/PDIF + * 4 -> ADAT? + * + * For device (0): + * * b and c are not used, at least not on the + * dumps I got. + * * d sets the control id to be modified + * (see below). + * * e sets the setting for that control. + * (so for the switches I was interested + * in it's 0/1) + * + * For output (0x65): + * * b is the output channel (see above). + * * c is zero. + * * e I guess the same as with mixer except 0x0109 + * which I didn't see in my dumps. + * + * The two fixed fields have the same values for + * mixer and output but a different set for device. + */ +struct s1810c_ctl_packet { + u32 a; + u32 b; + u32 fixed1; + u32 fixed2; + u32 c; + u32 d; + u32 e; +}; + +#define SC1810C_CTL_LINE_SW 0 +#define SC1810C_CTL_MUTE_SW 1 +#define SC1810C_CTL_AB_SW 3 +#define SC1810C_CTL_48V_SW 4 + +#define SC1810C_SET_STATE_REQ 161 +#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE +#define SC1810C_SET_STATE_F1 0x64656D73 +#define SC1810C_SET_STATE_F2 0xF4 + +#define SC1810C_GET_STATE_REQ 162 +#define SC1810C_GET_STATE_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN) +#define SC1810C_GET_STATE_F1 SC1810C_SET_STATE_F1 +#define SC1810C_GET_STATE_F2 SC1810C_SET_STATE_F2 + +#define SC1810C_STATE_F1_IDX 2 +#define SC1810C_STATE_F2_IDX 3 + +/* + * This packet includes mixer volumes and + * various other fields, it's an extended + * version of ctl_packet, with a and b + * being zero and different f1/f2. + */ +struct s1810c_state_packet { + u32 fields[63]; +}; + +#define SC1810C_STATE_48V_SW 58 +#define SC1810C_STATE_LINE_SW 59 +#define SC1810C_STATE_MUTE_SW 60 +#define SC1810C_STATE_AB_SW 62 + +struct s1810_mixer_state { + uint16_t seqnum; + struct mutex usb_mutex; + struct mutex data_mutex; +}; + +static int +snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a, + u32 b, u32 c, u32 d, u32 e) +{ + struct s1810c_ctl_packet pkt = { 0 }; + int ret = 0; + + pkt.fixed1 = SC1810C_CMD_F1; + pkt.fixed2 = SC1810C_CMD_F2; + + pkt.a = a; + pkt.b = b; + pkt.c = c; + pkt.d = d; + /* + * Value for settings 0/1 for this + * output channel is always 0 (probably because + * there is no ADAT output on 1810c) + */ + pkt.e = (c == 4) ? 0 : e; + + ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + SC1810C_CMD_REQ, + SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send ctl packet\n"); + return ret; + } + return 0; +} + +/* + * When opening Universal Control the program periodicaly + * sends and receives state packets for syncinc state between + * the device and the host. + * + * Note that if we send only the request to get data back we'll + * get an error, we need to first send an empty state packet and + * then ask to receive a filled. Their seqnumbers must also match. + */ +static int +snd_sc1810c_get_status_field(struct usb_device *dev, + u32 *field, int field_idx, uint16_t *seqnum) +{ + struct s1810c_state_packet pkt_out = { 0 }; + struct s1810c_state_packet pkt_in = { 0 }; + int ret = 0; + + pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; + pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2; + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_SET_STATE_REQ, + SC1810C_SET_STATE_REQTYPE, + (*seqnum), 0, &pkt_out, sizeof(pkt_out)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send state packet (%d)\n", ret); + return ret; + } + + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_GET_STATE_REQ, + SC1810C_GET_STATE_REQTYPE, + (*seqnum), 0, &pkt_in, sizeof(pkt_in)); + if (ret < 0) { + dev_warn(&dev->dev, "could not get state field %u (%d)\n", + field_idx, ret); + return ret; + } + + (*field) = pkt_in.fields[field_idx]; + (*seqnum)++; + return 0; +} + +/* + * This is what I got when bypassing the mixer with + * all channels split. I'm not 100% sure of what's going + * on, I could probably clean this up based on my observations + * but I prefer to keep the same behavior as the windows driver. + */ +static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip) +{ + u32 a, b, c, e, n, off; + struct usb_device *dev = chip->dev; + + /* Set initial volume levels ? */ + a = 0x64; + e = 0xbc; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off, c = 0; b < 18 + off; b++) { + /* This channel to all outputs ? */ + for (c = 0; c <= 8; c++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + /* This channel to main output (again) */ + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + /* + * I noticed on UC that DAW channels have different + * initial volumes, so this makes sense. + */ + e = 0xb53bf0; + } + + /* Connect analog outputs ? */ + a = 0x65; + e = 0x01000000; + for (b = 1; b < 3; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e); + + /* Set initial volume levels for S/PDIF mappings ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF output ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Connect all outputs (again) ? */ + a = 0x65; + e = 0x01000000; + for (b = 0; b < 4; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + + /* Basic routing to get sound out of the device */ + a = 0x64; + e = 0x01000000; + for (c = 0; c < 4; c++) { + for (b = 0; b < 36; b++) { + if ((c == 0 && b == 18) || /* DAW1/2 -> Main */ + (c == 1 && b == 20) || /* DAW3/4 -> Line3/4 */ + (c == 2 && b == 22) || /* DAW4/5 -> Line5/6 */ + (c == 3 && b == 24)) { /* DAW5/6 -> S/PDIF */ + /* Left */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + b++; + /* Right */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } else { + /* Leave the rest disconnected */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + } + } + } + + /* Set initial volume levels for S/PDIF (again) ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF outputs (again) ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Again ? */ + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + return 0; +} + +/* + * Sync state with the device and retrieve the requested field, + * whose index is specified in (kctl->private_value & 0xFF), + * from the received fields array. + */ +static int +snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl, u32 *state) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 field = 0; + u32 ctl_idx = (u32) (kctl->private_value & 0xFF); + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_sc1810c_get_status_field(chip->dev, &field, + ctl_idx, &private->seqnum); + if (ret < 0) + goto unlock; + + *state = field; + unlock: + mutex_unlock(&private->usb_mutex); + return ret ? ret : 0; +} + +/* + * Send a control packet to the device for the control id + * specified in (kctl->private_value >> 8) with value + * specified in (kctl->private_value >> 16). + */ +static int +snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_id = (pval >> 8) & 0xFF; + u32 ctl_val = (pval >> 16) & 0x1; + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val); + mutex_unlock(&private->usb_mutex); + return ret; +} + +/* Generic get/set/init functions for switch controls */ + +static int +snd_s1810c_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 state = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &state); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + ctl_elem->value.enumerated.item[0] = (int)state; + break; + default: + ctl_elem->value.integer.value[0] = (long)state; + } + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? ret : 0; +} + +static int +snd_s1810c_switch_set(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 curval = 0; + u32 newval = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &curval); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + newval = (u32) ctl_elem->value.enumerated.item[0]; + break; + default: + newval = (u32) ctl_elem->value.integer.value[0]; + } + + if (curval == newval) + goto unlock; + + kctl->private_value &= ~(0x1 << 16); + kctl->private_value |= (unsigned int)(newval & 0x1) << 16; + ret = snd_s1810c_set_switch_state(mixer, kctl); + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? 0 : 1; +} + +static int +snd_s1810c_switch_init(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *new_kctl) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + + elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = 0; + elem->head.id = 0; + elem->channels = 1; + + kctl = snd_ctl_new1(new_kctl, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + return snd_usb_mixer_add_control(&elem->head, kctl); +} + +static int +snd_s1810c_line_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Preamp On (Mic/Inst)", + "Preamp Off (Line in)" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_line_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line 1/2 Source Type", + .info = snd_s1810c_line_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_mute_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mute Main Out Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_48v_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "48V Phantom Power On Mic Inputs Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8) +}; + +static int +snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "1/2", + "3/4" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_ab_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone 1 Source Route", + .info = snd_s1810c_ab_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8) +}; + +static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = mixer->private_data; + kfree(private); + mixer->private_data = NULL; +} + +/* Entry point, called from mixer_quirks.c */ +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = NULL; + struct snd_usb_audio *chip = mixer->chip; + struct usb_device *dev = chip->dev; + int ret = 0; + + /* Run this only once */ + if (!list_empty(&chip->mixer_list)) + return 0; + + dev_info(&dev->dev, + "Presonus Studio 1810c, device_setup: %u\n", chip->setup); + if (chip->setup == 1) + dev_info(&dev->dev, "(8out/18in @ 48KHz)\n"); + else if (chip->setup == 2) + dev_info(&dev->dev, "(6out/8in @ 192KHz)\n"); + else + dev_info(&dev->dev, "(8out/14in @ 96KHz)\n"); + + ret = snd_s1810c_init_mixer_maps(chip); + if (ret < 0) + return ret; + + private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL); + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + + mixer->private_data = private; + mixer->private_free = snd_sc1810_mixer_state_free; + + private->seqnum = 1; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); + if (ret < 0) + return ret; + return ret; +} diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h new file mode 100644 index 00000000000000..a79a3743cff330 --- /dev/null +++ b/sound/usb/mixer_s1810c.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis + */ + +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3a5242e383b24a..58761d40875915 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1252,6 +1252,38 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* + * Altno settings: + * + * Playback (Interface 1): + * 1: 6 Analog + 2 S/PDIF + * 2: 6 Analog + 2 S/PDIF + * 3: 6 Analog + * + * Capture (Interface 2): + * 1: 8 Analog + 2 S/PDIF + 8 ADAT + * 2: 8 Analog + 2 S/PDIF + 4 ADAT + * 3: 8 Analog + */ + + /* + * I'll leave 2 as the default one and + * use device_setup to switch to the + * other two. + */ + if ((chip->setup == 0 || chip->setup > 2) && altno != 2) + return 1; + else if (chip->setup == 1 && altno != 1) + return 1; + else if (chip->setup == 2 && altno != 3) + return 1; + + return 0; +} + int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, int iface, int altno) @@ -1265,6 +1297,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, /* fasttrackpro usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); + /* presonus studio 1810c: skip altsets incompatible with device_setup */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + return s1810c_skip_setting_quirk(chip, iface, altno); + return 0; } From 10fa9512769fa3b15ea29f4f331f4604c17b4b2c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Feb 2020 12:20:58 +0100 Subject: [PATCH 102/415] usb: audio-v2: Add uac2_effect_unit_descriptor definition The UAC2 Effect Unit Descriptor has a slightly different definition from other similar ones like Processing Unit or Extension Unit. Define it here so that it can be used in USB-audio driver in a later patch. Acked-by: Greg Kroah-Hartman Link: https://lore.kernel.org/r/20200213112059.18745-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/linux/usb/audio-v2.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index ba4b3e3327ff30..cb9900b34b6798 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -156,6 +156,18 @@ struct uac2_feature_unit_descriptor { __u8 bmaControls[0]; /* variable length */ } __attribute__((packed)); +/* 4.7.2.10 Effect Unit Descriptor */ + +struct uac2_effect_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __le16 wEffectType; + __u8 bSourceID; + __u8 bmaControls[]; /* variable length */ +} __attribute__((packed)); + /* 4.9.2 Class-Specific AS Interface Descriptor */ struct uac2_as_header_descriptor { From 60081b35c68ba6a466dee08de581be06999c930a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Feb 2020 12:20:59 +0100 Subject: [PATCH 103/415] ALSA: usb-audio: Parse source ID of UAC2 effect unit During parsing the input source, we currently cut off at the Effect Unit node without parsing further its source id. It's no big problem, so far, but it should be more consistent to parse it properly. This patch adds the recursive parsing in parse_term_effect_unit(). It doesn't add anything in the audio unit parser itself, and the effect unit itself is still skipped, though. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206147 Link: https://lore.kernel.org/r/20200213112059.18745-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81b2db0edd5f43..56d0878e49995a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -901,6 +901,12 @@ static int parse_term_effect_unit(struct mixer_build *state, struct usb_audio_term *term, void *p1, int id) { + struct uac2_effect_unit_descriptor *d = p1; + int err; + + err = __check_input_term(state, d->bSourceID, term); + if (err < 0) + return err; term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ term->id = id; return 0; From 2f0b42034bd75a938cdf144149d6db4fa4d51208 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 17 Feb 2020 10:03:11 +0800 Subject: [PATCH 104/415] ASoC: rt1015: fix typo for bypass boost control Fix typo for "Bypass Boost" control. Signed-off-by: Jack Yu Link: https://lore.kernel.org/r/20200217020311.12793-1-jack.yu@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1015.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 6d490e2dbc259b..d300b417dd508c 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -444,7 +444,7 @@ static int rt1015_boost_mode_put(struct snd_kcontrol *kcontrol, return 0; } -static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol, +static int rt1015_bypass_boost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -457,7 +457,7 @@ static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol, return 0; } -static int rt5518_bypass_boost_put(struct snd_kcontrol *kcontrol, +static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -497,7 +497,7 @@ static const struct snd_kcontrol_new rt1015_snd_controls[] = { rt1015_boost_mode_get, rt1015_boost_mode_put), SOC_ENUM("Mono LR Select", rt1015_mono_lr_sel), SOC_SINGLE_EXT("Bypass Boost", SND_SOC_NOPM, 0, 1, 0, - rt5518_bypass_boost_get, rt5518_bypass_boost_put), + rt1015_bypass_boost_get, rt1015_bypass_boost_put), }; static int rt1015_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, From f07980d4ed60fbb35857b655c94b111f4ddf2abf Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Mon, 17 Feb 2020 11:16:53 +0800 Subject: [PATCH 105/415] drm/mediatek: fix race condition for HDMI jack status reporting hdmi_conn_detect and mtk_hdmi_audio_hook_plugged_cb would be called by different threads. Imaging the following calling sequence: Thread A Thread B -------------------------------------------------------------------- mtk_hdmi_audio_hook_plugged_cb() mtk_cec_hpd_high() -> disconnected hdmi_conn_detect() mtk_cec_hpd_high() -> connected plugged_cb(connected) plugged_cb(disconnected) The latest disconnected is false reported. Makes mtk_cec_hpd_high and plugged_cb atomic to fix. Also uses the same lock to protect read/write of plugged_cb and codec_dev. Fixes: 5d3c64477392 ("drm/mediatek: support HDMI jack status reporting") Signed-off-by: Tzung-Bi Shih Acked-by: CK Hu Link: https://lore.kernel.org/r/20200217105513.2.I477092c2f104fd589133436c3ae4590e6fc6323b@changeid Signed-off-by: Mark Brown --- drivers/gpu/drm/mediatek/mtk_hdmi.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index 03aeb73005ef88..d80017e3d84a35 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include #include @@ -171,6 +172,7 @@ struct mtk_hdmi { bool enabled; hdmi_codec_plugged_cb plugged_cb; struct device *codec_dev; + struct mutex update_plugged_status_lock; }; static inline struct mtk_hdmi *hdmi_ctx_from_bridge(struct drm_bridge *b) @@ -1199,10 +1201,13 @@ static void mtk_hdmi_clk_disable_audio(struct mtk_hdmi *hdmi) static enum drm_connector_status mtk_hdmi_update_plugged_status(struct mtk_hdmi *hdmi) { - bool connected = mtk_cec_hpd_high(hdmi->cec_dev); + bool connected; + mutex_lock(&hdmi->update_plugged_status_lock); + connected = mtk_cec_hpd_high(hdmi->cec_dev); if (hdmi->plugged_cb && hdmi->codec_dev) hdmi->plugged_cb(hdmi->codec_dev, connected); + mutex_unlock(&hdmi->update_plugged_status_lock); return connected ? connector_status_connected : connector_status_disconnected; @@ -1669,8 +1674,11 @@ static int mtk_hdmi_audio_hook_plugged_cb(struct device *dev, void *data, { struct mtk_hdmi *hdmi = data; + mutex_lock(&hdmi->update_plugged_status_lock); hdmi->plugged_cb = fn; hdmi->codec_dev = codec_dev; + mutex_unlock(&hdmi->update_plugged_status_lock); + mtk_hdmi_update_plugged_status(hdmi); return 0; @@ -1729,6 +1737,7 @@ static int mtk_drm_hdmi_probe(struct platform_device *pdev) return ret; } + mutex_init(&hdmi->update_plugged_status_lock); platform_set_drvdata(pdev, hdmi); ret = mtk_hdmi_output_init(hdmi); From 0247142233239dc235f8239aab5c7991250d4e66 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 17 Feb 2020 10:20:19 +0100 Subject: [PATCH 106/415] ASoC: meson: aiu: simplify component addition Now that the component name is unique within ASoC, there is no need to hack the debugfs prefix to add more than one ASoC component to a linux device. Remove the unnecessary function and use snd_soc_register_component() directly. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200217092019.433402-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-acodec-ctrl.c | 7 +++---- sound/soc/meson/aiu-codec-ctrl.c | 7 +++---- sound/soc/meson/aiu.c | 20 -------------------- sound/soc/meson/aiu.h | 8 -------- 4 files changed, 6 insertions(+), 36 deletions(-) diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index b8e88b1a4fc87e..7078197e0cc502 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -197,8 +197,7 @@ static const struct snd_soc_component_driver aiu_acodec_ctrl_component = { int aiu_acodec_ctrl_register_component(struct device *dev) { - return aiu_add_component(dev, &aiu_acodec_ctrl_component, - aiu_acodec_ctrl_dai_drv, - ARRAY_SIZE(aiu_acodec_ctrl_dai_drv), - "acodec"); + return snd_soc_register_component(dev, &aiu_acodec_ctrl_component, + aiu_acodec_ctrl_dai_drv, + ARRAY_SIZE(aiu_acodec_ctrl_dai_drv)); } diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c index 8646a953e3b33f..4b773d3e8b0735 100644 --- a/sound/soc/meson/aiu-codec-ctrl.c +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -144,9 +144,8 @@ static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = { int aiu_hdmi_ctrl_register_component(struct device *dev) { - return aiu_add_component(dev, &aiu_hdmi_ctrl_component, - aiu_hdmi_ctrl_dai_drv, - ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv), - "hdmi"); + return snd_soc_register_component(dev, &aiu_hdmi_ctrl_component, + aiu_hdmi_ctrl_dai_drv, + ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv)); } diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 34b40b8b82997f..d3e2d40e956258 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -71,26 +71,6 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, return 0; } -int aiu_add_component(struct device *dev, - const struct snd_soc_component_driver *component_driver, - struct snd_soc_dai_driver *dai_drv, - int num_dai, - const char *debugfs_prefix) -{ - struct snd_soc_component *component; - - component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL); - if (!component) - return -ENOMEM; - -#ifdef CONFIG_DEBUG_FS - component->debugfs_prefix = debugfs_prefix; -#endif - - return snd_soc_add_component(dev, component, component_driver, - dai_drv, num_dai); -} - static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name) diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 097c26de7b7c60..06a968c557287b 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -11,9 +11,7 @@ struct clk; struct clk_bulk_data; struct device; struct of_phandle_args; -struct snd_soc_component_driver; struct snd_soc_dai; -struct snd_soc_dai_driver; struct snd_soc_dai_ops; enum aiu_clk_ids { @@ -45,12 +43,6 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, const char **dai_name, unsigned int component_id); -int aiu_add_component(struct device *dev, - const struct snd_soc_component_driver *component_driver, - struct snd_soc_dai_driver *dai_drv, - int num_dai, - const char *debugfs_prefix); - int aiu_hdmi_ctrl_register_component(struct device *dev); int aiu_acodec_ctrl_register_component(struct device *dev); From a4877a6fb2bd2e356a5eaacd86d6b6d69ff84e69 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Tue, 18 Feb 2020 11:38:24 +0100 Subject: [PATCH 107/415] ASoC: soc-pcm: fix regression in soc_new_pcm() Commit af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") swapped the SNDRV_PCM_STREAM_* parameter in the snd_soc_dai_stream_valid(cpu_dai, ...) checks. But that works only for codec2codec links. For normal links it breaks registration of playback/capture-only PCM devices. E.g. on qcom/apq8016_sbc there is usually one playback-only and one capture-only PCM device, but they disappeared after the commit. The codec2codec case was added in commit a342031cdd08 ("ASoC: create pcm for codec2codec links as well") as an extra check (e.g. `playback = playback && cpu_playback->channels_min`). We should be able to simplify the code by checking directly for the correct stream type in the loop. This also fixes the regression because we check for PLAYBACK for both codec and cpu dai again when codec2codec is not used. Fixes: af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") Signed-off-by: Stephan Gerhold Tested-by: Jerome Brunet Reviewed-by: Jerome Brunet Cc: Jerome Brunet Cc: Sameer Pujar Link: https://lore.kernel.org/r/20200218103824.26708-1-stephan@gerhold.net Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6630fadd6e098f..65a3856be2505c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2858,22 +2858,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { /* Adapt stream for codec2codec links */ - struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ? - &cpu_dai->driver->playback : &cpu_dai->driver->capture; - struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ? - &cpu_dai->driver->capture : &cpu_dai->driver->playback; + int cpu_capture = rtd->dai_link->params ? + SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int cpu_playback = rtd->dai_link->params ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; for_each_rtd_codec_dai(rtd, i, codec_dai) { if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + snd_soc_dai_stream_valid(cpu_dai, cpu_playback)) playback = 1; if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + snd_soc_dai_stream_valid(cpu_dai, cpu_capture)) capture = 1; } - - capture = capture && cpu_capture->channels_min; - playback = playback && cpu_playback->channels_min; } if (rtd->dai_link->playback_only) { From 386dd54b3a2eedb91aa6e465e7c3a57db04f3960 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:16 +0100 Subject: [PATCH 108/415] ALSA: core: Expand DMA buffer information Update DMA buffer definition for snd_compr_runtime so it is represented similarly as in snd_pcm_runtime. While at it, modify snd_compr_set_runtime_buffer to account for newly added members. Signed-off-by: Cezary Rojewski Reviewed-by: Takashi Iwai Acked-by: Vinod Koul Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- include/sound/compress_driver.h | 35 ++++++++++++++++++++++++--------- 1 file changed, 26 insertions(+), 9 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index bc88d6f964da9e..00f633c0c3ba5f 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -23,7 +23,6 @@ struct snd_compr_ops; * struct snd_compr_runtime: runtime stream description * @state: stream state * @ops: pointer to DSP callbacks - * @dma_buffer_p: runtime dma buffer pointer * @buffer: pointer to kernel buffer, valid only when not in mmap mode or * DSP doesn't implement copy * @buffer_size: size of the above buffer @@ -34,11 +33,14 @@ struct snd_compr_ops; * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep * @private_data: driver private data pointer + * @dma_area: virtual buffer address + * @dma_addr: physical buffer address (not accessible from main CPU) + * @dma_bytes: size of DMA area + * @dma_buffer_p: runtime dma buffer pointer */ struct snd_compr_runtime { snd_pcm_state_t state; struct snd_compr_ops *ops; - struct snd_dma_buffer *dma_buffer_p; void *buffer; u64 buffer_size; u32 fragment_size; @@ -47,6 +49,11 @@ struct snd_compr_runtime { u64 total_bytes_transferred; wait_queue_head_t sleep; void *private_data; + + unsigned char *dma_area; + dma_addr_t dma_addr; + size_t dma_bytes; + struct snd_dma_buffer *dma_buffer_p; }; /** @@ -180,19 +187,29 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) /** * snd_compr_set_runtime_buffer - Set the Compress runtime buffer - * @substream: compress substream to set + * @stream: compress stream to set * @bufp: the buffer information, NULL to clear * * Copy the buffer information to runtime buffer when @bufp is non-NULL. * Otherwise it clears the current buffer information. */ -static inline void snd_compr_set_runtime_buffer( - struct snd_compr_stream *substream, - struct snd_dma_buffer *bufp) +static inline void +snd_compr_set_runtime_buffer(struct snd_compr_stream *stream, + struct snd_dma_buffer *bufp) { - struct snd_compr_runtime *runtime = substream->runtime; - - runtime->dma_buffer_p = bufp; + struct snd_compr_runtime *runtime = stream->runtime; + + if (bufp) { + runtime->dma_buffer_p = bufp; + runtime->dma_area = bufp->area; + runtime->dma_addr = bufp->addr; + runtime->dma_bytes = bufp->bytes; + } else { + runtime->dma_buffer_p = NULL; + runtime->dma_area = NULL; + runtime->dma_addr = 0; + runtime->dma_bytes = 0; + } } int snd_compr_stop_error(struct snd_compr_stream *stream, From b9759ef2fd1acb0d3f3dce7991c44a4c5e9e68a3 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:17 +0100 Subject: [PATCH 109/415] ALSA: core: Implement compress page allocation and free routines Add simple malloc and free methods for memory management for compress streams. Based on snd_pcm_lib_malloc_pages and snd_pcm_lib_free_pages implementation. Signed-off-by: Divya Prakash Signed-off-by: Cezary Rojewski Reviewed-by: Takashi Iwai Acked-by: Vinod Koul Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- include/sound/compress_driver.h | 5 ++++ sound/core/compress_offload.c | 42 +++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 00f633c0c3ba5f..6ce8effa0b128f 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data + * @dma_buffer: allocated buffer if any */ struct snd_compr_stream { const char *name; @@ -78,6 +79,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; void *private_data; + struct snd_dma_buffer dma_buffer; }; /** @@ -212,6 +214,9 @@ snd_compr_set_runtime_buffer(struct snd_compr_stream *stream, } } +int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size); +int snd_compr_free_pages(struct snd_compr_stream *stream); + int snd_compr_stop_error(struct snd_compr_stream *stream, snd_pcm_state_t state); diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 9de1c9a0173e04..509290f2efa8ec 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -488,6 +488,48 @@ snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) } #endif /* !COMPR_CODEC_CAPS_OVERFLOW */ +int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size) +{ + struct snd_dma_buffer *dmab; + int ret; + + if (snd_BUG_ON(!(stream) || !(stream)->runtime)) + return -EINVAL; + dmab = kzalloc(sizeof(*dmab), GFP_KERNEL); + if (!dmab) + return -ENOMEM; + dmab->dev = stream->dma_buffer.dev; + ret = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev, size, dmab); + if (ret < 0) { + kfree(dmab); + return ret; + } + + snd_compr_set_runtime_buffer(stream, dmab); + stream->runtime->dma_bytes = size; + return 1; +} +EXPORT_SYMBOL(snd_compr_malloc_pages); + +int snd_compr_free_pages(struct snd_compr_stream *stream) +{ + struct snd_compr_runtime *runtime = stream->runtime; + + if (snd_BUG_ON(!(stream) || !(stream)->runtime)) + return -EINVAL; + if (runtime->dma_area == NULL) + return 0; + if (runtime->dma_buffer_p != &stream->dma_buffer) { + /* It's a newly allocated buffer. Release it now. */ + snd_dma_free_pages(runtime->dma_buffer_p); + kfree(runtime->dma_buffer_p); + } + + snd_compr_set_runtime_buffer(stream, NULL); + return 0; +} +EXPORT_SYMBOL(snd_compr_free_pages); + /* revisit this with snd_pcm_preallocate_xxx */ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, struct snd_compr_params *params) From 4a9ce6e4d9fb9c4acc44f647a68e59ea50ff1caf Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:18 +0100 Subject: [PATCH 110/415] ASoC: SOF: Intel: Account for compress streams when servicing IRQs Update stream irq handler definition to correctly set hdac_stream current position when servicing stream interrupts for compress streams. Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- include/sound/hdaudio.h | 2 ++ sound/soc/sof/intel/hda-stream.c | 25 +++++++++++++++++++++++-- 2 files changed, 25 insertions(+), 2 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index d4299e146d95e5..affedc2801c4be 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -513,6 +513,7 @@ struct hdac_stream { struct snd_pcm_substream *substream; /* assigned substream, * set in PCM open */ + struct snd_compr_stream *cstream; unsigned int format_val; /* format value to be set in the * controller and the codec */ @@ -527,6 +528,7 @@ struct hdac_stream { bool locked:1; bool stripe:1; /* apply stripe control */ + u64 curr_pos; /* timestamp */ unsigned long start_wallclk; /* start + minimum wallclk */ unsigned long period_wallclk; /* wallclk for period */ diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index c0ab9bb2a7978c..7daa913dbde08a 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -571,6 +571,22 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) return ret; } +static void +hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size) +{ + u64 prev_pos, pos, num_bytes; + + div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos); + pos = snd_hdac_stream_get_pos_posbuf(hstream); + + if (pos < prev_pos) + num_bytes = (buffer_size - prev_pos) + pos; + else + num_bytes = pos - prev_pos; + + hstream->curr_pos += num_bytes; +} + static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) { struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus); @@ -588,14 +604,19 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) snd_hdac_stream_writeb(s, SD_STS, sd_status); active = true; - if (!s->substream || + if ((!s->substream && !s->cstream) || !s->running || (sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0) continue; /* Inform ALSA only in case not do that with IPC */ - if (sof_hda->no_ipc_position) + if (s->substream && sof_hda->no_ipc_position) { snd_sof_pcm_period_elapsed(s->substream); + } else if (s->cstream) { + hda_dsp_set_bytes_transferred(s, + s->cstream->runtime->buffer_size); + snd_compr_fragment_elapsed(s->cstream); + } } } From f3b433e4699fa358ce5b7bd7688bebe36068c199 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:19 +0100 Subject: [PATCH 111/415] ASoC: SOF: Implement Probe IPC API Add all required types and methods to support each and every request that driver could sent to firmware. Probe is one of SOF firmware features which allows for data extraction and injection directly from or to DMA stream. Exposes eight IPCs: - addition and removal of injection DMAs - addition and removal of probe points - info retrieval of injection DMAs and probe points - probe initialization and cleanup Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- include/sound/sof/header.h | 11 ++ sound/soc/sof/Kconfig | 8 + sound/soc/sof/Makefile | 1 + sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/probe.c | 286 ++++++++++++++++++++++++++++++++++ sound/soc/sof/probe.h | 85 ++++++++++ 6 files changed, 394 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index bf3edd9c08b46d..b79479575cc890 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -51,6 +51,7 @@ #define SOF_IPC_GLB_TRACE_MSG SOF_GLB_TYPE(0x9U) #define SOF_IPC_GLB_GDB_DEBUG SOF_GLB_TYPE(0xAU) #define SOF_IPC_GLB_TEST_MSG SOF_GLB_TYPE(0xBU) +#define SOF_IPC_GLB_PROBE SOF_GLB_TYPE(0xCU) /* * DSP Command Message Types @@ -102,6 +103,16 @@ #define SOF_IPC_STREAM_VORBIS_PARAMS SOF_CMD_TYPE(0x010) #define SOF_IPC_STREAM_VORBIS_FREE SOF_CMD_TYPE(0x011) +/* probe */ +#define SOF_IPC_PROBE_INIT SOF_CMD_TYPE(0x001) +#define SOF_IPC_PROBE_DEINIT SOF_CMD_TYPE(0x002) +#define SOF_IPC_PROBE_DMA_ADD SOF_CMD_TYPE(0x003) +#define SOF_IPC_PROBE_DMA_INFO SOF_CMD_TYPE(0x004) +#define SOF_IPC_PROBE_DMA_REMOVE SOF_CMD_TYPE(0x005) +#define SOF_IPC_PROBE_POINT_ADD SOF_CMD_TYPE(0x006) +#define SOF_IPC_PROBE_POINT_INFO SOF_CMD_TYPE(0x007) +#define SOF_IPC_PROBE_POINT_REMOVE SOF_CMD_TYPE(0x008) + /* trace */ #define SOF_IPC_TRACE_DMA_PARAMS SOF_CMD_TYPE(0x001) #define SOF_IPC_TRACE_DMA_POSITION SOF_CMD_TYPE(0x002) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 827b0ec92522f6..65c3cfbcb812f7 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -41,6 +41,14 @@ config SND_SOC_SOF_OF required to enable i.MX8 devices. Say Y if you need this option. If unsure select "N". +config SND_SOC_SOF_DEBUG_PROBES + bool "SOF enable data probing" + help + This option enables the data probing feature that can be used to + gather data directly from specific points of the audio pipeline. + Say Y if you want to enable probes. + If unsure, select "N". + config SND_SOC_SOF_DEVELOPER_SUPPORT bool "SOF developer options support" depends on EXPERT diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 0a8bc72c28a5a3..18d7cab9046e4c 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,6 +2,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o utils.o sof-audio.o +snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o snd-sof-pci-objs := sof-pci-dev.o snd-sof-acpi-objs := sof-acpi-dev.o diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 1837f66e361fd5..922052883b0ad5 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -106,7 +106,9 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) ret = reply.error; } else { /* reply correct size ? */ - if (reply.hdr.size != msg->reply_size) { + if (reply.hdr.size != msg->reply_size && + /* getter payload is never known upfront */ + !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", msg->reply_size, reply.hdr.size); ret = -EINVAL; diff --git a/sound/soc/sof/probe.c b/sound/soc/sof/probe.c new file mode 100644 index 00000000000000..2b2f3dcfc7e9fd --- /dev/null +++ b/sound/soc/sof/probe.c @@ -0,0 +1,286 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski +// + +#include "sof-priv.h" +#include "probe.h" + +/** + * sof_ipc_probe_init - initialize data probing + * @sdev: SOF sound device + * @stream_tag: Extractor stream tag + * @buffer_size: DMA buffer size to set for extractor + * + * Host chooses whether extraction is supported or not by providing + * valid stream tag to DSP. Once specified, stream described by that + * tag will be tied to DSP for extraction for the entire lifetime of + * probe. + * + * Probing is initialized only once and each INIT request must be + * matched by DEINIT call. + */ +int sof_ipc_probe_init(struct snd_sof_dev *sdev, + u32 stream_tag, size_t buffer_size) +{ + struct sof_ipc_probe_dma_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, dma, 1); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_INIT; + msg->num_elems = 1; + msg->dma[0].stream_tag = stream_tag; + msg->dma[0].dma_buffer_size = buffer_size; + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_init); + +/** + * sof_ipc_probe_deinit - cleanup after data probing + * @sdev: SOF sound device + * + * Host sends DEINIT request to free previously initialized probe + * on DSP side once it is no longer needed. DEINIT only when there + * are no probes connected and with all injectors detached. + */ +int sof_ipc_probe_deinit(struct snd_sof_dev *sdev) +{ + struct sof_ipc_cmd_hdr msg; + struct sof_ipc_reply reply; + + msg.size = sizeof(msg); + msg.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DEINIT; + + return sof_ipc_tx_message(sdev->ipc, msg.cmd, &msg, msg.size, + &reply, sizeof(reply)); +} +EXPORT_SYMBOL(sof_ipc_probe_deinit); + +static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd, + void **params, size_t *num_params) +{ + struct sof_ipc_probe_info_params msg = {{{0}}}; + struct sof_ipc_probe_info_params *reply; + size_t bytes; + int ret; + + *params = NULL; + *num_params = 0; + + reply = kzalloc(SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); + if (!reply) + return -ENOMEM; + msg.rhdr.hdr.size = sizeof(msg); + msg.rhdr.hdr.cmd = SOF_IPC_GLB_PROBE | cmd; + + ret = sof_ipc_tx_message(sdev->ipc, msg.rhdr.hdr.cmd, &msg, + msg.rhdr.hdr.size, reply, SOF_IPC_MSG_MAX_SIZE); + if (ret < 0 || reply->rhdr.error < 0) + goto exit; + + if (!reply->num_elems) + goto exit; + + bytes = reply->num_elems * sizeof(reply->dma[0]); + *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL); + if (!*params) { + ret = -ENOMEM; + goto exit; + } + *num_params = msg.num_elems; + +exit: + kfree(reply); + return ret; +} + +/** + * sof_ipc_probe_dma_info - retrieve list of active injection dmas + * @sdev: SOF sound device + * @dma: Returned list of active dmas + * @num_dma: Returned count of active dmas + * + * Host sends DMA_INFO request to obtain list of injection dmas it + * can use to transfer data over with. + * + * Note that list contains only injection dmas as there is only one + * extractor (dma) and it is always assigned on probing init. + * DSP knows exactly where data from extraction probes is going to, + * which is not the case for injection where multiple streams + * could be engaged. + */ +int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev, + struct sof_probe_dma **dma, size_t *num_dma) +{ + return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_DMA_INFO, + (void **)dma, num_dma); +} +EXPORT_SYMBOL(sof_ipc_probe_dma_info); + +/** + * sof_ipc_probe_dma_add - attach to specified dmas + * @sdev: SOF sound device + * @dma: List of streams (dmas) to attach to + * @num_dma: Number of elements in @dma + * + * Contrary to extraction, injection streams are never assigned + * on init. Before attempting any data injection, host is responsible + * for specifying streams which will be later used to transfer data + * to connected probe points. + */ +int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev, + struct sof_probe_dma *dma, size_t num_dma) +{ + struct sof_ipc_probe_dma_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, dma, num_dma); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_dma; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_ADD; + memcpy(&msg->dma[0], dma, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_dma_add); + +/** + * sof_ipc_probe_dma_remove - detach from specified dmas + * @sdev: SOF sound device + * @stream_tag: List of stream tags to detach from + * @num_stream_tag: Number of elements in @stream_tag + * + * Host sends DMA_REMOVE request to free previously attached stream + * from being occupied for injection. Each detach operation should + * match equivalent DMA_ADD. Detach only when all probes tied to + * given stream have been disconnected. + */ +int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev, + unsigned int *stream_tag, size_t num_stream_tag) +{ + struct sof_ipc_probe_dma_remove_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, stream_tag, num_stream_tag); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_stream_tag; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_REMOVE; + memcpy(&msg->stream_tag[0], stream_tag, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_dma_remove); + +/** + * sof_ipc_probe_points_info - retrieve list of active probe points + * @sdev: SOF sound device + * @desc: Returned list of active probes + * @num_desc: Returned count of active probes + * + * Host sends PROBE_POINT_INFO request to obtain list of active probe + * points, valid for disconnection when given probe is no longer + * required. + */ +int sof_ipc_probe_points_info(struct snd_sof_dev *sdev, + struct sof_probe_point_desc **desc, size_t *num_desc) +{ + return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_POINT_INFO, + (void **)desc, num_desc); +} +EXPORT_SYMBOL(sof_ipc_probe_points_info); + +/** + * sof_ipc_probe_points_add - connect specified probes + * @sdev: SOF sound device + * @desc: List of probe points to connect + * @num_desc: Number of elements in @desc + * + * Dynamically connects to provided set of endpoints. Immediately + * after connection is established, host must be prepared to + * transfer data from or to target stream given the probing purpose. + * + * Each probe point should be removed using PROBE_POINT_REMOVE + * request when no longer needed. + */ +int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, + struct sof_probe_point_desc *desc, size_t num_desc) +{ + struct sof_ipc_probe_point_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, desc, num_desc); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_desc; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_ADD; + memcpy(&msg->desc[0], desc, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_points_add); + +/** + * sof_ipc_probe_points_remove - disconnect specified probes + * @sdev: SOF sound device + * @buffer_id: List of probe points to disconnect + * @num_buffer_id: Number of elements in @desc + * + * Removes previously connected probes from list of active probe + * points and frees all resources on DSP side. + */ +int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, + unsigned int *buffer_id, size_t num_buffer_id) +{ + struct sof_ipc_probe_point_remove_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, buffer_id, num_buffer_id); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_buffer_id; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_REMOVE; + memcpy(&msg->buffer_id[0], buffer_id, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_points_remove); diff --git a/sound/soc/sof/probe.h b/sound/soc/sof/probe.h new file mode 100644 index 00000000000000..45daa555283420 --- /dev/null +++ b/sound/soc/sof/probe.h @@ -0,0 +1,85 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2019-2020 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski + */ + +#ifndef __SOF_PROBE_H +#define __SOF_PROBE_H + +#include + +struct snd_sof_dev; + +#define SOF_PROBE_INVALID_NODE_ID UINT_MAX + +struct sof_probe_dma { + unsigned int stream_tag; + unsigned int dma_buffer_size; +} __packed; + +enum sof_connection_purpose { + SOF_CONNECTION_PURPOSE_EXTRACT = 1, + SOF_CONNECTION_PURPOSE_INJECT, +}; + +struct sof_probe_point_desc { + unsigned int buffer_id; + unsigned int purpose; + unsigned int stream_tag; +} __packed; + +struct sof_ipc_probe_dma_add_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + struct sof_probe_dma dma[0]; +} __packed; + +struct sof_ipc_probe_info_params { + struct sof_ipc_reply rhdr; + unsigned int num_elems; + union { + struct sof_probe_dma dma[0]; + struct sof_probe_point_desc desc[0]; + }; +} __packed; + +struct sof_ipc_probe_dma_remove_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + unsigned int stream_tag[0]; +} __packed; + +struct sof_ipc_probe_point_add_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + struct sof_probe_point_desc desc[0]; +} __packed; + +struct sof_ipc_probe_point_remove_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + unsigned int buffer_id[0]; +} __packed; + +int sof_ipc_probe_init(struct snd_sof_dev *sdev, + u32 stream_tag, size_t buffer_size); +int sof_ipc_probe_deinit(struct snd_sof_dev *sdev); +int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev, + struct sof_probe_dma **dma, size_t *num_dma); +int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev, + struct sof_probe_dma *dma, size_t num_dma); +int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev, + unsigned int *stream_tag, size_t num_stream_tag); +int sof_ipc_probe_points_info(struct snd_sof_dev *sdev, + struct sof_probe_point_desc **desc, size_t *num_desc); +int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, + struct sof_probe_point_desc *desc, size_t num_desc); +int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, + unsigned int *buffer_id, size_t num_buffer_id); + +#endif From e145e9af231adff081e0e16e1dacfb6e3c4e968f Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:20 +0100 Subject: [PATCH 112/415] ASoC: SOF: Generic probe compress operations Define system-agnostic probe compress flow which serves as a base for actual, hardware-dependent implementations. As per firmware spec, maximum of one extraction stream is allowed, while for injection, there can be plenty. Apart from probe_pointer, all probe compress operations are mandatory. Copy operation is defined as unified as its flow should be shared across all SOF systems. Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 1 + sound/soc/sof/Makefile | 2 +- sound/soc/sof/compress.c | 141 +++++++++++++++++++++++++++++++++++++++ sound/soc/sof/compress.h | 29 ++++++++ sound/soc/sof/core.c | 6 ++ sound/soc/sof/ops.h | 43 ++++++++++++ sound/soc/sof/sof-priv.h | 25 +++++++ 7 files changed, 246 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 65c3cfbcb812f7..4dda4b62509f04 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_SOF_OF config SND_SOC_SOF_DEBUG_PROBES bool "SOF enable data probing" + select SND_SOC_COMPRESS help This option enables the data probing feature that can be used to gather data directly from specific points of the audio pipeline. diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 18d7cab9046e4c..8eca2f85c90e39 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,7 +2,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o utils.o sof-audio.o -snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o +snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o compress.o snd-sof-pci-objs := sof-pci-dev.o snd-sof-acpi-objs := sof-acpi-dev.o diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c new file mode 100644 index 00000000000000..e87cc81a0599ed --- /dev/null +++ b/sound/soc/sof/compress.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski +// + +#include +#include "compress.h" +#include "ops.h" +#include "probe.h" + +int sof_probe_compr_open(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + int ret; + + ret = snd_sof_probe_compr_assign(sdev, cstream, dai); + if (ret < 0) { + dev_err(dai->dev, "Failed to assign probe stream: %d\n", ret); + return ret; + } + + sdev->extractor_stream_tag = ret; + return 0; +} +EXPORT_SYMBOL(sof_probe_compr_open); + +int sof_probe_compr_free(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + struct sof_probe_point_desc *desc; + size_t num_desc; + int i, ret; + + /* disconnect all probe points */ + ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc); + if (ret < 0) { + dev_err(dai->dev, "Failed to get probe points: %d\n", ret); + goto exit; + } + + for (i = 0; i < num_desc; i++) + sof_ipc_probe_points_remove(sdev, &desc[i].buffer_id, 1); + kfree(desc); + +exit: + ret = sof_ipc_probe_deinit(sdev); + if (ret < 0) + dev_err(dai->dev, "Failed to deinit probe: %d\n", ret); + + sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID; + snd_compr_free_pages(cstream); + + return snd_sof_probe_compr_free(sdev, cstream, dai); +} +EXPORT_SYMBOL(sof_probe_compr_free); + +int sof_probe_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + int ret; + + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; + cstream->dma_buffer.dev.dev = sdev->dev; + ret = snd_compr_malloc_pages(cstream, rtd->buffer_size); + if (ret < 0) + return ret; + + ret = snd_sof_probe_compr_set_params(sdev, cstream, params, dai); + if (ret < 0) + return ret; + + ret = sof_ipc_probe_init(sdev, sdev->extractor_stream_tag, + rtd->dma_bytes); + if (ret < 0) { + dev_err(dai->dev, "Failed to init probe: %d\n", ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL(sof_probe_compr_set_params); + +int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + return snd_sof_probe_compr_trigger(sdev, cstream, cmd, dai); +} +EXPORT_SYMBOL(sof_probe_compr_trigger); + +int sof_probe_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai); +} +EXPORT_SYMBOL(sof_probe_compr_pointer); + +int sof_probe_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + unsigned int offset, n; + void *ptr; + int ret; + + if (count > rtd->buffer_size) + count = rtd->buffer_size; + + div_u64_rem(rtd->total_bytes_transferred, rtd->buffer_size, &offset); + ptr = rtd->dma_area + offset; + n = rtd->buffer_size - offset; + + if (count < n) { + ret = copy_to_user(buf, ptr, count); + } else { + ret = copy_to_user(buf, ptr, n); + ret += copy_to_user(buf + n, rtd->dma_area, count - n); + } + + if (ret) + return count - ret; + return count; +} +EXPORT_SYMBOL(sof_probe_compr_copy); diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h new file mode 100644 index 00000000000000..dccc9e008f819c --- /dev/null +++ b/sound/soc/sof/compress.h @@ -0,0 +1,29 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2019-2020 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski + */ + +#ifndef __SOF_COMPRESS_H +#define __SOF_COMPRESS_H + +#include + +int sof_probe_compr_open(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int sof_probe_compr_free(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int sof_probe_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai); +int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); +int sof_probe_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai); +int sof_probe_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count); + +#endif diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 1d07450aff7774..91acfae7935c99 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -14,6 +14,9 @@ #include #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "probe.h" +#endif /* see SOF_DBG_ flags */ int sof_core_debug; @@ -292,6 +295,9 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) sdev->pdata = plat_data; sdev->first_boot = true; sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID; +#endif dev_set_drvdata(dev, sdev); /* check all mandatory ops */ diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 7f532bcc8e9dd3..a771500ac44241 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -393,6 +393,49 @@ snd_sof_pcm_platform_pointer(struct snd_sof_dev *sdev, return 0; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +static inline int +snd_sof_probe_compr_assign(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_assign(sdev, cstream, dai); +} + +static inline int +snd_sof_probe_compr_free(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_free(sdev, cstream, dai); +} + +static inline int +snd_sof_probe_compr_set_params(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_set_params(sdev, cstream, params, dai); +} + +static inline int +snd_sof_probe_compr_trigger(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_trigger(sdev, cstream, cmd, dai); +} + +static inline int +snd_sof_probe_compr_pointer(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) +{ + if (sof_ops(sdev) && sof_ops(sdev)->probe_pointer) + return sof_ops(sdev)->probe_pointer(sdev, cstream, tstamp, dai); + + return 0; +} +#endif + /* machine driver */ static inline int snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 00084471d0de19..5d16f668d16aa7 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -170,6 +170,27 @@ struct snd_sof_dsp_ops { snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + /* Except for probe_pointer, all probe ops are mandatory */ + int (*probe_assign)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_free)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_set_params)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_trigger)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_pointer)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, + struct snd_soc_dai *dai); /* optional */ +#endif + /* host read DSP stream data */ void (*ipc_msg_data)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, @@ -405,6 +426,10 @@ struct snd_sof_dev { wait_queue_head_t waitq; int code_loading; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + unsigned int extractor_stream_tag; +#endif + /* DMA for Trace */ struct snd_dma_buffer dmatb; struct snd_dma_buffer dmatp; From 49d7948ed174cc170041bf3d22e1f085fd8b87f0 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:21 +0100 Subject: [PATCH 113/415] ASoC: SOF: Intel: Expose SDnFMT helpers Hda stream is setup in similar fashion for compress as it is for pcm operations. To reuse existing code in compress path, expose SDnFMT helper routines. Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-pcm.c | 8 ++++---- sound/soc/sof/intel/hda.h | 2 ++ 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 23872f6e708dc6..a46a6baa1c3f2e 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -27,7 +27,7 @@ #define SDnFMT_BITS(x) ((x) << 4) #define SDnFMT_CHAN(x) ((x) << 0) -static inline u32 get_mult_div(struct snd_sof_dev *sdev, int rate) +u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate) { switch (rate) { case 8000: @@ -61,7 +61,7 @@ static inline u32 get_mult_div(struct snd_sof_dev *sdev, int rate) } }; -static inline u32 get_bits(struct snd_sof_dev *sdev, int sample_bits) +u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits) { switch (sample_bits) { case 8: @@ -95,8 +95,8 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, u32 size, rate, bits; size = params_buffer_bytes(params); - rate = get_mult_div(sdev, params_rate(params)); - bits = get_bits(sdev, params_width(params)); + rate = hda_dsp_get_mult_div(sdev, params_rate(params)); + bits = hda_dsp_get_bits(sdev, params_width(params)); hstream->substream = substream; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index a46b66437a3d20..2b5fde372790c6 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -502,6 +502,8 @@ void hda_dsp_d0i3_work(struct work_struct *work); /* * DSP PCM Operations. */ +u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate); +u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits); int hda_dsp_pcm_open(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); int hda_dsp_pcm_close(struct snd_sof_dev *sdev, From 4c414da93a4642d02c67fbe82f1834be7bf586b7 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:22 +0100 Subject: [PATCH 114/415] ASoC: SOF: Intel: Probe compress operations Add HDA handlers for soc_compr_ops and snd_compr_ops which cover probe related operations. Implementation supports both connection purposes. These merely define stream setups as core flow is covered by SOF compress core. Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 9 +++ sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 +++ sound/soc/sof/intel/cnl.c | 9 +++ sound/soc/sof/intel/hda-compress.c | 114 +++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 24 ++++++ 6 files changed, 166 insertions(+) create mode 100644 sound/soc/sof/intel/hda-compress.c diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 56a837d2cb95bb..3bc64dee7c39c3 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -305,6 +305,15 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC Say Y if you want to enable HDAudio codecs with SOF. If unsure select "N". +config SND_SOC_SOF_HDA_PROBES + bool "SOF enable probes over HDA" + depends on SND_SOC_SOF_DEBUG_PROBES + help + This option enables the data probing for Intel(R). + Intel(R) Skylake and newer platforms. + Say Y if you want to enable probes. + If unsure, select "N". + config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1 bool "SOF enable DMI Link L1" help diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index b8f58e006e29e5..cee02a2e00f402 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -9,6 +9,7 @@ snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \ hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ hda-dai.o hda-bus.o \ apl.o cnl.o +snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-compress.o snd-sof-intel-hda-objs := hda-codec.o diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 2483b15699e73e..02218d22e51f49 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -73,6 +73,15 @@ const struct snd_sof_dsp_ops sof_apl_ops = { .pcm_trigger = hda_dsp_pcm_trigger, .pcm_pointer = hda_dsp_pcm_pointer, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) + /* probe callbacks */ + .probe_assign = hda_probe_compr_assign, + .probe_free = hda_probe_compr_free, + .probe_set_params = hda_probe_compr_set_params, + .probe_trigger = hda_probe_compr_trigger, + .probe_pointer = hda_probe_compr_pointer, +#endif + /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 8a59fec7291999..05125cb0be6e28 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -284,6 +284,15 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { .pcm_trigger = hda_dsp_pcm_trigger, .pcm_pointer = hda_dsp_pcm_pointer, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) + /* probe callbacks */ + .probe_assign = hda_probe_compr_assign, + .probe_free = hda_probe_compr_free, + .probe_set_params = hda_probe_compr_set_params, + .probe_trigger = hda_probe_compr_trigger, + .probe_pointer = hda_probe_compr_pointer, +#endif + /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-compress.c b/sound/soc/sof/intel/hda-compress.c new file mode 100644 index 00000000000000..38a1ebec84780b --- /dev/null +++ b/sound/soc/sof/intel/hda-compress.c @@ -0,0 +1,114 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski +// + +#include +#include +#include "../sof-priv.h" +#include "hda.h" + +static inline struct hdac_ext_stream * +hda_compr_get_stream(struct snd_compr_stream *cstream) +{ + return cstream->runtime->private_data; +} + +int hda_probe_compr_assign(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream; + + stream = hda_dsp_stream_get(sdev, cstream->direction); + if (!stream) + return -EBUSY; + + hdac_stream(stream)->curr_pos = 0; + hdac_stream(stream)->cstream = cstream; + cstream->runtime->private_data = stream; + + return hdac_stream(stream)->stream_tag; +} + +int hda_probe_compr_free(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + int ret; + + ret = hda_dsp_stream_put(sdev, cstream->direction, + hdac_stream(stream)->stream_tag); + if (ret < 0) { + dev_dbg(sdev->dev, "stream put failed: %d\n", ret); + return ret; + } + + hdac_stream(stream)->cstream = NULL; + cstream->runtime->private_data = NULL; + + return 0; +} + +int hda_probe_compr_set_params(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + struct hdac_stream *hstream = hdac_stream(stream); + struct snd_dma_buffer *dmab; + u32 bits, rate; + int bps, ret; + + dmab = cstream->runtime->dma_buffer_p; + /* compr params do not store bit depth, default to S32_LE */ + bps = snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32_LE); + if (bps < 0) + return bps; + bits = hda_dsp_get_bits(sdev, bps); + rate = hda_dsp_get_mult_div(sdev, params->codec.sample_rate); + + hstream->format_val = rate | bits | (params->codec.ch_out - 1); + hstream->bufsize = cstream->runtime->buffer_size; + hstream->period_bytes = cstream->runtime->fragment_size; + hstream->no_period_wakeup = 0; + + ret = hda_dsp_stream_hw_params(sdev, stream, dmab, NULL); + if (ret < 0) { + dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret); + return ret; + } + + return 0; +} + +int hda_probe_compr_trigger(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + + return hda_dsp_stream_trigger(sdev, stream, cmd); +} + +int hda_probe_compr_pointer(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + struct snd_soc_pcm_stream *pstream; + + pstream = &dai->driver->capture; + tstamp->copied_total = hdac_stream(stream)->curr_pos; + tstamp->sampling_rate = snd_pcm_rate_bit_to_rate(pstream->rates); + + return 0; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 2b5fde372790c6..ca44ecb7653448 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -11,6 +11,7 @@ #ifndef __SOF_INTEL_HDA_H #define __SOF_INTEL_HDA_H +#include #include #include #include "shim.h" @@ -552,6 +553,29 @@ int hda_ipc_pcm_params(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, const struct sof_ipc_pcm_params_reply *reply); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +/* + * Probe Compress Operations. + */ +int hda_probe_compr_assign(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int hda_probe_compr_free(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int hda_probe_compr_set_params(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, + struct snd_soc_dai *dai); +int hda_probe_compr_trigger(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); +int hda_probe_compr_pointer(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, + struct snd_soc_dai *dai); +#endif + /* * DSP IPC Operations. */ From 394695f410c1cd3208906451ba5420f45c420a51 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:23 +0100 Subject: [PATCH 115/415] ASoC: SOF: Provide probe debugfs support Define debugfs subdirectory delegated for IPC communication with DSP. Input format: uint,uint,(...) which are later translated into DWORDS sequence and further into instances of struct of interest given the IPC type. For Extractor probes, following have been enabled: - PROBE_POINT_ADD (echo <..> probe_points) - PROBE_POINT_REMOVE (echo <..> probe_points_remove) - PROBE_POINT_INFO (cat probe_points) Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/sof/debug.c | 226 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 226 insertions(+) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index d2b3b99d3a2044..b5c0d6cf72cc62 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -17,6 +17,221 @@ #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "probe.h" + +/** + * strsplit_u32 - Split string into sequence of u32 tokens + * @buf: String to split into tokens. + * @delim: String containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + */ +static int +strsplit_u32(char **buf, const char *delim, u32 **tkns, size_t *num_tkns) +{ + char *s; + u32 *data, *tmp; + size_t count = 0; + size_t cap = 32; + int ret = 0; + + *tkns = NULL; + *num_tkns = 0; + data = kcalloc(cap, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + while ((s = strsep(buf, delim)) != NULL) { + ret = kstrtouint(s, 0, data + count); + if (ret) + goto exit; + if (++count >= cap) { + cap *= 2; + tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL); + if (!tmp) { + ret = -ENOMEM; + goto exit; + } + data = tmp; + } + } + + if (!count) + goto exit; + *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL); + if (*tkns == NULL) { + ret = -ENOMEM; + goto exit; + } + *num_tkns = count; + +exit: + kfree(data); + return ret; +} + +static int tokenize_input(const char __user *from, size_t count, + loff_t *ppos, u32 **tkns, size_t *num_tkns) +{ + char *buf; + int ret; + + buf = kmalloc(count + 1, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = simple_write_to_buffer(buf, count, ppos, from, count); + if (ret != count) { + ret = ret >= 0 ? -EIO : ret; + goto exit; + } + + buf[count] = '\0'; + ret = strsplit_u32((char **)&buf, ",", tkns, num_tkns); +exit: + kfree(buf); + return ret; +} + +static ssize_t probe_points_read(struct file *file, + char __user *to, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + struct sof_probe_point_desc *desc; + size_t num_desc, len = 0; + char *buf; + int i, ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + buf = kzalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc); + if (ret < 0) + goto exit; + + for (i = 0; i < num_desc; i++) { + ret = snprintf(buf + len, PAGE_SIZE - len, + "Id: %#010x Purpose: %d Node id: %#x\n", + desc[i].buffer_id, desc[i].purpose, desc[i].stream_tag); + if (ret < 0) + goto free_desc; + len += ret; + } + + ret = simple_read_from_buffer(to, count, ppos, buf, len); +free_desc: + kfree(desc); +exit: + kfree(buf); + return ret; +} + +static ssize_t probe_points_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + struct sof_probe_point_desc *desc; + size_t num_tkns, bytes; + u32 *tkns; + int ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + ret = tokenize_input(from, count, ppos, &tkns, &num_tkns); + if (ret < 0) + return ret; + bytes = sizeof(*tkns) * num_tkns; + if (!num_tkns || (bytes % sizeof(*desc))) { + ret = -EINVAL; + goto exit; + } + + desc = (struct sof_probe_point_desc *)tkns; + ret = sof_ipc_probe_points_add(sdev, + desc, bytes / sizeof(*desc)); + if (!ret) + ret = count; +exit: + kfree(tkns); + return ret; +} + +static const struct file_operations probe_points_fops = { + .open = simple_open, + .read = probe_points_read, + .write = probe_points_write, + .llseek = default_llseek, +}; + +static ssize_t probe_points_remove_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + size_t num_tkns; + u32 *tkns; + int ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + ret = tokenize_input(from, count, ppos, &tkns, &num_tkns); + if (ret < 0) + return ret; + if (!num_tkns) { + ret = -EINVAL; + goto exit; + } + + ret = sof_ipc_probe_points_remove(sdev, tkns, num_tkns); + if (!ret) + ret = count; +exit: + kfree(tkns); + return ret; +} + +static const struct file_operations probe_points_remove_fops = { + .open = simple_open, + .write = probe_points_remove_write, + .llseek = default_llseek, +}; + +static int snd_sof_debugfs_probe_item(struct snd_sof_dev *sdev, + const char *name, mode_t mode, + const struct file_operations *fops) +{ + struct snd_sof_dfsentry *dfse; + + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); + if (!dfse) + return -ENOMEM; + + dfse->type = SOF_DFSENTRY_TYPE_BUF; + dfse->sdev = sdev; + + debugfs_create_file(name, mode, sdev->debugfs_root, dfse, fops); + /* add to dfsentry list */ + list_add(&dfse->list, &sdev->dfsentry_list); + + return 0; +} +#endif + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) #define MAX_IPC_FLOOD_DURATION_MS 1000 #define MAX_IPC_FLOOD_COUNT 10000 @@ -436,6 +651,17 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev) return err; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + err = snd_sof_debugfs_probe_item(sdev, "probe_points", + 0644, &probe_points_fops); + if (err < 0) + return err; + err = snd_sof_debugfs_probe_item(sdev, "probe_points_remove", + 0200, &probe_points_remove_fops); + if (err < 0) + return err; +#endif + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) /* create read-write ipc_flood_count debugfs entry */ err = snd_sof_debugfs_buf_item(sdev, NULL, 0, From 70368106467cd8c420176bf3ab0acc797f6584bf Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 18 Feb 2020 15:39:24 +0100 Subject: [PATCH 116/415] ASoC: SOF: Intel: Add Probe compress CPU DAIs Declare extraction CPU DAI as well as sof_probe_compr_ops. FE DAIs can link against these new CPU DAI to create new compress devices. Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200218143924.10565-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/sof/compress.c | 5 +++++ sound/soc/sof/compress.h | 2 ++ sound/soc/sof/intel/hda-dai.c | 28 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 6 ++++++ sound/soc/sof/pcm.c | 7 +++++++ 5 files changed, 48 insertions(+) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index e87cc81a0599ed..7354dc6a49cf88 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -13,6 +13,11 @@ #include "ops.h" #include "probe.h" +struct snd_compr_ops sof_probe_compressed_ops = { + .copy = sof_probe_compr_copy, +}; +EXPORT_SYMBOL(sof_probe_compressed_ops); + int sof_probe_compr_open(struct snd_compr_stream *cstream, struct snd_soc_dai *dai) { diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h index dccc9e008f819c..800f163603e1e1 100644 --- a/sound/soc/sof/compress.h +++ b/sound/soc/sof/compress.h @@ -13,6 +13,8 @@ #include +extern struct snd_compr_ops sof_probe_compressed_ops; + int sof_probe_compr_open(struct snd_compr_stream *cstream, struct snd_soc_dai *dai); int sof_probe_compr_free(struct snd_compr_stream *cstream, diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 9c6e3f990ee31d..ed5e7d2c0d432a 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -399,6 +399,19 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = { .trigger = hda_link_pcm_trigger, .prepare = hda_link_pcm_prepare, }; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +#include "../compress.h" + +static struct snd_soc_cdai_ops sof_probe_compr_ops = { + .startup = sof_probe_compr_open, + .shutdown = sof_probe_compr_free, + .set_params = sof_probe_compr_set_params, + .trigger = sof_probe_compr_trigger, + .pointer = sof_probe_compr_pointer, +}; + +#endif #endif /* @@ -460,5 +473,20 @@ struct snd_soc_dai_driver skl_dai[] = { .name = "Alt Analog CPU DAI", .ops = &hda_link_dai_ops, }, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +{ + .name = "Probe Extraction CPU DAI", + .compress_new = snd_soc_new_compress, + .cops = &sof_probe_compr_ops, + .capture = { + .stream_name = "Probe Extraction", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + }, +}, +#endif #endif }; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index ca44ecb7653448..537c0a930a15c8 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -349,7 +349,13 @@ /* Number of DAIs */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +#define SOF_SKL_NUM_DAIS 16 +#else #define SOF_SKL_NUM_DAIS 15 +#endif + #else #define SOF_SKL_NUM_DAIS 8 #endif diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index db3df02c7398f1..b239bbff4b5cda 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -16,6 +16,9 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "compress.h" +#endif /* Create DMA buffer page table for DSP */ static int create_page_table(struct snd_soc_component *component, @@ -787,6 +790,10 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS) pd->compr_ops = &sof_compressed_ops; +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + /* override cops when probe support is enabled */ + pd->compr_ops = &sof_probe_compressed_ops; #endif pd->pcm_construct = sof_pcm_new; pd->ignore_machine = drv_name; From ebbfabc16d23dfd20eecd4b6e68212fec37ae7c6 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Tue, 18 Feb 2020 21:51:51 +0800 Subject: [PATCH 117/415] ASoC: rt5682: Add CCF usage for providing I2S clks There is a need to use RT5682 as DAI clock master for other codecs within a platform, which means that the DAI clocks are required to remain, regardless of whether the RT5682 is actually running playback/capture. The RT5682 CCF basic functions are implemented almost by the existing internal functions and asoc apis. It needs a clk provider (rt5682 mclk) to generate the bclk and wclk outputs. The RT5682 CCF supports and restricts as below: 1. Fmt of DAI-AIF1 must be configured to master before using CCF. 2. Only accept a 48MHz clk as the clk provider. 3. Only provide a 48kHz wclk and a set of multiples of wclk as bclk. There are some temporary limitations in this patch until a better implementation. Signed-off-by: Derek Fang Link: https://lore.kernel.org/r/1582033912-6841-1-git-send-email-derek.fang@realtek.com Signed-off-by: Mark Brown --- include/sound/rt5682.h | 8 + sound/soc/codecs/rt5682.c | 407 +++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5682.h | 4 +- 3 files changed, 415 insertions(+), 4 deletions(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index bc2c31734df118..6bf0e3581056eb 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -24,6 +24,12 @@ enum rt5682_jd_src { RT5682_JD1, }; +enum rt5682_dai_clks { + RT5682_DAI_WCLK_IDX, + RT5682_DAI_BCLK_IDX, + RT5682_DAI_NUM_CLKS, +}; + struct rt5682_platform_data { int ldo1_en; /* GPIO for LDO1_EN */ @@ -32,6 +38,8 @@ struct rt5682_platform_data { enum rt5682_dmic1_clk_pin dmic1_clk_pin; enum rt5682_jd_src jd_src; unsigned int btndet_delay; + + const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; #endif diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 9fbb3862f8d70d..6774813e0eea53 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -27,6 +27,9 @@ #include #include #include +#include +#include +#include #include #include "rl6231.h" @@ -45,6 +48,8 @@ static const struct rt5682_platform_data i2s_default_platform_data = { .dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3, .jd_src = RT5682_JD1, .btndet_delay = 16, + .dai_clk_names[RT5682_DAI_WCLK_IDX] = "rt5682-dai-wclk", + .dai_clk_names[RT5682_DAI_BCLK_IDX] = "rt5682-dai-bclk", }; struct rt5682_priv { @@ -58,6 +63,13 @@ struct rt5682_priv { struct mutex calibrate_mutex; bool is_sdw; +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; + struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; + struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + int sysclk; int sysclk_src; int lrck[RT5682_AIFS]; @@ -921,6 +933,7 @@ static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = &component->dapm; unsigned int val, count; if (jack_insert) { @@ -963,8 +976,13 @@ static int rt5682_headset_detect(struct snd_soc_component *component, rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); + else + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -1633,6 +1651,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -2459,12 +2478,380 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, return 0; } +#ifdef CONFIG_COMMON_CLK +#define CLK_PLL2_FIN 48000000 +#define CLK_PLL2_FOUT 24576000 +#define CLK_48 48000 + +static bool rt5682_clk_check(struct rt5682_priv *rt5682) +{ + if (!rt5682->master[RT5682_AIF1]) { + dev_err(rt5682->component->dev, "sysclk/dai not set correctly\n"); + return false; + } + return true; +} + +static int rt5682_wclk_prepare(struct clk_hw *hw) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, RT5682_PWR_MB); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + return 0; +} + +static void rt5682_wclk_unprepare(struct clk_hw *hw) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682_clk_check(rt5682)) + return; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + if (!rt5682->jack_type) + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + + if (!rt5682_clk_check(rt5682)) + return 0; + /* + * Only accept to set wclk rate to 48kHz temporarily. + */ + return CLK_48; +} + +static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + /* + * Only accept to set wclk rate to 48kHz temporarily. + */ + return CLK_48; +} + +static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct clk *parent_clk; + const char * const clk_name = __clk_get_name(hw->clk); + int pre_div; + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + /* + * Whether the wclk's parent clk (mclk) exists or not, please ensure + * it is fixed or set to 48MHz before setting wclk rate. It's a + * temporary limitation. Only accept 48MHz clk as the clk provider. + * + * It will set the codec anyway by assuming mclk is 48MHz. + */ + parent_clk = clk_get_parent(hw->clk); + if (!parent_clk) + dev_warn(component->dev, + "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", + CLK_PLL2_FIN); + + if (parent_rate != CLK_PLL2_FIN) + dev_warn(component->dev, "clk %s only support %d Hz input\n", + clk_name, CLK_PLL2_FIN); + + /* + * It's a temporary limitation. Only accept to set wclk rate to 48kHz. + * It will force wclk to 48kHz even it's not. + */ + if (rate != CLK_48) { + dev_warn(component->dev, "clk %s only support %d Hz output\n", + clk_name, CLK_48); + rate = CLK_48; + } + + /* + * To achieve the rate conversion from 48MHz to 48kHz, PLL2 is needed. + */ + rt5682_set_component_pll(component, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CLK_PLL2_FIN, CLK_PLL2_FOUT); + + rt5682_set_component_sysclk(component, RT5682_SCLK_S_PLL2, 0, + CLK_PLL2_FOUT, SND_SOC_CLOCK_IN); + + pre_div = rl6231_get_clk_info(rt5682->sysclk, rate); + + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, + RT5682_I2S_M_DIV_MASK | RT5682_I2S_CLK_SRC_MASK, + pre_div << RT5682_I2S_M_DIV_SFT | + (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT); + + return 0; +} + +static unsigned long rt5682_bclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + unsigned int bclks_per_wclk; + + snd_soc_component_read(component, RT5682_TDM_TCON_CTRL, + &bclks_per_wclk); + + switch (bclks_per_wclk & RT5682_TDM_BCLK_MS1_MASK) { + case RT5682_TDM_BCLK_MS1_256: + return parent_rate * 256; + case RT5682_TDM_BCLK_MS1_128: + return parent_rate * 128; + case RT5682_TDM_BCLK_MS1_64: + return parent_rate * 64; + case RT5682_TDM_BCLK_MS1_32: + return parent_rate * 32; + default: + return 0; + } +} + +static unsigned long rt5682_bclk_get_factor(unsigned long rate, + unsigned long parent_rate) +{ + unsigned long factor; + + factor = rate / parent_rate; + if (factor < 64) + return 32; + else if (factor < 128) + return 64; + else if (factor < 256) + return 128; + else + return 256; +} + +static long rt5682_bclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + unsigned long factor; + + if (!*parent_rate || !rt5682_clk_check(rt5682)) + return -EINVAL; + + /* + * BCLK rates are set as a multiplier of WCLK in HW. + * We don't allow changing the parent WCLK. We just do + * some rounding down based on the parent WCLK rate + * and find the appropriate multiplier of BCLK to + * get the rounded down BCLK value. + */ + factor = rt5682_bclk_get_factor(rate, *parent_rate); + + return *parent_rate * factor; +} + +static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dai *dai = NULL; + unsigned long factor; + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + factor = rt5682_bclk_get_factor(rate, parent_rate); + + for_each_component_dais(component, dai) + if (dai->id == RT5682_AIF1) + break; + if (!dai) { + dev_err(component->dev, "dai %d not found in component\n", + RT5682_AIF1); + return -ENODEV; + } + + return rt5682_set_bclk1_ratio(dai, factor); +} + +static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = { + [RT5682_DAI_WCLK_IDX] = { + .prepare = rt5682_wclk_prepare, + .unprepare = rt5682_wclk_unprepare, + .recalc_rate = rt5682_wclk_recalc_rate, + .round_rate = rt5682_wclk_round_rate, + .set_rate = rt5682_wclk_set_rate, + }, + [RT5682_DAI_BCLK_IDX] = { + .recalc_rate = rt5682_bclk_recalc_rate, + .round_rate = rt5682_bclk_round_rate, + .set_rate = rt5682_bclk_set_rate, + }, +}; + +static int rt5682_register_dai_clks(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct rt5682_platform_data *pdata = &rt5682->pdata; + struct clk_init_data init; + struct clk *dai_clk; + struct clk_lookup *dai_clk_lookup; + struct clk_hw *dai_clk_hw; + const char *parent_name; + int i, ret; + + for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { + dai_clk_hw = &rt5682->dai_clks_hw[i]; + + switch (i) { + case RT5682_DAI_WCLK_IDX: + /* Make MCLK the parent of WCLK */ + if (rt5682->mclk) { + parent_name = __clk_get_name(rt5682->mclk); + init.parent_names = &parent_name; + init.num_parents = 1; + } else { + init.parent_names = NULL; + init.num_parents = 0; + } + break; + case RT5682_DAI_BCLK_IDX: + /* Make WCLK the parent of BCLK */ + parent_name = __clk_get_name( + rt5682->dai_clks[RT5682_DAI_WCLK_IDX]); + init.parent_names = &parent_name; + init.num_parents = 1; + break; + default: + dev_err(dev, "Invalid clock index\n"); + ret = -EINVAL; + goto err; + } + + init.name = pdata->dai_clk_names[i]; + init.ops = &rt5682_dai_clk_ops[i]; + init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; + dai_clk_hw->init = &init; + + dai_clk = devm_clk_register(dev, dai_clk_hw); + if (IS_ERR(dai_clk)) { + dev_warn(dev, "Failed to register %s: %ld\n", + init.name, PTR_ERR(dai_clk)); + ret = PTR_ERR(dai_clk); + goto err; + } + rt5682->dai_clks[i] = dai_clk; + + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, + dai_clk_hw); + } else { + dai_clk_lookup = clkdev_create(dai_clk, init.name, + "%s", dev_name(dev)); + if (!dai_clk_lookup) { + ret = -ENOMEM; + goto err; + } else { + rt5682->dai_clks_lookup[i] = dai_clk_lookup; + } + } + } + + return 0; + +err: + do { + if (rt5682->dai_clks_lookup[i]) + clkdev_drop(rt5682->dai_clks_lookup[i]); + } while (i-- > 0); + + return ret; +} +#endif /* CONFIG_COMMON_CLK */ + static int rt5682_probe(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); +#ifdef CONFIG_COMMON_CLK + int ret; +#endif rt5682->component = component; +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) { + if (PTR_ERR(rt5682->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682->mclk); + return ret; + } + rt5682->mclk = NULL; + } + + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(component); + if (ret) + return ret; + + /* Initial setup for CCF */ + rt5682->lrck[RT5682_AIF1] = CLK_48; +#endif + return 0; } @@ -2472,6 +2859,15 @@ static void rt5682_remove(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); +#ifdef CONFIG_COMMON_CLK + int i; + + for (i = RT5682_DAI_NUM_CLKS - 1; i >= 0; --i) { + if (rt5682->dai_clks_lookup[i]) + clkdev_drop(rt5682->dai_clks_lookup[i]); + } +#endif + rt5682_reset(rt5682); } @@ -2606,6 +3002,13 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); + if (device_property_read_string_array(dev, "clock-output-names", + rt5682->pdata.dai_clk_names, + RT5682_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], + rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); + return 0; } diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 465c99b7f906d7..f82126a6f2118e 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -841,8 +841,8 @@ #define RT5682_TDM_M_LP_INV (0x1 << 1) #define RT5682_TDM_MS_MASK (0x1 << 0) #define RT5682_TDM_MS_SFT 0 -#define RT5682_TDM_MS_M (0x0 << 0) -#define RT5682_TDM_MS_S (0x1 << 0) +#define RT5682_TDM_MS_S (0x0 << 0) +#define RT5682_TDM_MS_M (0x1 << 0) /* Global Clock Control (0x0080) */ #define RT5682_SCLK_SRC_MASK (0x7 << 13) From 8b59e642d05f0ae9800b057350c063fe7debd6bc Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Tue, 18 Feb 2020 21:51:52 +0800 Subject: [PATCH 118/415] ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage This patch describes that rt5682 can expose WCLK and BCLK clocks and how to use. Signed-off-by: Derek Fang Link: https://lore.kernel.org/r/1582033912-6841-2-git-send-email-derek.fang@realtek.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5682.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 30e927a283690a..ac98151d29e4ba 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -32,6 +32,12 @@ Optional properties: The delay time is realtek,btndet-delay value multiple of 8.192 ms. If absent, the default is 16. +- #clock-cells : Should be set to '<1>', wclk and bclk sources provided. +- clock-output-names : Name given for DAI clocks output. + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 @@ -53,4 +59,10 @@ rt5682 { realtek,dmic1-clk-pin = <1>; realtek,jd-src = <1>; realtek,btndet-delay = <16>; + + #clock-cells = <1>; + clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; + + clocks = <&osc>; + clock-names = "mclk"; }; From d9303690f753dfdae51304fc89f4b04c0549a9f7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:27:39 +0900 Subject: [PATCH 119/415] ASoC: soc-pcm: move dai_get_widget() This patch moves dai_get_widget() to top side. This is prepare for cleanup soc-pcm.c Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87h7zpbouu.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 65a3856be2505c..23e36f4f965c18 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -82,6 +82,15 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd, return 0; } +static inline +struct snd_soc_dapm_widget *dai_get_widget(struct snd_soc_dai *dai, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return dai->playback_widget; + else + return dai->capture_widget; +} + static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { @@ -1287,15 +1296,6 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, return NULL; } -static inline struct snd_soc_dapm_widget * - dai_get_widget(struct snd_soc_dai *dai, int stream) -{ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - return dai->playback_widget; - else - return dai->capture_widget; -} - static int widget_in_list(struct snd_soc_dapm_widget_list *list, struct snd_soc_dapm_widget *widget) { From 93597fae552a35d27cd1f399ffab6a6862cf9dc3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:27:43 +0900 Subject: [PATCH 120/415] ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() dpcm_get_be() has very duplicate code. dpcm_get_be() { ... if (stream == SNDRV_PCM_STREAM_PLAYBACK) { (1) /* code for Playback */ } else { (2) /* code for Capture */ } } The difference between Playback (1) and Capture (2) code is pointer only (= "playback_widget" or "caputre_widget"). OTOH, now we already has dai_get_widget() for it. This means we can merge (1) and (2). This patch do it and remove duplicated code. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87ftf9bouq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 41 ++++++++++++----------------------------- 1 file changed, 12 insertions(+), 29 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 23e36f4f965c18..b708db9723107c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1246,47 +1246,30 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; + struct snd_soc_dapm_widget *w; struct snd_soc_dai *dai; int i; dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - for_each_card_rtds(card, be) { + for_each_card_rtds(card, be) { - if (!be->dai_link->no_pcm) - continue; + if (!be->dai_link->no_pcm) + continue; - dev_dbg(card->dev, "ASoC: try BE : %s\n", - be->cpu_dai->playback_widget ? - be->cpu_dai->playback_widget->name : "(not set)"); + w = dai_get_widget(be->cpu_dai, stream); - if (be->cpu_dai->playback_widget == widget) - return be; + dev_dbg(card->dev, "ASoC: try BE : %s\n", + w ? w->name : "(not set)"); - for_each_rtd_codec_dai(be, i, dai) { - if (dai->playback_widget == widget) - return be; - } - } - } else { - - for_each_card_rtds(card, be) { - - if (!be->dai_link->no_pcm) - continue; + if (w == widget) + return be; - dev_dbg(card->dev, "ASoC: try BE %s\n", - be->cpu_dai->capture_widget ? - be->cpu_dai->capture_widget->name : "(not set)"); + for_each_rtd_codec_dai(be, i, dai) { + w = dai_get_widget(dai, stream); - if (be->cpu_dai->capture_widget == widget) + if (w == widget) return be; - - for_each_rtd_codec_dai(be, i, dai) { - if (dai->capture_widget == widget) - return be; - } } } From c2cd821603c216a6a7242b2b4c1a093051e26aaf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:27:48 +0900 Subject: [PATCH 121/415] ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() dpcm_end_walk_at_be() has very duplicate code. dpcm_end_walk_at_be() { ... if (stream == SNDRV_PCM_STREAM_PLAYBACK) { (1) /* code for Playback */ } else { (2) /* code for Capture */ } } The difference between Playback (1) and Capture (2) code is pointer only (= "playback_widget" or "caputre_widget"). OTOH, now we already has dai_get_widget() for it. This means we can merge (1) and (2). This patch do it and remove duplicated code. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87eeutboul.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 37 ++++++++++++++++--------------------- 1 file changed, 16 insertions(+), 21 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b708db9723107c..7d4419ae63f668 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1297,34 +1297,29 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, { struct snd_soc_card *card = widget->dapm->card; struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dapm_widget *w; struct snd_soc_dai *dai; + int stream; int i; - if (dir == SND_SOC_DAPM_DIR_OUT) { - for_each_card_rtds(card, rtd) { - if (!rtd->dai_link->no_pcm) - continue; + /* adjust dir to stream */ + if (dir == SND_SOC_DAPM_DIR_OUT) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + stream = SNDRV_PCM_STREAM_CAPTURE; - if (rtd->cpu_dai->playback_widget == widget) - return true; + for_each_card_rtds(card, rtd) { + if (!rtd->dai_link->no_pcm) + continue; - for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_widget == widget) - return true; - } - } - } else { /* SND_SOC_DAPM_DIR_IN */ - for_each_card_rtds(card, rtd) { - if (!rtd->dai_link->no_pcm) - continue; + w = dai_get_widget(rtd->cpu_dai, stream); + if (w == widget) + return true; - if (rtd->cpu_dai->capture_widget == widget) + for_each_rtd_codec_dai(rtd, i, dai) { + w = dai_get_widget(dai, stream); + if (w == widget) return true; - - for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->capture_widget == widget) - return true; - } } } From 027a483871832044fa0cb8e9df208cca5230ae91 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:27:53 +0900 Subject: [PATCH 122/415] ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() dpcm_end_walk_at_be() and dpcm_get_be() are almost same code. This patch uses dpcm_get_be() from dpcm_end_walk_at_be(). Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87d0adbouh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 20 +++----------------- 1 file changed, 3 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7d4419ae63f668..1d48be24bfaa44 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1297,10 +1297,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, { struct snd_soc_card *card = widget->dapm->card; struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dapm_widget *w; - struct snd_soc_dai *dai; int stream; - int i; /* adjust dir to stream */ if (dir == SND_SOC_DAPM_DIR_OUT) @@ -1308,20 +1305,9 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, else stream = SNDRV_PCM_STREAM_CAPTURE; - for_each_card_rtds(card, rtd) { - if (!rtd->dai_link->no_pcm) - continue; - - w = dai_get_widget(rtd->cpu_dai, stream); - if (w == widget) - return true; - - for_each_rtd_codec_dai(rtd, i, dai) { - w = dai_get_widget(dai, stream); - if (w == widget) - return true; - } - } + rtd = dpcm_get_be(card, widget, stream); + if (rtd) + return true; return false; } From c9645d2a952b7925b6708b24242cd5ed04975648 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:27:57 +0900 Subject: [PATCH 123/415] ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() No one is using soc_dpcm_be_digital_mute(). If it exists only by assumption that "it may be necessary someday", let's remove it now. Otherwise code maintenance will be difficult. We can revive it when we really needed it. Let's remove it, so far. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87blpxbouc.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 1 - sound/soc/soc-pcm.c | 27 --------------------------- 2 files changed, 28 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index b654ebfc87665e..6655163876714a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -141,7 +141,6 @@ void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream, enum snd_soc_dpcm_state state); /* internal use only */ -int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_runtime_update(struct snd_soc_card *); #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1d48be24bfaa44..b8ea4d892031d1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2719,33 +2719,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) mutex_unlock(&card->mutex); return ret; } -int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) -{ - struct snd_soc_dpcm *dpcm; - struct snd_soc_dai *dai; - - for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { - - struct snd_soc_pcm_runtime *be = dpcm->be; - int i; - - if (be->dai_link->ignore_suspend) - continue; - - for_each_rtd_codec_dai(be, i, dai) { - struct snd_soc_dai_driver *drv = dai->driver; - - dev_dbg(be->dev, "ASoC: BE digital mute %s\n", - be->dai_link->name); - - if (drv->ops && drv->ops->digital_mute && - dai->playback_active) - drv->ops->digital_mute(dai, mute); - } - } - - return 0; -} static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) { From 289a7e64f8583aaa45847c7fa3b7fabf8d48fd6b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:04 +0900 Subject: [PATCH 124/415] ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() No one is using snd_soc_dpcm_be_get/set_state(). If it exists only by assumption that "it may be necessary someday", let's remove it now. Otherwise code maintenance will be difficult. We can revive it when we really needed it. Let's remove it, so far. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87a75hbou7.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 8 -------- sound/soc/soc-pcm.c | 16 ---------------- 2 files changed, 24 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 6655163876714a..3e7819d2a6aab4 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -132,14 +132,6 @@ int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, struct snd_pcm_substream * snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream); -/* get the BE runtime state */ -enum snd_soc_dpcm_state - snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream); - -/* set the BE runtime state */ -void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream, - enum snd_soc_dpcm_state state); - /* internal use only */ int soc_dpcm_runtime_update(struct snd_soc_card *); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b8ea4d892031d1..bd4e4f86f5b252 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2954,22 +2954,6 @@ struct snd_pcm_substream * } EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); -/* get the BE runtime state */ -enum snd_soc_dpcm_state - snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream) -{ - return be->dpcm[stream].state; -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state); - -/* set the BE runtime state */ -void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, - int stream, enum snd_soc_dpcm_state state) -{ - be->dpcm[stream].state = state; -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state); - /* * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE * are not running, paused or suspended for the specified stream direction. From 085d22be035db245c44714cf879a73eae06c9f6b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:07 +0900 Subject: [PATCH 125/415] ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code Below functions are doing very similar things, the difference is used state only. snd_soc_dpcm_can_be_free_stop() snd_soc_dpcm_can_be_params() This patch adds common snd_soc_dpcm_check_state(), and use it from snd_soc_dpcm_can_be_free_stop() / snd_soc_dpcm_can_be_params(). It can reduce duplicate code. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/878sl1bou2.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 70 ++++++++++++++++++++++----------------------- 1 file changed, 35 insertions(+), 35 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index bd4e4f86f5b252..d77a2c22a04f12 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2954,17 +2954,17 @@ struct snd_pcm_substream * } EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) +static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, + int stream, + const enum snd_soc_dpcm_state *states, + int num_states) { struct snd_soc_dpcm *dpcm; int state; int ret = 1; unsigned long flags; + int i; spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { @@ -2973,18 +2973,34 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, continue; state = dpcm->fe->dpcm[stream].state; - if (state == SND_SOC_DPCM_STATE_START || - state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) { - ret = 0; - break; + for (i = 0; i < num_states; i++) { + if (state == states[i]) { + ret = 0; + break; + } } } spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); - /* it's safe to free/stop this BE DAI */ + /* it's safe to do this BE DAI */ return ret; } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2994,30 +3010,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { - struct snd_soc_dpcm *dpcm; - int state; - int ret = 1; - unsigned long flags; - - spin_lock_irqsave(&fe->card->dpcm_lock, flags); - for_each_dpcm_fe(be, stream, dpcm) { - - if (dpcm->fe == fe) - continue; + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + SND_SOC_DPCM_STATE_PREPARE, + }; - state = dpcm->fe->dpcm[stream].state; - if (state == SND_SOC_DPCM_STATE_START || - state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) { - ret = 0; - break; - } - } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); - - /* it's safe to change hw_params */ - return ret; + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); From cae06eb92557f0a073835380e57abee5f8173d73 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:11 +0900 Subject: [PATCH 126/415] ASoC: soc-pcm: use goto and remove multi return When we use some kind of lock, we need to do unlock. In that time, multi unlock/return is not good implementation. This patch add label and use goto at dpcm_fe_dai_open() to reduce such code. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/877e0lboty.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d77a2c22a04f12..5a79a830ee18e8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2733,8 +2733,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) ret = dpcm_path_get(fe, stream, &list); if (ret < 0) { - mutex_unlock(&fe->card->mutex); - return ret; + goto open_end; } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); @@ -2755,6 +2754,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); +open_end: mutex_unlock(&fe->card->mutex); return ret; } From 0f6011fd79a2fb92cb80177fd6bdc8aac3a3cd93 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:15 +0900 Subject: [PATCH 127/415] ASoC: soc-pcm: merge playback/cature_active into stream_active DAI has playback_active and capture_active to care usage count. OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. But because of this kind of implementation mismatch, ALSA SoC has many verbose code. To solve this issue, this patch merge playback_active/capture_active into stream_active[2]; Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/875zg5botu.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 +-- sound/soc/codecs/cs4271.c | 4 ++-- sound/soc/dwc/dwc-i2s.c | 4 ++-- sound/soc/soc-core.c | 17 +++++++++-------- sound/soc/soc-pcm.c | 25 ++++++++++++------------- 5 files changed, 26 insertions(+), 27 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 04c23ac0dfffdf..7481e468be3916 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -322,8 +322,7 @@ struct snd_soc_dai { struct snd_soc_dai_driver *driver; /* DAI runtime info */ - unsigned int capture_active; /* stream usage count */ - unsigned int playback_active; /* stream usage count */ + unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */ unsigned int active; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 04b86a51e05546..62f412d6f9f260 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -356,9 +356,9 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, */ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - !dai->capture_active) || + !dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && - !dai->playback_active)) { + !dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])) { ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN, CS4271_MODE2_PDN); diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 7eeca2150b2d62..a8bff6f08a69cb 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -427,9 +427,9 @@ static int dw_i2s_resume(struct snd_soc_component *component) clk_enable(dev->clk); for_each_component_dais(component, dai) { - if (dai->playback_active) + if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]) dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK); - if (dai->capture_active) + if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6a58a8f6e3c4e9..f0ae1a7d7e0911 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -365,19 +365,20 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; + int playback = SNDRV_PCM_STREAM_PLAYBACK; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, - codec_dai->playback_active ? "active" : "inactive", + codec_dai->stream_active[playback] ? "active" : "inactive", rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ if (rtd->pop_wait == 1) { rtd->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + snd_soc_dapm_stream_event(rtd, playback, SND_SOC_DAPM_STREAM_STOP); } @@ -514,6 +515,7 @@ int snd_soc_suspend(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_component *component; struct snd_soc_pcm_runtime *rtd; + int playback = SNDRV_PCM_STREAM_PLAYBACK; int i; /* If the card is not initialized yet there is nothing to do */ @@ -537,9 +539,8 @@ int snd_soc_suspend(struct device *dev) continue; for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_active) - snd_soc_dai_digital_mute(dai, 1, - SNDRV_PCM_STREAM_PLAYBACK); + if (dai->stream_active[playback]) + snd_soc_dai_digital_mute(dai, 1, playback); } } @@ -680,14 +681,14 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for_each_card_rtds(card, rtd) { struct snd_soc_dai *dai; + int playback = SNDRV_PCM_STREAM_PLAYBACK; if (rtd->dai_link->ignore_suspend) continue; for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_active) - snd_soc_dai_digital_mute(dai, 0, - SNDRV_PCM_STREAM_PLAYBACK); + if (dai->stream_active[playback]) + snd_soc_dai_digital_mute(dai, 0, playback); } } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5a79a830ee18e8..6fd69574ca31b8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -100,15 +100,9 @@ static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, lockdep_assert_held(&rtd->card->pcm_mutex); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active += action; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->playback_active += action; - } else { - cpu_dai->capture_active += action; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->capture_active += action; - } + cpu_dai->stream_active[stream] += action; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->stream_active[stream] += action; cpu_dai->active += action; cpu_dai->component->active += action; @@ -967,8 +961,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* apply codec digital mute */ for_each_rtd_codec_dai(rtd, i, codec_dai) { - if ((playback && codec_dai->playback_active == 1) || - (!playback && codec_dai->capture_active == 1)) + int playback_active = codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]; + int capture_active = codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]; + + if ((playback && playback_active == 1) || + (!playback && capture_active == 1)) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); } @@ -2634,7 +2631,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) goto capture; /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] || + !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]) goto capture; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); @@ -2665,7 +2663,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) return 0; /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] || + !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) return 0; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); From 3193abd26b515ccac65e1c323533cb7f53d06176 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:19 +0900 Subject: [PATCH 128/415] ALSA: pcm.h: add for_each_pcm_streams() ALSA code has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere. Having for_each_xxxx macro is useful. This patch adds for_each_pcm_streams() for it. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Acked-by: Takashi Iwai Link: https://lore.kernel.org/r/874kvpbotq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/pcm.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index f657ff08f317d0..2628246b76fa76 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -644,6 +644,11 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, #define snd_pcm_group_for_each_entry(s, substream) \ list_for_each_entry(s, &substream->group->substreams, link_list) +#define for_each_pcm_streams(stream) \ + for (stream = SNDRV_PCM_STREAM_PLAYBACK; \ + stream <= SNDRV_PCM_STREAM_LAST; \ + stream++) + /** * snd_pcm_running - Check whether the substream is in a running state * @substream: substream to check From d74c2a156b710e9ad81193a60e037430f8894c0c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:25 +0900 Subject: [PATCH 129/415] ASoC: soc-core: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/8736b9botk.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 31 +++++++++++++------------------ 1 file changed, 13 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f0ae1a7d7e0911..30c17fde14caf5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -432,6 +432,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_component *component; struct device *dev; int ret; + int stream; /* * for rtd->dev @@ -466,10 +467,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->dev = dev; INIT_LIST_HEAD(&rtd->list); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); + for_each_pcm_streams(stream) { + INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients); + } dev_set_drvdata(dev, rtd); INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); @@ -559,17 +560,14 @@ int snd_soc_suspend(struct device *dev) snd_soc_flush_all_delayed_work(card); for_each_card_rtds(card, rtd) { + int stream; if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_SUSPEND); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_SUSPEND); + for_each_pcm_streams(stream) + snd_soc_dapm_stream_event(rtd, stream, + SND_SOC_DAPM_STREAM_SUSPEND); } /* Recheck all endpoints too, their state is affected by suspend */ @@ -665,17 +663,14 @@ static void soc_resume_deferred(struct work_struct *work) } for_each_card_rtds(card, rtd) { + int stream; if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_RESUME); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_RESUME); + for_each_pcm_streams(stream) + snd_soc_dapm_stream_event(rtd, stream, + SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */ From 7083f877ea66e106f90e9a1a0dabb19ebbacc4e6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:28 +0900 Subject: [PATCH 130/415] ASoC: soc-pcm: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/871rqtboth.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 97 ++++++++++++++++----------------------------- 1 file changed, 35 insertions(+), 62 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6fd69574ca31b8..63f67eb7c077ca 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2612,6 +2612,7 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { struct snd_soc_dapm_widget_list *list; + int stream; int count, paths; if (!fe->dai_link->dynamic) @@ -2625,69 +2626,42 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", new ? "new" : "old", fe->dai_link->name); - /* skip if FE doesn't have playback capability */ - if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK) || - !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK)) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] || - !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - return paths; - } - - /* update any playback paths */ - count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); - if (count) { - if (new) - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - else - dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + for_each_pcm_streams(stream) { - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - - dpcm_path_put(&list); + /* skip if FE doesn't have playback/capture capability */ + if (!snd_soc_dai_stream_valid(fe->cpu_dai, stream) || + !snd_soc_dai_stream_valid(fe->codec_dai, stream)) + continue; -capture: - /* skip if FE doesn't have capture capability */ - if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE) || - !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE)) - return 0; + /* skip if FE isn't currently playing/capturing */ + if (!fe->cpu_dai->stream_active[stream] || + !fe->codec_dai->stream_active[stream]) + continue; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] || - !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) - return 0; + paths = dpcm_path_get(fe, stream, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, + stream == SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"); + return paths; + } - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - return paths; - } + /* update any playback/capture paths */ + count = dpcm_process_paths(fe, stream, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, stream); + else + dpcm_run_old_update(fe, stream); - /* update any old capture paths */ - count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); - if (count) { - if (new) - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - else - dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_clear_pending_state(fe, stream); + dpcm_be_disconnect(fe, stream); + } - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_path_put(&list); } - dpcm_path_put(&list); - return 0; } @@ -3114,19 +3088,18 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, { struct snd_soc_pcm_runtime *fe = file->private_data; ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0; + int stream; char *buf; buf = kmalloc(out_count, GFP_KERNEL); if (!buf) return -ENOMEM; - if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) - offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK, - buf + offset, out_count - offset); - - if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) - offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, - buf + offset, out_count - offset); + for_each_pcm_streams(stream) + if (snd_soc_dai_stream_valid(fe->cpu_dai, stream)) + offset += dpcm_show_state(fe, stream, + buf + offset, + out_count - offset); ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); From ee10fbe1cdf7cb4ae62f5e23ccd771e696b8f404 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:32 +0900 Subject: [PATCH 131/415] ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87zhdhaa8x.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 2cc25651661caa..d6b4831e8aec9e 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -237,7 +237,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, max_buffer_size = SIZE_MAX; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + for_each_pcm_streams(i) { substream = rtd->pcm->streams[i].substream; if (!substream) continue; @@ -371,8 +371,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, dev = config->dma_dev; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; - i++) { + for_each_pcm_streams(i) { if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) name = "rx-tx"; else @@ -401,8 +400,7 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) { unsigned int i; - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; - i++) { + for_each_pcm_streams(i) { if (!pcm->chan[i]) continue; dma_release_channel(pcm->chan[i]); From 0a170be9631ea8335e494f3c5f7ab720287023a2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:36 +0900 Subject: [PATCH 132/415] ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87y2t1aa8t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/dwc/dwc-i2s.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index a8bff6f08a69cb..515f88456dbd81 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -422,15 +422,15 @@ static int dw_i2s_resume(struct snd_soc_component *component) { struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component); struct snd_soc_dai *dai; + int stream; if (dev->capability & DW_I2S_MASTER) clk_enable(dev->clk); for_each_component_dais(component, dai) { - if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]) - dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK); - if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) - dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE); + for_each_pcm_streams(stream) + if (dai->stream_active[stream]) + dw_i2s_config(dev, stream); } return 0; From fa7b2a1fcb92906a284b0824b45866a7b8afb599 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:40 +0900 Subject: [PATCH 133/415] ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87wo8laa8p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index ece130f59d15c9..44e5924be87056 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component, return ret; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (!substream) continue; @@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component, struct snd_pcm_substream *substream; int i; - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (!substream) continue; From 4c260c3f19bd16e6b11841aad1162f5a105ed24e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:44 +0900 Subject: [PATCH 134/415] ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87v9o5aa8m.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index b05091c283b725..5d1bc57571698c 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -529,7 +529,7 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component, struct snd_pcm_substream *substream; int i; - for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (substream) { snd_dma_free_pages(&substream->dma_buffer); From 525c4107da8c0a86aa3548dc6e1d0014749e95f7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:47 +0900 Subject: [PATCH 135/415] ASoC: sof: sof-audio: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87tv3paa8i.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 75f2ef2bd94be0..fc4ed2a8a914da 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -23,7 +23,7 @@ bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev) int dir; list_for_each_entry(spcm, &sdev->pcm_list, list) { - for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + for_each_pcm_streams(dir) { substream = spcm->stream[dir].substream; if (!substream || !substream->runtime) continue; @@ -71,7 +71,7 @@ int sof_set_hw_params_upon_resume(struct device *dev) * have been suspended. */ list_for_each_entry(spcm, &sdev->pcm_list, list) { - for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + for_each_pcm_streams(dir) { /* * do not reset hw_params upon resume for streams that * were kept running during suspend @@ -319,16 +319,11 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp, int dir; list_for_each_entry(spcm, &sdev->pcm_list, list) { - dir = SNDRV_PCM_STREAM_PLAYBACK; - if (spcm->stream[dir].comp_id == comp_id) { - *direction = dir; - return spcm; - } - - dir = SNDRV_PCM_STREAM_CAPTURE; - if (spcm->stream[dir].comp_id == comp_id) { - *direction = dir; - return spcm; + for_each_pcm_streams(dir) { + if (spcm->stream[dir].comp_id == comp_id) { + *direction = dir; + return spcm; + } } } From ffd11d1e7ad4602d5d1c2b4517ad316f7587d4d9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Feb 2020 17:28:51 +0900 Subject: [PATCH 136/415] ALSA: usx2y: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Takashi Iwai Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87sgj9aa8e.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/usb/usx2y/usbusx2yaudio.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 772f6f3ccbb11c..37d290fe9d4350 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -906,11 +906,12 @@ static const struct snd_pcm_ops snd_usX2Y_pcm_ops = */ static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream) { - kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]); - usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL; + int stream; - kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]); - usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL; + for_each_pcm_streams(stream) { + kfree(usX2Y_substream[stream]); + usX2Y_substream[stream] = NULL; + } } static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm) From eaa2330bfcbf1d600776e219c5d2080f36a3c59c Mon Sep 17 00:00:00 2001 From: Jeff Chang Date: Wed, 19 Feb 2020 17:04:24 +0800 Subject: [PATCH 137/415] ASoC: MT6660 update to 1.0.8_G 1. add mt6660_component_settign for Component INIT Setting Signed-off-by: Jeff Chang Link: https://lore.kernel.org/r/1582103064-25088-1-git-send-email-richtek.jeff.chang@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/mt6660.c | 78 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 75 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index 1a3515df17641e..bcec82aa57fb9d 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -8,7 +8,6 @@ #include #include #include -#include #include #include #include @@ -224,14 +223,87 @@ static int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) 0x01, on_off ? 0x00 : 0x01); } +struct reg_table { + uint32_t addr; + uint32_t mask; + uint32_t val; +}; + +static const struct reg_table mt6660_setting_table[] = { + { 0x20, 0x80, 0x00 }, + { 0x30, 0x01, 0x00 }, + { 0x50, 0x1c, 0x04 }, + { 0xB1, 0x0c, 0x00 }, + { 0xD3, 0x03, 0x03 }, + { 0xE0, 0x01, 0x00 }, + { 0x98, 0x44, 0x04 }, + { 0xB9, 0xff, 0x82 }, + { 0xB7, 0x7777, 0x7273 }, + { 0xB6, 0x07, 0x03 }, + { 0x6B, 0xe0, 0x20 }, + { 0x07, 0xff, 0x70 }, + { 0xBB, 0xff, 0x20 }, + { 0x69, 0xff, 0x40 }, + { 0xBD, 0xffff, 0x17f8 }, + { 0x70, 0xff, 0x15 }, + { 0x7C, 0xff, 0x00 }, + { 0x46, 0xff, 0x1d }, + { 0x1A, 0xffffffff, 0x7fdb7ffe }, + { 0x1B, 0xffffffff, 0x7fdb7ffe }, + { 0x51, 0xff, 0x58 }, + { 0xA2, 0xff, 0xce }, + { 0x33, 0xffff, 0x7fff }, + { 0x4C, 0xffff, 0x0116 }, + { 0x16, 0x1800, 0x0800 }, + { 0x68, 0x1f, 0x07 }, +}; + +static int mt6660_component_setting(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + size_t i = 0; + + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(component->dev, "%s chip power on failed\n", __func__); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_setting_table); i++) { + ret = snd_soc_component_update_bits(component, + mt6660_setting_table[i].addr, + mt6660_setting_table[i].mask, + mt6660_setting_table[i].val); + if (ret < 0) { + dev_err(component->dev, "%s update 0x%02x failed\n", + __func__, mt6660_setting_table[i].addr); + return ret; + } + } + + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(component->dev, "%s chip power off failed\n", __func__); + return ret; + } + + return 0; +} + static int mt6660_component_probe(struct snd_soc_component *component) { struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret; dev_dbg(component->dev, "%s\n", __func__); snd_soc_component_init_regmap(component, chip->regmap); - return 0; + ret = mt6660_component_setting(component); + if (ret < 0) + dev_err(chip->dev, "mt6660 component setting failed\n"); + + return ret; } static void mt6660_component_remove(struct snd_soc_component *component) @@ -505,4 +577,4 @@ module_i2c_driver(mt6660_i2c_driver); MODULE_AUTHOR("Jeff Chang "); MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); MODULE_LICENSE("GPL"); -MODULE_VERSION("1.0.7_G"); +MODULE_VERSION("1.0.8_G"); From 6b62fa95b56bcc77cbbcc76e45f5170b4ec229b1 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 19 Feb 2020 11:25:26 +0100 Subject: [PATCH 138/415] ASoC: fix card registration regression. This reverts commit b2354e4009a773c00054b964d937e1b81cb92078. This change might have been desirable to ensure the uniqueness of the component name. It would have helped to better support linux devices which register multiple components, something is which more common than initially thought. However, some card driver are directly using dev_name() to fill the component names of the dai_link which is a problem if want to change the way ASoC generates the component names. Until we figure out the appropriate way to deal with this, revert the change and keep the names as they were. There might be a couple of warning related to debugfs (which were already present before the change) but it is still better than breaking working audio cards. Signed-off-by: Jerome Brunet Tested-by: Marek Szyprowski Cc: Marek Szyprowski Link: https://lore.kernel.org/r/20200219102526.692126-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 29 +---------------------------- 1 file changed, 1 insertion(+), 28 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30c17fde14caf5..518b652cf872d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2442,33 +2442,6 @@ static int snd_soc_register_dais(struct snd_soc_component *component, return ret; } -static char *snd_soc_component_unique_name(struct device *dev, - struct snd_soc_component *component) -{ - struct snd_soc_component *pos; - int count = 0; - char *name, *unique; - - name = fmt_single_name(dev, &component->id); - if (!name) - return name; - - /* Count the number of components registred by the device */ - for_each_component(pos) { - if (dev == pos->dev) - count++; - } - - /* Keep naming as it is for the 1st component */ - if (!count) - return name; - - unique = devm_kasprintf(dev, GFP_KERNEL, "%s-%d", name, count); - devm_kfree(dev, name); - - return unique; -} - static int snd_soc_component_initialize(struct snd_soc_component *component, const struct snd_soc_component_driver *driver, struct device *dev) { @@ -2477,7 +2450,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->card_list); mutex_init(&component->io_mutex); - component->name = snd_soc_component_unique_name(dev, component); + component->name = fmt_single_name(dev, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; From ec06dc15c358d3f41e9fd05872d772ed0f9fa32a Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Wed, 19 Feb 2020 17:38:38 +0800 Subject: [PATCH 139/415] ASoC: dapm: select sleep_state when initializing PINCTRL widget Selects sleep_state when initializing PINCTRL widget. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200219170951.1.I61f6559a37a6a40a6fde0737cb16100fb17c0480@changeid Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cc17a3730d3d2a..69eff234b26fd2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3628,6 +3628,9 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, ret = PTR_ERR(w->pinctrl); goto request_failed; } + + /* set to sleep_state when initializing */ + dapm_pinctrl_event(w, NULL, SND_SOC_DAPM_POST_PMD); break; case snd_soc_dapm_clock_supply: w->clk = devm_clk_get(dapm->dev, w->name); From c77b8317ee3ab43634421afb73fdb1ea253d3d47 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Wed, 19 Feb 2020 17:38:39 +0800 Subject: [PATCH 140/415] ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL in TDM out Uses SND_SOC_DAPM_PINCTRL in TDM out to simplify code. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200219170951.2.I7ed16ef57d9e0bcafc37e766142f68cbad5b54c6@changeid Signed-off-by: Mark Brown --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 98 ++----------------- 1 file changed, 10 insertions(+), 88 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c0c85972cfb749..03d104fbe1855c 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -16,20 +16,7 @@ #include "../../codecs/da7219-aad.h" #include "../../codecs/da7219.h" -enum PINCTRL_PIN_STATE { - PIN_STATE_DEFAULT = 0, - PIN_TDM_OUT_ON, - PIN_TDM_OUT_OFF, - PIN_STATE_MAX -}; - -static const char * const mt8183_pin_str[PIN_STATE_MAX] = { - "default", "aud_tdm_out_on", "aud_tdm_out_off", -}; - struct mt8183_da7219_max98357_priv { - struct pinctrl *pinctrl; - struct pinctrl_state *pin_states[PIN_STATE_MAX]; struct snd_soc_jack headset_jack; }; @@ -259,47 +246,6 @@ SND_SOC_DAILINK_DEFS(tdm, DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); -static int mt8183_da7219_tdm_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mt8183_da7219_max98357_priv *priv = - snd_soc_card_get_drvdata(rtd->card); - int ret; - - if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON])) - return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]); - - ret = pinctrl_select_state(priv->pinctrl, - priv->pin_states[PIN_TDM_OUT_ON]); - if (ret) - dev_err(rtd->card->dev, "%s failed to select state %d\n", - __func__, ret); - - return ret; -} - -static void mt8183_da7219_tdm_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mt8183_da7219_max98357_priv *priv = - snd_soc_card_get_drvdata(rtd->card); - int ret; - - if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) - return; - - ret = pinctrl_select_state(priv->pinctrl, - priv->pin_states[PIN_TDM_OUT_OFF]); - if (ret) - dev_err(rtd->card->dev, "%s failed to select state %d\n", - __func__, ret); -} - -static struct snd_soc_ops mt8183_da7219_tdm_ops = { - .startup = mt8183_da7219_tdm_startup, - .shutdown = mt8183_da7219_tdm_shutdown, -}; - static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { /* FE */ { @@ -455,7 +401,6 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { .dpcm_playback = 1, .ignore_suspend = 1, .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, - .ops = &mt8183_da7219_tdm_ops, SND_SOC_DAILINK_REG(tdm), }, }; @@ -482,10 +427,13 @@ static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = { static const struct snd_soc_dapm_widget mt8183_da7219_max98357_dapm_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL", + "aud_tdm_out_on", "aud_tdm_out_off"), }; static const struct snd_soc_dapm_route mt8183_da7219_max98357_dapm_routes[] = { {"Speakers", NULL, "Speaker"}, + {"I2S Playback", NULL, "TDM_OUT_PINCTRL"}, }; static struct snd_soc_card mt8183_da7219_max98357_card = { @@ -534,6 +482,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct device_node *platform_node; struct snd_soc_dai_link *dai_link; struct mt8183_da7219_max98357_priv *priv; + struct pinctrl *pinctrl; int ret, i; card->dev = &pdev->dev; @@ -566,39 +515,12 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); - priv->pinctrl = devm_pinctrl_get(&pdev->dev); - if (IS_ERR(priv->pinctrl)) { - dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n", - __func__); - return PTR_ERR(priv->pinctrl); - } - - for (i = 0; i < PIN_STATE_MAX; i++) { - priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, - mt8183_pin_str[i]); - if (IS_ERR(priv->pin_states[i])) { - ret = PTR_ERR(priv->pin_states[i]); - dev_info(&pdev->dev, "%s Can't find pin state %s %d\n", - __func__, mt8183_pin_str[i], ret); - } - } - - if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) { - ret = pinctrl_select_state(priv->pinctrl, - priv->pin_states[PIN_TDM_OUT_OFF]); - if (ret) - dev_info(&pdev->dev, - "%s failed to select state %d\n", - __func__, ret); - } - - if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) { - ret = pinctrl_select_state(priv->pinctrl, - priv->pin_states[PIN_STATE_DEFAULT]); - if (ret) - dev_info(&pdev->dev, - "%s failed to select state %d\n", - __func__, ret); + pinctrl = devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); + if (IS_ERR(pinctrl)) { + ret = PTR_ERR(pinctrl); + dev_err(&pdev->dev, "%s failed to select default state %d\n", + __func__, ret); + return ret; } return devm_snd_soc_register_card(&pdev->dev, card); From 69e53129d01317d94e8b97ec11688880106a2f97 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Wed, 19 Feb 2020 07:46:22 -0600 Subject: [PATCH 141/415] ASoC: tas2562: Add support for ISENSE and VSENSE Add additional support for ISENSE and VSENSE feature for the TAS2562. This feature monitors the output to the loud speaker attempts to eliminate IR drop errors due to packaging. This feature is defined in Section 8.4.5 IV Sense of the data sheet. Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200219134622.22066-1-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2562.c | 32 +++++++++++++++++++++++++++----- sound/soc/codecs/tas2562.h | 6 +++--- 2 files changed, 30 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 729acd874c485c..b517ada7e8092a 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -382,18 +382,34 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; switch (event) { case SND_SOC_DAPM_POST_PMU: - dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_MUTE); + if (ret) + goto end; break; case SND_SOC_DAPM_PRE_PMD: - dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_SHUTDOWN); + if (ret) + goto end; break; default: - break; + dev_err(tas2562->dev, "Not supported evevt\n"); + return -EINVAL; } +end: + if (ret < 0) + return ret; + return 0; } @@ -415,7 +431,6 @@ static const struct snd_kcontrol_new tas2562_snd_controls[] = { static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux), - SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch), @@ -430,7 +445,7 @@ static const struct snd_soc_dapm_route tas2562_audio_map[] = { {"ASI1 Sel", "Left", "ASI1"}, {"ASI1 Sel", "Right", "ASI1"}, {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, - { "DAC", NULL, "DAC IN" }, + { "DAC", NULL, "ASI1 Sel" }, { "OUT", NULL, "DAC" }, {"ISENSE", "Switch", "IMON"}, {"VSENSE", "Switch", "VMON"}, @@ -471,6 +486,13 @@ static struct snd_soc_dai_driver tas2562_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = TAS2562_FORMATS, }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2562_FORMATS, + }, .ops = &tas2562_speaker_dai_ops, }, }; diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 62e659ab786d9f..6f55ebcf19ea7c 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -40,7 +40,7 @@ #define TAS2562_RESET BIT(0) -#define TAS2562_MODE_MASK 0x3 +#define TAS2562_MODE_MASK GENMASK(1,0) #define TAS2562_ACTIVE 0x0 #define TAS2562_MUTE 0x1 #define TAS2562_SHUTDOWN 0x2 @@ -73,8 +73,8 @@ #define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3) #define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3)) -#define TAS2562_VSENSE_POWER_EN BIT(2) -#define TAS2562_ISENSE_POWER_EN BIT(3) +#define TAS2562_VSENSE_POWER_EN 2 +#define TAS2562_ISENSE_POWER_EN 3 #define TAS2562_TDM_CFG5_VSNS_EN BIT(6) #define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0) From ce83baca8526689c332135c0abca2667ba056009 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:55:53 +0900 Subject: [PATCH 142/415] ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87o8tvjcbc.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- drivers/soundwire/qcom.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 1c6c6a2e0def55..fb30bbec999ad8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -594,6 +594,7 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev); struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sdw_stream_runtime *sruntime; + struct snd_soc_dai *codec_dai; int ret, i; sruntime = sdw_alloc_stream(dai->name); @@ -602,12 +603,12 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, ctrl->sruntime[dai->id] = sruntime; - for (i = 0; i < rtd->num_codecs; i++) { - ret = snd_soc_dai_set_sdw_stream(rtd->codec_dais[i], sruntime, + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_set_sdw_stream(codec_dai, sruntime, substream->stream); if (ret < 0 && ret != -ENOTSUPP) { dev_err(dai->dev, "Failed to set sdw stream on %s", - rtd->codec_dais[i]->name); + codec_dai->name); sdw_release_stream(sruntime); return ret; } From a4eb41eef331d31b8593defa10b249e155e0314f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:01 +0900 Subject: [PATCH 143/415] ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87mu9fjcb4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 3b5547a27aad9f..5a23597261acfc 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -43,14 +43,14 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0, i; - for (i = 0 ; i < dai_link->num_codecs; i++) { - ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i], + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); if (ret != 0 && ret != -ENOTSUPP) { @@ -77,6 +77,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int ret = 0, j; int channels, slot_width; @@ -125,8 +126,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, } } - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { ret = snd_soc_dai_set_tdm_slot( @@ -214,7 +214,6 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); struct snd_jack *jack; - struct snd_soc_dai_link *dai_link = rtd->dai_link; /* * Codec SLIMBUS configuration * RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13 @@ -266,8 +265,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) } break; case SLIMBUS_0_RX...SLIMBUS_6_TX: - for (i = 0 ; i < dai_link->num_codecs; i++) { - rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i], + for_each_rtd_codec_dai(rtd, i, codec_dai) { + rval = snd_soc_dai_set_channel_map(codec_dai, ARRAY_SIZE(tx_ch), tx_ch, ARRAY_SIZE(rx_ch), @@ -275,7 +274,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) if (rval != 0 && rval != -ENOTSUPP) return rval; - snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0, + snd_soc_dai_set_sysclk(codec_dai, 0, WCD934X_DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); } @@ -345,8 +344,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B; - for (j = 0; j < rtd->num_codecs; j++) { - codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { From cf4dae032096f1299cf390fd55da489cb445dcbb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:09 +0900 Subject: [PATCH 144/415] ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87lfozjcaw.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/qcom/apq8016_sbc.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index ac75838bbfabe9..2d064f3bc9b6d0 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -34,8 +34,8 @@ struct apq8016_sbc_data { static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; struct snd_soc_component *component; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card); int i, rval; @@ -90,10 +90,9 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for (i = 0 ; i < dai_link->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { - component = dai->component; + component = codec_dai->component; /* Set default mclk for internal codec */ rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE, SND_SOC_CLOCK_IN); From 225c53a8cfb6fdd8defbbf72e8dcfb3801f7f51e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:15 +0900 Subject: [PATCH 145/415] ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87k14jjcaq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index dd80d0186a6c3c..02aa18d24319cf 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -164,8 +164,7 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params); - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* 100 Fs to drive 24 bit data */ ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, From 89a2870f6be6aa75de2df22f8baa982c2d7d86e8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:20 +0900 Subject: [PATCH 146/415] ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87imk3jcal.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 7a13e9b3518715..88f69e3697d21a 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -176,10 +176,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *runtime = substream->private_data; + struct snd_soc_dai *codec_dai; int ret, j; - for (j = 0; j < runtime->num_codecs; j++) { - struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + for_each_rtd_codec_dai(runtime, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); @@ -221,10 +221,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int j, ret; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { const char *name = codec_dai->component->name; struct snd_soc_component *component = codec_dai->component; struct snd_soc_dapm_context *dapm = From 56f1003f65830697bd68ae1e26b5561e4cbe6523 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:25 +0900 Subject: [PATCH 147/415] ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87h7znjcag.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 03d104fbe1855c..4a5ef07e956b55 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -40,6 +40,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; unsigned int rate = params_rate(params); unsigned int mclk_fs_ratio = 256; unsigned int mclk_fs = rate * mclk_fs_ratio; @@ -51,8 +52,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) dev_err(rtd->dev, "failed to set cpu dai sysclk\n"); - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_sysclk(codec_dai, @@ -82,10 +82,10 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int ret = 0, j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_pll(codec_dai, From e14980976534d9d94f5cddd70033707965482ede Mon Sep 17 00:00:00 2001 From: Robin Murphy Date: Tue, 18 Feb 2020 21:31:58 +0000 Subject: [PATCH 148/415] ASoC: dt-bindings: Make RK3328 codec GPIO explicit Existing RK3328 codec drivers have overloaded the GRF phandle to assume implicit control of the limited-function GPIO_MUTE pin, which is usually used to enable an external audio line driver IC. Since this pin has a proper binding of its own (see gpio/rockchip,rk3328-grf-gpio.txt), make a GPIO explicit in the codec binding too. This will help avoid ambiguity on boards that use that pin for some other purpose. (and while touching the example, enforce the "don't include status" rule) Signed-off-by: Robin Murphy Link: https://lore.kernel.org/r/5f7a399dea8a9dedef57f6f99f0f6ab1c1fdc56a.1581376744.git.robin.murphy@arm.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/rockchip,rk3328-codec.txt | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt index 2469588c7ccbb7..1ecd75d2032a00 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt @@ -10,6 +10,11 @@ Required properties: - clock-names: should be "pclk". - spk-depop-time-ms: speak depop time msec. +Optional properties: + +- mute-gpios: GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + Example for rk3328 internal codec: codec: codec@ff410000 { @@ -18,6 +23,6 @@ codec: codec@ff410000 { rockchip,grf = <&grf>; clocks = <&cru PCLK_ACODEC>; clock-names = "pclk"; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; spk-depop-time-ms = 100; - status = "disabled"; }; From 87d12d5545fa72d67d99d797cdb464c0c7efb9c9 Mon Sep 17 00:00:00 2001 From: Robin Murphy Date: Tue, 18 Feb 2020 21:31:59 +0000 Subject: [PATCH 149/415] ASoC: rockchip: Make RK3328 GPIO_MUTE control explicit The RK3328 reference design uses an external line driver IC as a buffer on the analog codec output, enabled by the GPIO_MUTE pin, and such a configuration is currently assumed in the codec driver's direct poking of GRF_SOC_CON10 to control the GPIO_MUTE output value. However, some boards wire up analog audio yet use that pin for some other purpose, so that assumption doesn't always hold. Update this functionality to rely on an explicit GPIO descriptor, such that it can be managed at the board level. Signed-off-by: Robin Murphy Link: https://lore.kernel.org/r/5bc383ed1832f0f5d1dcb3c97ad92fd68e5217e3.1581376744.git.robin.murphy@arm.com Signed-off-by: Mark Brown --- sound/soc/codecs/rk3328_codec.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 287c962ba00d71..115706a55577d1 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -7,6 +7,7 @@ #include #include #include +#include #include #include #include @@ -31,7 +32,7 @@ struct rk3328_codec_priv { struct regmap *regmap; - struct regmap *grf; + struct gpio_desc *mute; struct clk *mclk; struct clk *pclk; unsigned int sclk; @@ -106,16 +107,6 @@ static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static void rk3328_analog_output(struct rk3328_codec_priv *rk3328, int mute) -{ - unsigned int val = BIT(17); - - if (mute) - val |= BIT(1); - - regmap_write(rk3328->grf, RK3328_GRF_SOC_CON10, val); -} - static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute) { struct rk3328_codec_priv *rk3328 = @@ -205,7 +196,7 @@ static int rk3328_codec_open_playback(struct rk3328_codec_priv *rk3328) } msleep(rk3328->spk_depop_time); - rk3328_analog_output(rk3328, 1); + gpiod_set_value(rk3328->mute, 0); regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, HPOUTL_GAIN_MASK, OUT_VOLUME); @@ -246,7 +237,7 @@ static int rk3328_codec_close_playback(struct rk3328_codec_priv *rk3328) { size_t i; - rk3328_analog_output(rk3328, 0); + gpiod_set_value(rk3328->mute, 1); regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, HPOUTL_GAIN_MASK, 0); @@ -446,7 +437,6 @@ static int rk3328_platform_probe(struct platform_device *pdev) dev_err(&pdev->dev, "missing 'rockchip,grf'\n"); return PTR_ERR(grf); } - rk3328->grf = grf; /* enable i2s_acodec_en */ regmap_write(grf, RK3328_GRF_SOC_CON2, (BIT(14) << 16 | BIT(14))); @@ -458,7 +448,18 @@ static int rk3328_platform_probe(struct platform_device *pdev) rk3328->spk_depop_time = 200; } - rk3328_analog_output(rk3328, 0); + rk3328->mute = gpiod_get_optional(&pdev->dev, "mute", GPIOD_OUT_HIGH); + if (IS_ERR(rk3328->mute)) + return PTR_ERR(rk3328->mute); + /* + * Rock64 is the only supported platform to have widely relied on + * this; if we do happen to come across an old DTB, just leave the + * external mute forced off. + */ + if (!rk3328->mute && of_machine_is_compatible("pine64,rock64")) { + dev_warn(&pdev->dev, "assuming implicit control of GPIO_MUTE; update devicetree if possible\n"); + regmap_write(grf, RK3328_GRF_SOC_CON10, BIT(17) | BIT(1)); + } rk3328->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(rk3328->mclk)) From 5c36abcd2621adc3d50d05628f0ef0be6e7840a9 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 19 Feb 2020 18:35:02 +0100 Subject: [PATCH 150/415] ASoC: meson: add t9015 internal codec binding documentation Add the DT binding documention of the internal DAC found in the Amlogic gxl, g12a and sm1 SoC family. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200219173503.1112561-2-jbrunet@baylibre.com Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,t9015.yaml | 58 +++++++++++++++++++ 1 file changed, 58 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml diff --git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml new file mode 100644 index 00000000000000..b7c38c2b5b54d6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic T9015 Internal Audio DAC + +maintainers: + - Jerome Brunet + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 0 + + compatible: + items: + - const: amlogic,t9015 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: pclk + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - resets + +examples: + - | + #include + #include + + acodec: audio-controller@32000 { + compatible = "amlogic,t9015"; + reg = <0x0 0x32000 0x0 0x14>; + #sound-dai-cells = <0>; + clocks = <&clkc CLKID_AUDIO_CODEC>; + clock-names = "pclk"; + resets = <&reset RESET_AUDIO_CODEC>; + }; + From 33901f5b9b16d212ee58865e9e8e80fc813f12da Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 19 Feb 2020 18:35:03 +0100 Subject: [PATCH 151/415] ASoC: meson: add t9015 internal DAC driver Add the codec driver of the internal DAC found on Amlogic gxl, g12a and sm1 family. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200219173503.1112561-3-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/t9015.c | 333 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 343 insertions(+) create mode 100644 sound/soc/meson/t9015.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 22d2af75b59e37..897a706dcda01e 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -6,6 +6,7 @@ config SND_MESON_AIU tristate "Amlogic AIU" select SND_MESON_CODEC_GLUE select SND_PCM_IEC958 + imply SND_SOC_MESON_T9015 imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI help Select Y or M to add support for the Audio output subsystem found @@ -116,4 +117,11 @@ config SND_MESON_G12A_TOHDMITX help Select Y or M to add support for HDMI audio on the g12a SoC family + +config SND_SOC_MESON_T9015 + tristate "Amlogic T9015 DAC" + select REGMAP_MMIO + help + Say Y or M if you want to add support for the internal DAC found + on GXL, G12 and SM1 SoC family. endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index f9c90c39149819..3c9d4884681638 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -23,6 +23,7 @@ snd-soc-meson-card-utils-objs := meson-card-utils.o snd-soc-meson-codec-glue-objs := meson-codec-glue.o snd-soc-meson-gx-sound-card-objs := gx-card.o snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o +snd-soc-meson-t9015-objs := t9015.o obj-$(CONFIG_SND_MESON_AIU) += snd-soc-meson-aiu.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o @@ -40,3 +41,4 @@ obj-$(CONFIG_SND_MESON_CARD_UTILS) += snd-soc-meson-card-utils.o obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o obj-$(CONFIG_SND_MESON_GX_SOUND_CARD) += snd-soc-meson-gx-sound-card.o obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o +obj-$(CONFIG_SND_SOC_MESON_T9015) += snd-soc-meson-t9015.o diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c new file mode 100644 index 00000000000000..56d2592c16d53f --- /dev/null +++ b/sound/soc/meson/t9015.c @@ -0,0 +1,333 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include +#include +#include + +#define BLOCK_EN 0x00 +#define LORN_EN 0 +#define LORP_EN 1 +#define LOLN_EN 2 +#define LOLP_EN 3 +#define DACR_EN 4 +#define DACL_EN 5 +#define DACR_INV 20 +#define DACL_INV 21 +#define DACR_SRC 22 +#define DACL_SRC 23 +#define REFP_BUF_EN BIT(12) +#define BIAS_CURRENT_EN BIT(13) +#define VMID_GEN_FAST BIT(14) +#define VMID_GEN_EN BIT(15) +#define I2S_MODE BIT(30) +#define VOL_CTRL0 0x04 +#define GAIN_H 31 +#define GAIN_L 23 +#define VOL_CTRL1 0x08 +#define DAC_MONO 8 +#define RAMP_RATE 10 +#define VC_RAMP_MODE 12 +#define MUTE_MODE 13 +#define UNMUTE_MODE 14 +#define DAC_SOFT_MUTE 15 +#define DACR_VC 16 +#define DACL_VC 24 +#define LINEOUT_CFG 0x0c +#define LORN_POL 0 +#define LORP_POL 4 +#define LOLN_POL 8 +#define LOLP_POL 12 +#define POWER_CFG 0x10 + +struct t9015 { + struct clk *pclk; + struct regulator *avdd; +}; + +static int t9015_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + unsigned int val; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + val = I2S_MODE; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + val = 0; + break; + + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, BLOCK_EN, I2S_MODE, val); + + if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) && + ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_LEFT_J)) + return -EINVAL; + + return 0; +} + +static const struct snd_soc_dai_ops t9015_dai_ops = { + .set_fmt = t9015_dai_set_fmt, +}; + +static struct snd_soc_dai_driver t9015_dai = { + .name = "t9015-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &t9015_dai_ops, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(dac_vol_tlv, -9525, 0); + +static const char * const ramp_rate_txt[] = { "Fast", "Slow" }; +static SOC_ENUM_SINGLE_DECL(ramp_rate_enum, VOL_CTRL1, RAMP_RATE, + ramp_rate_txt); + +static const char * const dacr_in_txt[] = { "Right", "Left" }; +static SOC_ENUM_SINGLE_DECL(dacr_in_enum, BLOCK_EN, DACR_SRC, dacr_in_txt); + +static const char * const dacl_in_txt[] = { "Left", "Right" }; +static SOC_ENUM_SINGLE_DECL(dacl_in_enum, BLOCK_EN, DACL_SRC, dacl_in_txt); + +static const char * const mono_txt[] = { "Stereo", "Mono"}; +static SOC_ENUM_SINGLE_DECL(mono_enum, VOL_CTRL1, DAC_MONO, mono_txt); + +static const struct snd_kcontrol_new t9015_snd_controls[] = { + /* Volume Controls */ + SOC_ENUM("Playback Channel Mode", mono_enum), + SOC_SINGLE("Playback Switch", VOL_CTRL1, DAC_SOFT_MUTE, 1, 1), + SOC_DOUBLE_TLV("Playback Volume", VOL_CTRL1, DACL_VC, DACR_VC, + 0xff, 0, dac_vol_tlv), + + /* Ramp Controls */ + SOC_ENUM("Ramp Rate", ramp_rate_enum), + SOC_SINGLE("Volume Ramp Switch", VOL_CTRL1, VC_RAMP_MODE, 1, 0), + SOC_SINGLE("Mute Ramp Switch", VOL_CTRL1, MUTE_MODE, 1, 0), + SOC_SINGLE("Unmute Ramp Switch", VOL_CTRL1, UNMUTE_MODE, 1, 0), +}; + +static const struct snd_kcontrol_new t9015_right_dac_mux = + SOC_DAPM_ENUM("Right DAC Source", dacr_in_enum); +static const struct snd_kcontrol_new t9015_left_dac_mux = + SOC_DAPM_ENUM("Left DAC Source", dacl_in_enum); + +static const struct snd_soc_dapm_widget t9015_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("Right IN", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("Left IN", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("Right DAC Sel", SND_SOC_NOPM, 0, 0, + &t9015_right_dac_mux), + SND_SOC_DAPM_MUX("Left DAC Sel", SND_SOC_NOPM, 0, 0, + &t9015_left_dac_mux), + SND_SOC_DAPM_DAC("Right DAC", NULL, BLOCK_EN, DACR_EN, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, BLOCK_EN, DACL_EN, 0), + SND_SOC_DAPM_OUT_DRV("Right- Driver", BLOCK_EN, LORN_EN, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("Right+ Driver", BLOCK_EN, LORP_EN, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("Left- Driver", BLOCK_EN, LOLN_EN, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("Left+ Driver", BLOCK_EN, LOLP_EN, 0, + NULL, 0), + SND_SOC_DAPM_OUTPUT("LORN"), + SND_SOC_DAPM_OUTPUT("LORP"), + SND_SOC_DAPM_OUTPUT("LOLN"), + SND_SOC_DAPM_OUTPUT("LOLP"), +}; + +static const struct snd_soc_dapm_route t9015_dapm_routes[] = { + { "Right IN", NULL, "Playback" }, + { "Left IN", NULL, "Playback" }, + { "Right DAC Sel", "Right", "Right IN" }, + { "Right DAC Sel", "Left", "Left IN" }, + { "Left DAC Sel", "Right", "Right IN" }, + { "Left DAC Sel", "Left", "Left IN" }, + { "Right DAC", NULL, "Right DAC Sel" }, + { "Left DAC", NULL, "Left DAC Sel" }, + { "Right- Driver", NULL, "Right DAC" }, + { "Right+ Driver", NULL, "Right DAC" }, + { "Left- Driver", NULL, "Left DAC" }, + { "Left+ Driver", NULL, "Left DAC" }, + { "LORN", NULL, "Right- Driver", }, + { "LORP", NULL, "Right+ Driver", }, + { "LOLN", NULL, "Left- Driver", }, + { "LOLP", NULL, "Left+ Driver", }, +}; + +static int t9015_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct t9015 *priv = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_component_update_bits(component, BLOCK_EN, + BIAS_CURRENT_EN, + BIAS_CURRENT_EN); + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, BLOCK_EN, + BIAS_CURRENT_EN, + 0); + break; + case SND_SOC_BIAS_STANDBY: + ret = regulator_enable(priv->avdd); + if (ret) { + dev_err(component->dev, "AVDD enable failed\n"); + return ret; + } + + if (now == SND_SOC_BIAS_OFF) { + snd_soc_component_update_bits(component, BLOCK_EN, + VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN, + VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN); + + mdelay(200); + snd_soc_component_update_bits(component, BLOCK_EN, + VMID_GEN_FAST, + 0); + } + + break; + case SND_SOC_BIAS_OFF: + snd_soc_component_update_bits(component, BLOCK_EN, + VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN, + 0); + + regulator_disable(priv->avdd); + break; + } + + return 0; +} + +static const struct snd_soc_component_driver t9015_codec_driver = { + .set_bias_level = t9015_set_bias_level, + .controls = t9015_snd_controls, + .num_controls = ARRAY_SIZE(t9015_snd_controls), + .dapm_widgets = t9015_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(t9015_dapm_widgets), + .dapm_routes = t9015_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(t9015_dapm_routes), + .suspend_bias_off = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config t9015_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = POWER_CFG, +}; + +static int t9015_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct t9015 *priv; + void __iomem *regs; + struct regmap *regmap; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + if (PTR_ERR(priv->pclk) != -EPROBE_DEFER) + dev_err(dev, "failed to get core clock\n"); + return PTR_ERR(priv->pclk); + } + + priv->avdd = devm_regulator_get(dev, "AVDD"); + if (IS_ERR(priv->avdd)) { + if (PTR_ERR(priv->avdd) != -EPROBE_DEFER) + dev_err(dev, "failed to AVDD\n"); + return PTR_ERR(priv->avdd); + } + + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dev, "core clock enable failed\n"); + return ret; + } + + ret = devm_add_action_or_reset(dev, + (void(*)(void *))clk_disable_unprepare, + priv->pclk); + if (ret) + return ret; + + ret = device_reset(dev); + if (ret) { + dev_err(dev, "reset failed\n"); + return ret; + } + + regs = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(regs)) { + dev_err(dev, "register map failed\n"); + return PTR_ERR(regs); + } + + regmap = devm_regmap_init_mmio(dev, regs, &t9015_regmap_config); + if (IS_ERR(regmap)) { + dev_err(dev, "regmap init failed\n"); + return PTR_ERR(regmap); + } + + /* + * Initialize output polarity: + * ATM the output polarity is fixed but in the future it might useful + * to add DT property to set this depending on the platform needs + */ + regmap_write(regmap, LINEOUT_CFG, 0x1111); + + return devm_snd_soc_register_component(dev, &t9015_codec_driver, + &t9015_dai, 1); +} + +static const struct of_device_id t9015_ids[] = { + { .compatible = "amlogic,t9015", }, + { } +}; +MODULE_DEVICE_TABLE(of, t9015_ids); + +static struct platform_driver t9015_driver = { + .driver = { + .name = "t9015-codec", + .of_match_table = of_match_ptr(t9015_ids), + }, + .probe = t9015_probe, +}; + +module_platform_driver(t9015_driver); + +MODULE_DESCRIPTION("ASoC Amlogic T9015 codec driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL"); From 95e9e205fcbe34d003c558e0a98e6ae6f9ab3a61 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Fri, 7 Feb 2020 13:03:45 +0100 Subject: [PATCH 152/415] ASoC: dt-bindings: stm32: convert i2s to json-schema Convert the STM32 I2S bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan Reviewed-by: Rob Herring Link: https://lore.kernel.org/r/20200207120345.24672-1-olivier.moysan@st.com Signed-off-by: Mark Brown --- .../bindings/sound/st,stm32-i2s.txt | 62 ------------- .../bindings/sound/st,stm32-i2s.yaml | 87 +++++++++++++++++++ 2 files changed, 87 insertions(+), 62 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt deleted file mode 100644 index cbf24bcd1b8d3c..00000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ /dev/null @@ -1,62 +0,0 @@ -STMicroelectronics STM32 SPI/I2S Controller - -The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. -Only some SPI instances support I2S. - -Required properties: - - compatible: Must be "st,stm32h7-i2s" - - reg: Offset and length of the device's register set. - - interrupts: Must contain the interrupt line id. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k". - "i2sclk": clock which feeds the internal clock generator - "pclk": clock which feeds the peripheral bus interface - "x8k": I2S parent clock for sampling rates multiple of 8kHz. - "x11k": I2S parent clock for sampling rates multiple of 11.025kHz. - - dmas: DMA specifiers for tx and rx dma. - See Documentation/devicetree/bindings/dma/stm32-dma.txt. - - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -Optional properties: - - resets: Reference to a reset controller asserting the reset controller - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&i2s2_port>; -}; - -i2s2: audio-controller@40003800 { - compatible = "st,stm32h7-i2s"; - reg = <0x40003800 0x400>; - interrupts = <36>; - clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "i2sclk", "x8k", "x11k"; - dmas = <&dmamux2 2 39 0x400 0x1>, - <&dmamux2 3 40 0x400 0x1>; - dma-names = "rx", "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s2>; - - i2s2_port: port@0 { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; -}; - -audio-codec { - codec_port: port@0 { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml new file mode 100644 index 00000000000000..f32410890589f6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 SPI/I2S Controller + +maintainers: + - Olivier Moysan + +description: + The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. + Only some SPI instances support I2S. + +properties: + compatible: + enum: + - st,stm32h7-i2s + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock feeding the peripheral bus interface. + - description: clock feeding the internal clock generator. + - description: I2S parent clock for sampling rates multiple of 8kHz. + - description: I2S parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: i2sclk + - const: x8k + - const: x11k + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio capture DMA. + - description: audio playback DMA. + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include + #include + i2s2: audio-controller@4000b000 { + compatible = "st,stm32h7-i2s"; + #sound-dai-cells = <0>; + reg = <0x4000b000 0x400>; + clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + interrupts = ; + dmas = <&dmamux1 39 0x400 0x01>, + <&dmamux1 40 0x400 0x01>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pins_a>; + }; + +... From 9d6ee3656a9fbfe906be5ce6f828f1639da1ee7f Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 19 Feb 2020 12:50:48 +0100 Subject: [PATCH 153/415] ASoC: dpcm: remove confusing trace in dpcm_get_be() Now that dpcm_get_be() is used in dpcm_end_walk_at_be(), it is not a error if this function does not find a BE for the provided widget. Remove the related dev_err() trace which is confusing since things might be working as expected. When called from dpcm_add_paths(), it is an error if dpcm_get_be() fails to find a BE for the provided widget. The necessary error trace is already done in this case. Fixes: 027a48387183 ("ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()") Signed-off-by: Jerome Brunet Tested-by: Pierre-Louis Bossart Acked-by: Kuninori Morimoto Cc: Kuninori Morimoto Link: https://lore.kernel.org/r/20200219115048.934678-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077ca..aff27c8599ef98 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1270,9 +1270,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, } } - /* dai link name and stream name set correctly ? */ - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", - stream ? "capture" : "playback", widget->name); + /* Widget provided is not a BE */ return NULL; } From dc7f090d9ab2f36fc404f8d903f806a1b811739e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 20 Feb 2020 12:56:54 +0000 Subject: [PATCH 154/415] ASoC: samsung: Update dependencies for Arizona machine drivers Currently it is possible to get the following bad config: WARNING: unmet direct dependencies detected for SND_SOC_WM5110 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_WM5110 [=n] commit ea00d95200d0 ("ASoC: Use imply for SND_SOC_ALL_CODECS") commit d8dd3f92a6ba ("ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout") After these two patches the machine drivers still selects the SND_SOC_WM5110 symbol which doesn't take account of the dependency added on the MFD_WM5110 symbol, fix this by also adding a dependency on MFD_WM5110 itself. Reported-by: Randy Dunlap Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20200220125654.7064-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1a0b163ca47b5c..112911dc271ba5 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -151,7 +151,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on MFD_ARIZONA && I2C && SPI_MASTER + depends on MFD_ARIZONA && MFD_WM5102 && MFD_WM5110 && I2C && SPI_MASTER depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5102 @@ -204,7 +204,7 @@ config SND_SOC_ARNDALE config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" - depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && MFD_ARIZONA && MFD_WM5110 && I2C && SPI_MASTER depends on GPIOLIB || COMPILE_TEST select SND_SOC_MAX98504 select SND_SOC_WM5110 From d2ad9d6ca5b2435754a0fd811f57d30914c612ce Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 20 Feb 2020 19:10:27 +0200 Subject: [PATCH 155/415] ASoC: intel/skl/hda - add no-HDMI cases to generic HDA driver Extend the generic HDA driver to support systems where iDisp/HDMI audio codecs are disabled for some reason. Switch codecs to SoC dummy in the affected DAI links. This allows to reuse existing topologies for this case. Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206085 BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1163677 BugLink: https://github.com/thesofproject/linux/issues/1658 Link: https://lore.kernel.org/r/20200220171028.22023-2-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_common.h | 4 ++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 25 ++++++++++++++++---- 2 files changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index d6150670ca0547..e8545d13062fd0 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -49,6 +49,10 @@ static inline int skl_hda_hdmi_build_controls(struct snd_soc_card *card) struct snd_soc_component *component; struct skl_hda_hdmi_pcm *pcm; + /* HDMI disabled, do not create controls */ + if (list_empty(&ctx->hdmi_pcm_list)) + return 0; + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct skl_hda_hdmi_pcm, head); component = pcm->codec_dai->component; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 11eaee9ae41f72..fe2d3a23a4efe2 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -61,6 +61,9 @@ static const struct snd_soc_dapm_route skl_hda_map[] = { { "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" }, }; +SND_SOC_DAILINK_DEF(dummy_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("snd-soc-dummy", "snd-soc-dummy-dai"))); + static int skl_hda_card_late_probe(struct snd_soc_card *card) { return skl_hda_hdmi_jack_init(card); @@ -114,13 +117,19 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) { struct snd_soc_card *card = &hda_soc_card; struct snd_soc_dai_link *dai_link; - u32 codec_count, codec_mask; + u32 codec_count, codec_mask, idisp_mask; int i, num_links, num_route; codec_mask = mach_params->codec_mask; codec_count = hweight_long(codec_mask); + idisp_mask = codec_mask & IDISP_CODEC_MASK; + + if (!codec_count || codec_count > 2 || + (codec_count == 2 && !idisp_mask)) + return -EINVAL; - if (codec_count == 1 && codec_mask & IDISP_CODEC_MASK) { + if (codec_mask == idisp_mask) { + /* topology with iDisp as the only HDA codec */ num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; @@ -135,13 +144,19 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) skl_hda_be_dai_links[IDISP_DAI_COUNT + HDAC_DAI_COUNT + i]; } - } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) { + } else { + /* topology with external and iDisp HDA codecs */ num_links = ARRAY_SIZE(skl_hda_be_dai_links); num_route = ARRAY_SIZE(skl_hda_map); card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); - } else { - return -EINVAL; + if (!idisp_mask) { + for (i = 0; i < IDISP_DAI_COUNT; i++) { + skl_hda_be_dai_links[i].codecs = dummy_codec; + skl_hda_be_dai_links[i].num_codecs = + ARRAY_SIZE(dummy_codec); + } + } } card->num_links = num_links; From 71cc8abb6ec705ce4efbb54e401004687d40a641 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 20 Feb 2020 19:10:28 +0200 Subject: [PATCH 156/415] ASoC: SOF: Intel: hda: allow operation without i915 gfx Add support to configure the HDA controller with an external HDA codec even if iDisp codec in i915 is not available. This can happen for multiple reasons: - internal graphics is disabled on the system - i915 driver is not enabled in kernel or it fails to init - i915 codec reports error in HDA codec probe - HDA codec driver probe fails Address all these scenarios, but keep using the existing topology. In case failures occur, HDMI PCM nodes are created, but they will report error if application tries to use them. No ALSA mixer controls are created. If the external HDA codec init fails as well, SOF probe will return error as before. Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206085 BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1163677 BugLink: https://github.com/thesofproject/linux/issues/1658 Link: https://lore.kernel.org/r/20200220171028.22023-3-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 11 ++++++++++- sound/soc/sof/intel/hda.c | 22 ++++++++-------------- 2 files changed, 18 insertions(+), 15 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index ff45075ef72038..3041fbbb010a22 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -113,8 +113,14 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (ret < 0) return ret; - if ((resp & 0xFFFF0000) == IDISP_VID_INTEL) + if ((resp & 0xFFFF0000) == IDISP_VID_INTEL) { + if (!hdev->bus->audio_component) { + dev_dbg(sdev->dev, + "iDisp hw present but no driver\n"); + return -ENOENT; + } hda_priv->need_display_power = true; + } /* * if common HDMI codec driver is not used, codec load @@ -203,6 +209,9 @@ int hda_codec_i915_exit(struct snd_sof_dev *sdev) struct hdac_bus *bus = sof_to_bus(sdev); int ret; + if (!bus->audio_component) + return 0; + /* power down unconditionally */ snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 726a9ef2d627f5..7ca887041a3441 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -288,10 +288,8 @@ static int hda_init(struct snd_sof_dev *sdev) /* init i915 and HDMI codecs */ ret = hda_codec_i915_init(sdev); - if (ret < 0) { - dev_err(sdev->dev, "error: init i915 and HDMI codec failed\n"); - return ret; - } + if (ret < 0) + dev_warn(sdev->dev, "init of i915 and HDMI codec failed\n"); /* get controller capabilities */ ret = hda_dsp_ctrl_get_caps(sdev); @@ -365,9 +363,6 @@ static int hda_init_caps(struct snd_sof_dev *sdev) if (ret < 0) { dev_err(bus->dev, "error: init chip failed with ret: %d\n", ret); -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - hda_codec_i915_exit(sdev); -#endif return ret; } @@ -379,7 +374,7 @@ static int hda_init_caps(struct snd_sof_dev *sdev) hda_codec_probe_bus(sdev, hda_codec_use_common_hdmi); if (!HDA_IDISP_CODEC(bus->codec_mask)) - hda_codec_i915_exit(sdev); + hda_codec_i915_display_power(sdev, false); /* * we are done probing so decrement link counts @@ -699,12 +694,11 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) /* * If no machine driver is found, then: * - * hda machine driver is used if : - * 1. there is one HDMI codec and one external HDAudio codec - * 2. only HDMI codec + * generic hda machine driver can handle: + * - one HDMI codec, and/or + * - one external HDAudio codec */ - if (!pdata->machine && codec_num <= 2 && - HDA_IDISP_CODEC(bus->codec_mask)) { + if (!pdata->machine && codec_num <= 2) { hda_mach = snd_soc_acpi_intel_hda_machines; /* topology: use the info from hda_machines */ @@ -714,7 +708,7 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) dev_info(bus->dev, "using HDA machine driver %s now\n", hda_mach->drv_name); - if (codec_num == 1) + if (codec_num == 1 && HDA_IDISP_CODEC(bus->codec_mask)) idisp_str = "-idisp"; else idisp_str = ""; From 4ee67cbd97668ab1b17d86d85348302c0b7490cd Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Thu, 20 Feb 2020 15:07:58 -0600 Subject: [PATCH 157/415] dt-bindings: sound: Add TLV320ADCx140 dt bindings Add dt bindings for the TLV320ADCx140 Burr-Brown ADC. The initial support is for the following: TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200220210759.31466-2-dmurphy@ti.com Signed-off-by: Mark Brown --- .../bindings/sound/tlv320adcx140.yaml | 83 +++++++++++++++++++ 1 file changed, 83 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml new file mode 100644 index 00000000000000..1433ff62b14f02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tlv320adcx140.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter + +maintainers: + - Dan Murphy + +description: | + The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital + PDM microphones recording), high-performance audio, analog-to-digital + converter (ADC) with analog inputs supporting up to 2V RMS. The TLV320ADCX140 + family supports line and microphone Inputs, and offers a programmable + microphone bias or supply voltage generation. + + Specifications can be found at: + http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf + http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf + http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf + +properties: + compatible: + oneOf: + - const: ti,tlv320adc3140 + - const: ti,tlv320adc5140 + - const: ti,tlv320adc6140 + + reg: + maxItems: 1 + description: | + I2C addresss of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpios: + description: | + GPIO used for hardware reset. + + areg-supply: + description: | + Regulator with AVDD at 3.3V. If not defined then the internal regulator + is enabled. + + ti,mic-bias-source: + description: | + Indicates the source for MIC Bias. + 0 - Mic bias is set to VREF + 1 - Mic bias is set to VREF × 1.096 + 6 - Mic bias is set to AVDD + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + - enum: [0, 1, 6] + + ti,vref-source: + description: | + Indicates the source for MIC Bias. + 0 - Set VREF to 2.75V + 1 - Set VREF to 2.5V + 2 - Set VREF to 1.375V + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + - enum: [0, 1, 2] + +required: + - compatible + - reg + +examples: + - | + #include + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tlv320adc5140"; + reg = <0x4c>; + ti,use-internal-areg; + ti,mic-bias-source = <6>; + reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; + }; + }; From 689c7655b50c5de2b6f0f42fecfb37bde5acf040 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Thu, 20 Feb 2020 15:07:59 -0600 Subject: [PATCH 158/415] ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add the tlv320adcx140 codec driver family. The TLV320ADCx140 is a Burr-Brown™ highperformance, audio analog-to-digital converter (ADC) that supports simultaneous sampling of up to four analog channels or eight digital channels for the pulse density modulation (PDM) microphone input. The device supports line and microphone inputs, and allows for both single-ended and differential input configurations. Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200220210759.31466-3-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adcx140.c | 849 +++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 130 +++++ 4 files changed, 990 insertions(+) create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d957fd6980b10a..9e9d54e4576ce5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -196,6 +196,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS6424 imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 + imply SND_SOC_TLV320ADCX140 imply SND_SOC_TLV320AIC23_I2C imply SND_SOC_TLV320AIC23_SPI imply SND_SOC_TLV320AIC26 @@ -1334,6 +1335,14 @@ config SND_SOC_TLV320DAC33 tristate depends on I2C +config SND_SOC_TLV320ADCX140 + tristate "Texas Instruments TLV320ADCX140 CODEC family" + depends on I2C + select REGMAP_I2C + help + Add support for Texas Instruments tlv320adc3140, tlv320adc5140 and + tlv320adc6140 quad channel ADCs. + config SND_SOC_TS3A227E tristate "TI Headset/Mic detect and keypress chip" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ba1b4b3fa2da12..943ebc93fbc1cf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -218,6 +218,7 @@ snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-tlv320adcx140-objs := tlv320adcx140.o snd-soc-tscs42xx-objs := tscs42xx.o snd-soc-tscs454-objs := tscs454.o snd-soc-ts3a227e-objs := ts3a227e.o @@ -516,6 +517,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI) += snd-soc-tlv320aic32x4-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TLV320ADCX140) += snd-soc-tlv320adcx140.o obj-$(CONFIG_SND_SOC_TSCS42XX) += snd-soc-tscs42xx.o obj-$(CONFIG_SND_SOC_TSCS454) += snd-soc-tscs454.o obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c new file mode 100644 index 00000000000000..8182c584de9ccd --- /dev/null +++ b/sound/soc/codecs/tlv320adcx140.c @@ -0,0 +1,849 @@ +// SPDX-License-Identifier: GPL-2.0 +// TLV320ADCX140 Sound driver +// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tlv320adcx140.h" + +struct adcx140_priv { + struct snd_soc_component *component; + struct regulator *supply_areg; + struct gpio_desc *gpio_reset; + struct regmap *regmap; + struct device *dev; + + int micbias_vg; + + unsigned int dai_fmt; + unsigned int tdm_delay; + unsigned int slot_width; +}; + +static const struct reg_default adcx140_reg_defaults[] = { + { ADCX140_PAGE_SELECT, 0x00 }, + { ADCX140_SW_RESET, 0x00 }, + { ADCX140_SLEEP_CFG, 0x00 }, + { ADCX140_SHDN_CFG, 0x05 }, + { ADCX140_ASI_CFG0, 0x30 }, + { ADCX140_ASI_CFG1, 0x00 }, + { ADCX140_ASI_CFG2, 0x00 }, + { ADCX140_ASI_CH1, 0x00 }, + { ADCX140_ASI_CH2, 0x01 }, + { ADCX140_ASI_CH3, 0x02 }, + { ADCX140_ASI_CH4, 0x03 }, + { ADCX140_ASI_CH5, 0x04 }, + { ADCX140_ASI_CH6, 0x05 }, + { ADCX140_ASI_CH7, 0x06 }, + { ADCX140_ASI_CH8, 0x07 }, + { ADCX140_MST_CFG0, 0x02 }, + { ADCX140_MST_CFG1, 0x48 }, + { ADCX140_ASI_STS, 0xff }, + { ADCX140_CLK_SRC, 0x10 }, + { ADCX140_PDMCLK_CFG, 0x40 }, + { ADCX140_PDM_CFG, 0x00 }, + { ADCX140_GPIO_CFG0, 0x22 }, + { ADCX140_GPO_CFG1, 0x00 }, + { ADCX140_GPO_CFG2, 0x00 }, + { ADCX140_GPO_CFG3, 0x00 }, + { ADCX140_GPO_CFG4, 0x00 }, + { ADCX140_GPO_VAL, 0x00 }, + { ADCX140_GPIO_MON, 0x00 }, + { ADCX140_GPI_CFG0, 0x00 }, + { ADCX140_GPI_CFG1, 0x00 }, + { ADCX140_GPI_MON, 0x00 }, + { ADCX140_INT_CFG, 0x00 }, + { ADCX140_INT_MASK0, 0xff }, + { ADCX140_INT_LTCH0, 0x00 }, + { ADCX140_BIAS_CFG, 0x00 }, + { ADCX140_CH1_CFG0, 0x00 }, + { ADCX140_CH1_CFG1, 0x00 }, + { ADCX140_CH1_CFG2, 0xc9 }, + { ADCX140_CH1_CFG3, 0x80 }, + { ADCX140_CH1_CFG4, 0x00 }, + { ADCX140_CH2_CFG0, 0x00 }, + { ADCX140_CH2_CFG1, 0x00 }, + { ADCX140_CH2_CFG2, 0xc9 }, + { ADCX140_CH2_CFG3, 0x80 }, + { ADCX140_CH2_CFG4, 0x00 }, + { ADCX140_CH3_CFG0, 0x00 }, + { ADCX140_CH3_CFG1, 0x00 }, + { ADCX140_CH3_CFG2, 0xc9 }, + { ADCX140_CH3_CFG3, 0x80 }, + { ADCX140_CH3_CFG4, 0x00 }, + { ADCX140_CH4_CFG0, 0x00 }, + { ADCX140_CH4_CFG1, 0x00 }, + { ADCX140_CH4_CFG2, 0xc9 }, + { ADCX140_CH4_CFG3, 0x80 }, + { ADCX140_CH4_CFG4, 0x00 }, + { ADCX140_CH5_CFG2, 0xc9 }, + { ADCX140_CH5_CFG3, 0x80 }, + { ADCX140_CH5_CFG4, 0x00 }, + { ADCX140_CH6_CFG2, 0xc9 }, + { ADCX140_CH6_CFG3, 0x80 }, + { ADCX140_CH6_CFG4, 0x00 }, + { ADCX140_CH7_CFG2, 0xc9 }, + { ADCX140_CH7_CFG3, 0x80 }, + { ADCX140_CH7_CFG4, 0x00 }, + { ADCX140_CH8_CFG2, 0xc9 }, + { ADCX140_CH8_CFG3, 0x80 }, + { ADCX140_CH8_CFG4, 0x00 }, + { ADCX140_DSP_CFG0, 0x01 }, + { ADCX140_DSP_CFG1, 0x40 }, + { ADCX140_DRE_CFG0, 0x7b }, + { ADCX140_IN_CH_EN, 0xf0 }, + { ADCX140_ASI_OUT_CH_EN, 0x00 }, + { ADCX140_PWR_CFG, 0x00 }, + { ADCX140_DEV_STS0, 0x00 }, + { ADCX140_DEV_STS1, 0x80 }, +}; + +static const struct regmap_range_cfg adcx140_ranges[] = { + { + .range_min = 0, + .range_max = 12 * 128, + .selector_reg = ADCX140_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static bool adcx140_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADCX140_SW_RESET: + case ADCX140_DEV_STS0: + case ADCX140_DEV_STS1: + case ADCX140_ASI_STS: + return true; + default: + return false; + } +} + +static const struct regmap_config adcx140_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adcx140_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adcx140_reg_defaults), + .cache_type = REGCACHE_FLAT, + .ranges = adcx140_ranges, + .num_ranges = ARRAY_SIZE(adcx140_ranges), + .max_register = 12 * 128, + .volatile_reg = adcx140_volatile, +}; + +/* Digital Volume control. From -100 to 27 dB in 0.5 dB steps */ +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); + +/* ADC gain. From 0 to 42 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); + +static const char * const resistor_text[] = { + "2.5 kOhm", "10 kOhm", "20 kOhm" +}; + +static SOC_ENUM_SINGLE_DECL(in1_resistor_enum, ADCX140_CH1_CFG0, 2, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in2_resistor_enum, ADCX140_CH2_CFG0, 2, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in3_resistor_enum, ADCX140_CH3_CFG0, 2, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in4_resistor_enum, ADCX140_CH4_CFG0, 2, + resistor_text); + +static const struct snd_kcontrol_new in1_resistor_controls[] = { + SOC_DAPM_ENUM("CH1 Resistor Select", in1_resistor_enum), +}; +static const struct snd_kcontrol_new in2_resistor_controls[] = { + SOC_DAPM_ENUM("CH2 Resistor Select", in2_resistor_enum), +}; +static const struct snd_kcontrol_new in3_resistor_controls[] = { + SOC_DAPM_ENUM("CH3 Resistor Select", in3_resistor_enum), +}; +static const struct snd_kcontrol_new in4_resistor_controls[] = { + SOC_DAPM_ENUM("CH4 Resistor Select", in4_resistor_enum), +}; + +/* Analog/Digital Selection */ +static const char *adcx140_mic_sel_text[] = {"Analog", "Line In", "Digital"}; +static const char *adcx140_analog_sel_text[] = {"Analog", "Line In"}; + +static SOC_ENUM_SINGLE_DECL(adcx140_mic1p_enum, + ADCX140_CH1_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic1p_control = +SOC_DAPM_ENUM("MIC1P MUX", adcx140_mic1p_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic1_analog_enum, + ADCX140_CH1_CFG0, 7, + adcx140_analog_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic1_analog_control = +SOC_DAPM_ENUM("MIC1 Analog MUX", adcx140_mic1_analog_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic1m_enum, + ADCX140_CH1_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic1m_control = +SOC_DAPM_ENUM("MIC1M MUX", adcx140_mic1m_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic2p_enum, + ADCX140_CH2_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic2p_control = +SOC_DAPM_ENUM("MIC2P MUX", adcx140_mic2p_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic2_analog_enum, + ADCX140_CH2_CFG0, 7, + adcx140_analog_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic2_analog_control = +SOC_DAPM_ENUM("MIC2 Analog MUX", adcx140_mic2_analog_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic2m_enum, + ADCX140_CH2_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic2m_control = +SOC_DAPM_ENUM("MIC2M MUX", adcx140_mic2m_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic3p_enum, + ADCX140_CH3_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic3p_control = +SOC_DAPM_ENUM("MIC3P MUX", adcx140_mic3p_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic3_analog_enum, + ADCX140_CH3_CFG0, 7, + adcx140_analog_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic3_analog_control = +SOC_DAPM_ENUM("MIC3 Analog MUX", adcx140_mic3_analog_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic3m_enum, + ADCX140_CH3_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic3m_control = +SOC_DAPM_ENUM("MIC3M MUX", adcx140_mic3m_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic4p_enum, + ADCX140_CH4_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic4p_control = +SOC_DAPM_ENUM("MIC4P MUX", adcx140_mic4p_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic4_analog_enum, + ADCX140_CH4_CFG0, 7, + adcx140_analog_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic4_analog_control = +SOC_DAPM_ENUM("MIC4 Analog MUX", adcx140_mic4_analog_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic4m_enum, + ADCX140_CH4_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic4m_control = +SOC_DAPM_ENUM("MIC4M MUX", adcx140_mic4m_enum); + +static const struct snd_kcontrol_new adcx140_dapm_ch1_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 7, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch2_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 6, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 5, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0); + +/* Output Mixer */ +static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Digital CH1 Switch", 0, 0, 0, 0), + SOC_DAPM_SINGLE("Digital CH2 Switch", 0, 0, 0, 0), + SOC_DAPM_SINGLE("Digital CH3 Switch", 0, 0, 0, 0), + SOC_DAPM_SINGLE("Digital CH4 Switch", 0, 0, 0, 0), +}; + +static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { + /* Analog Differential Inputs */ + SND_SOC_DAPM_INPUT("MIC1P"), + SND_SOC_DAPM_INPUT("MIC1M"), + SND_SOC_DAPM_INPUT("MIC2P"), + SND_SOC_DAPM_INPUT("MIC2M"), + SND_SOC_DAPM_INPUT("MIC3P"), + SND_SOC_DAPM_INPUT("MIC3M"), + SND_SOC_DAPM_INPUT("MIC4P"), + SND_SOC_DAPM_INPUT("MIC4M"), + + SND_SOC_DAPM_OUTPUT("CH1_OUT"), + SND_SOC_DAPM_OUTPUT("CH2_OUT"), + SND_SOC_DAPM_OUTPUT("CH3_OUT"), + SND_SOC_DAPM_OUTPUT("CH4_OUT"), + SND_SOC_DAPM_OUTPUT("CH5_OUT"), + SND_SOC_DAPM_OUTPUT("CH6_OUT"), + SND_SOC_DAPM_OUTPUT("CH7_OUT"), + SND_SOC_DAPM_OUTPUT("CH8_OUT"), + + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + &adcx140_output_mixer_controls[0], + ARRAY_SIZE(adcx140_output_mixer_controls)), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1P Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic1p_control), + SND_SOC_DAPM_MUX("MIC2P Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic2p_control), + SND_SOC_DAPM_MUX("MIC3P Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic3p_control), + SND_SOC_DAPM_MUX("MIC4P Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic4p_control), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1 Analog Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic1_analog_control), + SND_SOC_DAPM_MUX("MIC2 Analog Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic2_analog_control), + SND_SOC_DAPM_MUX("MIC3 Analog Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic3_analog_control), + SND_SOC_DAPM_MUX("MIC4 Analog Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic4_analog_control), + + SND_SOC_DAPM_MUX("MIC1M Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic1m_control), + SND_SOC_DAPM_MUX("MIC2M Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic2m_control), + SND_SOC_DAPM_MUX("MIC3M Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic3m_control), + SND_SOC_DAPM_MUX("MIC4M Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic4m_control), + + SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH4", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_ADC("CH1_ADC", "CH1 Capture", ADCX140_IN_CH_EN, 7, 0), + SND_SOC_DAPM_ADC("CH2_ADC", "CH2 Capture", ADCX140_IN_CH_EN, 6, 0), + SND_SOC_DAPM_ADC("CH3_ADC", "CH3 Capture", ADCX140_IN_CH_EN, 5, 0), + SND_SOC_DAPM_ADC("CH4_ADC", "CH4 Capture", ADCX140_IN_CH_EN, 4, 0), + + SND_SOC_DAPM_SWITCH("CH1_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch1_en_switch), + SND_SOC_DAPM_SWITCH("CH2_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch2_en_switch), + SND_SOC_DAPM_SWITCH("CH3_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch3_en_switch), + SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch4_en_switch), + + SND_SOC_DAPM_MUX("IN1 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, + in1_resistor_controls), + SND_SOC_DAPM_MUX("IN2 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, + in2_resistor_controls), + SND_SOC_DAPM_MUX("IN3 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, + in3_resistor_controls), + SND_SOC_DAPM_MUX("IN4 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, + in4_resistor_controls), +}; + +static const struct snd_soc_dapm_route adcx140_audio_map[] = { + /* Outputs */ + {"CH1_OUT", NULL, "Output Mixer"}, + {"CH2_OUT", NULL, "Output Mixer"}, + {"CH3_OUT", NULL, "Output Mixer"}, + {"CH4_OUT", NULL, "Output Mixer"}, + + {"CH1_ASI_EN", "Switch", "CH1_ADC"}, + {"CH2_ASI_EN", "Switch", "CH2_ADC"}, + {"CH3_ASI_EN", "Switch", "CH3_ADC"}, + {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + + /* Mic input */ + {"CH1_ADC", NULL, "MIC_GAIN_CTL_CH1"}, + {"CH2_ADC", NULL, "MIC_GAIN_CTL_CH2"}, + {"CH3_ADC", NULL, "MIC_GAIN_CTL_CH3"}, + {"CH4_ADC", NULL, "MIC_GAIN_CTL_CH4"}, + + {"MIC_GAIN_CTL_CH1", NULL, "IN1 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH1", NULL, "IN1 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH2", NULL, "IN2 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH2", NULL, "IN2 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH3", NULL, "IN3 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH3", NULL, "IN3 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH4", NULL, "IN4 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH4", NULL, "IN4 Analog Mic Resistor"}, + + {"IN1 Analog Mic Resistor", "2.5 kOhm", "MIC1P Input Mux"}, + {"IN1 Analog Mic Resistor", "10 kOhm", "MIC1P Input Mux"}, + {"IN1 Analog Mic Resistor", "20 kOhm", "MIC1P Input Mux"}, + + {"IN1 Analog Mic Resistor", "2.5 kOhm", "MIC1M Input Mux"}, + {"IN1 Analog Mic Resistor", "10 kOhm", "MIC1M Input Mux"}, + {"IN1 Analog Mic Resistor", "20 kOhm", "MIC1M Input Mux"}, + + {"IN2 Analog Mic Resistor", "2.5 kOhm", "MIC2P Input Mux"}, + {"IN2 Analog Mic Resistor", "10 kOhm", "MIC2P Input Mux"}, + {"IN2 Analog Mic Resistor", "20 kOhm", "MIC2P Input Mux"}, + + {"IN2 Analog Mic Resistor", "2.5 kOhm", "MIC2M Input Mux"}, + {"IN2 Analog Mic Resistor", "10 kOhm", "MIC2M Input Mux"}, + {"IN2 Analog Mic Resistor", "20 kOhm", "MIC2M Input Mux"}, + + {"IN3 Analog Mic Resistor", "2.5 kOhm", "MIC3P Input Mux"}, + {"IN3 Analog Mic Resistor", "10 kOhm", "MIC3P Input Mux"}, + {"IN3 Analog Mic Resistor", "20 kOhm", "MIC3P Input Mux"}, + + {"IN3 Analog Mic Resistor", "2.5 kOhm", "MIC3M Input Mux"}, + {"IN3 Analog Mic Resistor", "10 kOhm", "MIC3M Input Mux"}, + {"IN3 Analog Mic Resistor", "20 kOhm", "MIC3M Input Mux"}, + + {"IN4 Analog Mic Resistor", "2.5 kOhm", "MIC4P Input Mux"}, + {"IN4 Analog Mic Resistor", "10 kOhm", "MIC4P Input Mux"}, + {"IN4 Analog Mic Resistor", "20 kOhm", "MIC4P Input Mux"}, + + {"IN4 Analog Mic Resistor", "2.5 kOhm", "MIC4M Input Mux"}, + {"IN4 Analog Mic Resistor", "10 kOhm", "MIC4M Input Mux"}, + {"IN4 Analog Mic Resistor", "20 kOhm", "MIC4M Input Mux"}, + + {"MIC1 Analog Mux", "Line In", "MIC1P"}, + {"MIC2 Analog Mux", "Line In", "MIC2P"}, + {"MIC3 Analog Mux", "Line In", "MIC3P"}, + {"MIC4 Analog Mux", "Line In", "MIC4P"}, + + {"MIC1P Input Mux", "Analog", "MIC1P"}, + {"MIC1M Input Mux", "Analog", "MIC1M"}, + {"MIC2P Input Mux", "Analog", "MIC2P"}, + {"MIC2M Input Mux", "Analog", "MIC2M"}, + {"MIC3P Input Mux", "Analog", "MIC3P"}, + {"MIC3M Input Mux", "Analog", "MIC3M"}, + {"MIC4P Input Mux", "Analog", "MIC4P"}, + {"MIC4M Input Mux", "Analog", "MIC4M"}, +}; + +static const struct snd_kcontrol_new adcx140_snd_controls[] = { + SOC_SINGLE_TLV("Analog CH1 Mic Gain Volume", ADCX140_CH1_CFG1, 2, 42, 0, + adc_tlv), + SOC_SINGLE_TLV("Analog CH2 Mic Gain Volume", ADCX140_CH1_CFG2, 2, 42, 0, + adc_tlv), + SOC_SINGLE_TLV("Analog CH3 Mic Gain Volume", ADCX140_CH1_CFG3, 2, 42, 0, + adc_tlv), + SOC_SINGLE_TLV("Analog CH4 Mic Gain Volume", ADCX140_CH1_CFG4, 2, 42, 0, + adc_tlv), + + SOC_SINGLE_TLV("Digital CH1 Out Volume", ADCX140_CH1_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH2 Out Volume", ADCX140_CH2_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH3 Out Volume", ADCX140_CH3_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH4 Out Volume", ADCX140_CH4_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH5 Out Volume", ADCX140_CH5_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH6 Out Volume", ADCX140_CH6_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH7 Out Volume", ADCX140_CH7_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH8 Out Volume", ADCX140_CH8_CFG2, + 0, 0xff, 0, dig_vol_tlv), +}; + +static int adcx140_reset(struct adcx140_priv *adcx140) +{ + int ret = 0; + + if (adcx140->gpio_reset) { + gpiod_direction_output(adcx140->gpio_reset, 0); + /* 8.4.1: wait for hw shutdown (25ms) + >= 1ms */ + usleep_range(30000, 100000); + gpiod_direction_output(adcx140->gpio_reset, 1); + } else { + ret = regmap_write(adcx140->regmap, ADCX140_SW_RESET, + ADCX140_RESET); + } + + /* 8.4.2: wait >= 10 ms after entering sleep mode. */ + usleep_range(10000, 100000); + + return 0; +} + +static int adcx140_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + u8 data = 0; + + switch (params_width(params)) { + case 16: + data = ADCX140_16_BIT_WORD; + break; + case 20: + data = ADCX140_20_BIT_WORD; + break; + case 24: + data = ADCX140_24_BIT_WORD; + break; + case 32: + data = ADCX140_32_BIT_WORD; + break; + default: + dev_err(component->dev, "%s: Unsupported width %d\n", + __func__, params_width(params)); + return -EINVAL; + } + + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, + ADCX140_WORD_LEN_MSK, data); + + return 0; +} + +static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + u8 iface_reg1 = 0; + u8 iface_reg2 = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg2 |= ADCX140_BCLK_FSYNC_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(component->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* signal polarity */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + iface_reg1 |= ADCX140_FSYNCINV_BIT; + break; + case SND_SOC_DAIFMT_IB_IF: + iface_reg1 |= ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT; + break; + case SND_SOC_DAIFMT_IB_NF: + iface_reg1 |= ADCX140_BCLKINV_BIT; + break; + case SND_SOC_DAIFMT_NB_NF: + break; + default: + dev_err(component->dev, "Invalid DAI clock signal polarity\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg1 |= ADCX140_I2S_MODE_BIT; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= ADCX140_LEFT_JUST_BIT; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + break; + default: + dev_err(component->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, + ADCX140_FSYNCINV_BIT | + ADCX140_BCLKINV_BIT | + ADCX140_ASI_FORMAT_MSK, + iface_reg1); + snd_soc_component_update_bits(component, ADCX140_MST_CFG0, + ADCX140_BCLK_FSYNC_MASTER, iface_reg2); + + return 0; +} + +static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = codec_dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + unsigned int lsb; + + if (tx_mask != rx_mask) { + dev_err(component->dev, "tx and rx masks must be symmetric\n"); + return -EINVAL; + } + + /* TDM based on DSP mode requires slots to be adjacent */ + lsb = __ffs(tx_mask); + if ((lsb + 1) != __fls(tx_mask)) { + dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + return -EINVAL; + } + + switch (slot_width) { + case 16: + case 20: + case 24: + case 32: + break; + default: + dev_err(component->dev, "Unsupported slot width %d\n", slot_width); + return -EINVAL; + } + + adcx140->tdm_delay = lsb; + adcx140->slot_width = slot_width; + + return 0; +} + +static int adcx140_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + int offset = 0; + int width = adcx140->slot_width; + + if (!width) + width = substream->runtime->sample_bits; + + /* TDM slot selection only valid in DSP_A/_B mode */ + if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_A) + offset += (adcx140->tdm_delay * width + 1); + else if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_B) + offset += adcx140->tdm_delay * width; + + /* Configure data offset */ + snd_soc_component_update_bits(component, ADCX140_ASI_CFG1, + ADCX140_TX_OFFSET_MASK, offset); + + return 0; +} + +static const struct snd_soc_dai_ops adcx140_dai_ops = { + .hw_params = adcx140_hw_params, + .set_fmt = adcx140_set_dai_fmt, + .prepare = adcx140_prepare, + .set_tdm_slot = adcx140_set_dai_tdm_slot, +}; + +static int adcx140_codec_probe(struct snd_soc_component *component) +{ + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + int sleep_cfg_val = ADCX140_WAKE_DEV; + u8 bias_source; + u8 vref_source; + int ret; + + ret = device_property_read_u8(adcx140->dev, "ti,mic-bias-source", + &bias_source); + if (ret) + bias_source = ADCX140_MIC_BIAS_VAL_VREF; + + if (bias_source != ADCX140_MIC_BIAS_VAL_VREF && + bias_source != ADCX140_MIC_BIAS_VAL_VREF_1096 && + bias_source != ADCX140_MIC_BIAS_VAL_AVDD) { + dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); + return -EINVAL; + } + + ret = device_property_read_u8(adcx140->dev, "ti,vref-source", + &vref_source); + if (ret) + vref_source = ADCX140_MIC_BIAS_VREF_275V; + + if (vref_source != ADCX140_MIC_BIAS_VREF_275V && + vref_source != ADCX140_MIC_BIAS_VREF_25V && + vref_source != ADCX140_MIC_BIAS_VREF_1375V) { + dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); + return -EINVAL; + } + + bias_source |= vref_source; + + ret = adcx140_reset(adcx140); + if (ret) + goto out; + + if(adcx140->supply_areg == NULL) + sleep_cfg_val |= ADCX140_AREG_INTERNAL; + + ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val); + if (ret) { + dev_err(adcx140->dev, "setting sleep config failed %d\n", ret); + goto out; + } + + /* 8.4.3: Wait >= 1ms after entering active mode. */ + usleep_range(1000, 100000); + + ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG, + ADCX140_MIC_BIAS_VAL_MSK | + ADCX140_MIC_BIAS_VREF_MSK, bias_source); + if (ret) + dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret); +out: + return ret; +} + +static int adcx140_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + int pwr_cfg = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + pwr_cfg = ADCX140_PWR_CFG_BIAS_PDZ | ADCX140_PWR_CFG_PLL_PDZ | + ADCX140_PWR_CFG_ADC_PDZ; + break; + case SND_SOC_BIAS_OFF: + pwr_cfg = 0x0; + break; + } + + return regmap_write(adcx140->regmap, ADCX140_PWR_CFG, pwr_cfg); +} + +static const struct snd_soc_component_driver soc_codec_driver_adcx140 = { + .probe = adcx140_codec_probe, + .set_bias_level = adcx140_set_bias_level, + .controls = adcx140_snd_controls, + .num_controls = ARRAY_SIZE(adcx140_snd_controls), + .dapm_widgets = adcx140_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adcx140_dapm_widgets), + .dapm_routes = adcx140_audio_map, + .num_dapm_routes = ARRAY_SIZE(adcx140_audio_map), + .suspend_bias_off = 1, + .idle_bias_on = 0, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct snd_soc_dai_driver adcx140_dai_driver[] = { + { + .name = "tlv320adcx140-codec", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = ADCX140_MAX_CHANNELS, + .rates = ADCX140_RATES, + .formats = ADCX140_FORMATS, + }, + .ops = &adcx140_dai_ops, + .symmetric_rates = 1, + } +}; + +static const struct of_device_id tlv320adcx140_of_match[] = { + { .compatible = "ti,tlv320adc3140" }, + { .compatible = "ti,tlv320adc5140" }, + { .compatible = "ti,tlv320adc6140" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320adcx140_of_match); + +static int adcx140_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct adcx140_priv *adcx140; + int ret; + + adcx140 = devm_kzalloc(&i2c->dev, sizeof(*adcx140), GFP_KERNEL); + if (!adcx140) + return -ENOMEM; + + adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(adcx140->gpio_reset)) + dev_info(&i2c->dev, "Reset GPIO not defined\n"); + + adcx140->supply_areg = devm_regulator_get_optional(adcx140->dev, + "areg"); + if (IS_ERR(adcx140->supply_areg)) { + if (PTR_ERR(adcx140->supply_areg) == -EPROBE_DEFER) + return -EPROBE_DEFER; + else + adcx140->supply_areg = NULL; + } else { + ret = regulator_enable(adcx140->supply_areg); + if (ret) { + dev_err(adcx140->dev, "Failed to enable areg\n"); + return ret; + } + } + + adcx140->regmap = devm_regmap_init_i2c(i2c, &adcx140_i2c_regmap); + if (IS_ERR(adcx140->regmap)) { + ret = PTR_ERR(adcx140->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + adcx140->dev = &i2c->dev; + i2c_set_clientdata(i2c, adcx140); + + return devm_snd_soc_register_component(&i2c->dev, + &soc_codec_driver_adcx140, + adcx140_dai_driver, 1); +} + +static const struct i2c_device_id adcx140_i2c_id[] = { + { "tlv320adc3140", 0 }, + { "tlv320adc5140", 1 }, + { "tlv320adc6140", 2 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, adcx140_i2c_id); + +static struct i2c_driver adcx140_i2c_driver = { + .driver = { + .name = "tlv320adcx140-codec", + .of_match_table = of_match_ptr(tlv320adcx140_of_match), + }, + .probe = adcx140_i2c_probe, + .id_table = adcx140_i2c_id, +}; +module_i2c_driver(adcx140_i2c_driver); + +MODULE_AUTHOR("Dan Murphy "); +MODULE_DESCRIPTION("ASoC TLV320ADCX140 CODEC Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h new file mode 100644 index 00000000000000..66b1c3b33f1eed --- /dev/null +++ b/sound/soc/codecs/tlv320adcx140.h @@ -0,0 +1,130 @@ +// SPDX-License-Identifier: GPL-2.0 +// TLV320ADCX104 Sound driver +// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/ + +#ifndef _TLV320ADCX140_H +#define _TLV320ADCX140_H + +#define ADCX140_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define ADCX140_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define ADCX140_PAGE_SELECT 0x00 +#define ADCX140_SW_RESET 0x01 +#define ADCX140_SLEEP_CFG 0x02 +#define ADCX140_SHDN_CFG 0x05 +#define ADCX140_ASI_CFG0 0x07 +#define ADCX140_ASI_CFG1 0x08 +#define ADCX140_ASI_CFG2 0x09 +#define ADCX140_ASI_CH1 0x0b +#define ADCX140_ASI_CH2 0x0c +#define ADCX140_ASI_CH3 0x0d +#define ADCX140_ASI_CH4 0x0e +#define ADCX140_ASI_CH5 0x0f +#define ADCX140_ASI_CH6 0x10 +#define ADCX140_ASI_CH7 0x11 +#define ADCX140_ASI_CH8 0x12 +#define ADCX140_MST_CFG0 0x13 +#define ADCX140_MST_CFG1 0x14 +#define ADCX140_ASI_STS 0x15 +#define ADCX140_CLK_SRC 0x16 +#define ADCX140_PDMCLK_CFG 0x1f +#define ADCX140_PDM_CFG 0x20 +#define ADCX140_GPIO_CFG0 0x21 +#define ADCX140_GPO_CFG1 0x22 +#define ADCX140_GPO_CFG2 0x23 +#define ADCX140_GPO_CFG3 0x24 +#define ADCX140_GPO_CFG4 0x25 +#define ADCX140_GPO_VAL 0x29 +#define ADCX140_GPIO_MON 0x2a +#define ADCX140_GPI_CFG0 0x2b +#define ADCX140_GPI_CFG1 0x2c +#define ADCX140_GPI_MON 0x2f +#define ADCX140_INT_CFG 0x32 +#define ADCX140_INT_MASK0 0x33 +#define ADCX140_INT_LTCH0 0x36 +#define ADCX140_BIAS_CFG 0x3b +#define ADCX140_CH1_CFG0 0x3c +#define ADCX140_CH1_CFG1 0x3d +#define ADCX140_CH1_CFG2 0x3e +#define ADCX140_CH1_CFG3 0x3f +#define ADCX140_CH1_CFG4 0x40 +#define ADCX140_CH2_CFG0 0x41 +#define ADCX140_CH2_CFG1 0x42 +#define ADCX140_CH2_CFG2 0x43 +#define ADCX140_CH2_CFG3 0x44 +#define ADCX140_CH2_CFG4 0x45 +#define ADCX140_CH3_CFG0 0x46 +#define ADCX140_CH3_CFG1 0x47 +#define ADCX140_CH3_CFG2 0x48 +#define ADCX140_CH3_CFG3 0x49 +#define ADCX140_CH3_CFG4 0x4a +#define ADCX140_CH4_CFG0 0x4b +#define ADCX140_CH4_CFG1 0x4c +#define ADCX140_CH4_CFG2 0x4d +#define ADCX140_CH4_CFG3 0x4e +#define ADCX140_CH4_CFG4 0x4f +#define ADCX140_CH5_CFG2 0x52 +#define ADCX140_CH5_CFG3 0x53 +#define ADCX140_CH5_CFG4 0x54 +#define ADCX140_CH6_CFG2 0x57 +#define ADCX140_CH6_CFG3 0x58 +#define ADCX140_CH6_CFG4 0x59 +#define ADCX140_CH7_CFG2 0x5c +#define ADCX140_CH7_CFG3 0x5d +#define ADCX140_CH7_CFG4 0x5e +#define ADCX140_CH8_CFG2 0x61 +#define ADCX140_CH8_CFG3 0x62 +#define ADCX140_CH8_CFG4 0x63 +#define ADCX140_DSP_CFG0 0x6b +#define ADCX140_DSP_CFG1 0x6c +#define ADCX140_DRE_CFG0 0x6d +#define ADCX140_IN_CH_EN 0x73 +#define ADCX140_ASI_OUT_CH_EN 0x74 +#define ADCX140_PWR_CFG 0x75 +#define ADCX140_DEV_STS0 0x76 +#define ADCX140_DEV_STS1 0x77 + +#define ADCX140_RESET BIT(0) + +#define ADCX140_WAKE_DEV BIT(0) +#define ADCX140_AREG_INTERNAL BIT(7) + +#define ADCX140_BCLKINV_BIT BIT(2) +#define ADCX140_FSYNCINV_BIT BIT(3) +#define ADCX140_INV_MSK (ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT) +#define ADCX140_BCLK_FSYNC_MASTER BIT(7) +#define ADCX140_I2S_MODE_BIT BIT(6) +#define ADCX140_LEFT_JUST_BIT BIT(7) +#define ADCX140_ASI_FORMAT_MSK (ADCX140_I2S_MODE_BIT | ADCX140_LEFT_JUST_BIT) + +#define ADCX140_16_BIT_WORD 0x0 +#define ADCX140_20_BIT_WORD BIT(4) +#define ADCX140_24_BIT_WORD BIT(5) +#define ADCX140_32_BIT_WORD (BIT(4) | BIT(5)) +#define ADCX140_WORD_LEN_MSK 0x30 + +#define ADCX140_MAX_CHANNELS 8 + +#define ADCX140_MIC_BIAS_VAL_VREF 0 +#define ADCX140_MIC_BIAS_VAL_VREF_1096 1 +#define ADCX140_MIC_BIAS_VAL_AVDD 6 +#define ADCX140_MIC_BIAS_VAL_MSK GENMASK(6, 4) + +#define ADCX140_MIC_BIAS_VREF_275V 0 +#define ADCX140_MIC_BIAS_VREF_25V 1 +#define ADCX140_MIC_BIAS_VREF_1375V 2 +#define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0) + +#define ADCX140_PWR_CFG_BIAS_PDZ BIT(7) +#define ADCX140_PWR_CFG_ADC_PDZ BIT(6) +#define ADCX140_PWR_CFG_PLL_PDZ BIT(5) + +#define ADCX140_TX_OFFSET_MASK GENMASK(4, 0) + +#endif /* _TLV320ADCX140_ */ From 62209c9ad2ac29431e91392afb0bd6332ae4c33e Mon Sep 17 00:00:00 2001 From: Martin Blumenstingl Date: Thu, 20 Feb 2020 21:57:09 +0100 Subject: [PATCH 159/415] ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings The AIU audio output controller on the Meson8 and Meson8b SoC families is compatible with the one found in the GXBB family. Document the compatible string for these two older SoCs. Signed-off-by: Martin Blumenstingl Reviewed-by: Jerome Brunet Link: https://lore.kernel.org/r/20200220205711.77953-2-martin.blumenstingl@googlemail.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/amlogic,aiu.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml index 3ef7632dcb5992..a61bccf915d8f3 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml @@ -21,6 +21,8 @@ properties: - enum: - amlogic,aiu-gxbb - amlogic,aiu-gxl + - amlogic,aiu-meson8 + - amlogic,aiu-meson8b - const: amlogic,aiu From edc761805302db6d63916694d0cdb7468864a47a Mon Sep 17 00:00:00 2001 From: Martin Blumenstingl Date: Thu, 20 Feb 2020 21:57:10 +0100 Subject: [PATCH 160/415] ASoC: meson: aiu: introduce a struct for platform specific information Introduce a struct aiu_platform_data to make the driver aware of platform specific information. Convert the existing check for the internal stereo audio codec (only available on GXL) to this new struct. Support for the 32-bit SoCs will need this as well because the AIU_CLK_CTRL_MORE register doesn't have the I2S divider bits (and we need to use the I2S divider from AIU_CLK_CTRL instead). Signed-off-by: Martin Blumenstingl Reviewed-by: Jerome Brunet Link: https://lore.kernel.org/r/20200220205711.77953-3-martin.blumenstingl@googlemail.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu.c | 19 ++++++++++++++++--- sound/soc/meson/aiu.h | 5 +++++ 2 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index d3e2d40e956258..38209312a8c31a 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -273,6 +273,11 @@ static int aiu_probe(struct platform_device *pdev) aiu = devm_kzalloc(dev, sizeof(*aiu), GFP_KERNEL); if (!aiu) return -ENOMEM; + + aiu->platform = device_get_match_data(dev); + if (!aiu->platform) + return -ENODEV; + platform_set_drvdata(pdev, aiu); ret = device_reset(dev); @@ -322,7 +327,7 @@ static int aiu_probe(struct platform_device *pdev) } /* Register the internal dac control component on gxl */ - if (of_device_is_compatible(dev->of_node, "amlogic,aiu-gxl")) { + if (aiu->platform->has_acodec) { ret = aiu_acodec_ctrl_register_component(dev); if (ret) { dev_err(dev, @@ -344,9 +349,17 @@ static int aiu_remove(struct platform_device *pdev) return 0; } +static const struct aiu_platform_data aiu_gxbb_pdata = { + .has_acodec = false, +}; + +static const struct aiu_platform_data aiu_gxl_pdata = { + .has_acodec = true, +}; + static const struct of_device_id aiu_of_match[] = { - { .compatible = "amlogic,aiu-gxbb", }, - { .compatible = "amlogic,aiu-gxl", }, + { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, + { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, {} }; MODULE_DEVICE_TABLE(of, aiu_of_match); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 06a968c557287b..ab003638d5e593 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -27,11 +27,16 @@ struct aiu_interface { int irq; }; +struct aiu_platform_data { + bool has_acodec; +}; + struct aiu { struct clk *pclk; struct clk *spdif_mclk; struct aiu_interface i2s; struct aiu_interface spdif; + const struct aiu_platform_data *platform; }; #define AIU_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ From 3e25c44598aa44134207ad7b3c5ad6b586135777 Mon Sep 17 00:00:00 2001 From: Martin Blumenstingl Date: Thu, 20 Feb 2020 21:57:11 +0100 Subject: [PATCH 161/415] ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families The AIU audio controller on the Meson8 and Meson8b SoC families is compatible with the one found in the later GXBB family. Add compatible strings for these two older SoC families so the driver can be loaded for them. Instead of using the I2S divider from the AIU_CLK_CTRL_MORE register we need to use the I2S divider from the AIU_CLK_CTRL register. This older register is less flexible because it only supports four divider settings (1, 2, 4, 8) compared to the AIU_CLK_CTRL_MORE register (which supports dividers in the range 0..64). Signed-off-by: Martin Blumenstingl Reviewed-by: Jerome Brunet Link: https://lore.kernel.org/r/20200220205711.77953-4-martin.blumenstingl@googlemail.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 2 +- sound/soc/meson/aiu-encoder-i2s.c | 92 +++++++++++++++++++++++-------- sound/soc/meson/aiu.c | 9 +++ sound/soc/meson/aiu.h | 1 + 4 files changed, 81 insertions(+), 23 deletions(-) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 897a706dcda01e..d27e9180b45377 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -10,7 +10,7 @@ config SND_MESON_AIU imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI help Select Y or M to add support for the Audio output subsystem found - in the Amlogic GX SoC family + in the Amlogic Meson8, Meson8b and GX SoC families config SND_MESON_AXG_FIFO tristate diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index 4900e38e7e494e..cc73b5d5c2b704 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -111,34 +111,40 @@ static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component, return 0; } -static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, - struct snd_pcm_hw_params *params) +static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, + struct snd_pcm_hw_params *params, + unsigned int bs) { - struct aiu *aiu = snd_soc_component_get_drvdata(component); - unsigned int srate = params_rate(params); - unsigned int fs, bs; - - /* Get the oversampling factor */ - fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); + switch (bs) { + case 1: + case 2: + case 4: + case 8: + /* These are the only valid legacy dividers */ + break; - if (fs % 64) + default: + dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); return -EINVAL; + }; - /* Send data MSB first */ - snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, - AIU_I2S_DAC_CFG_MSB_FIRST, - AIU_I2S_DAC_CFG_MSB_FIRST); + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, + __ffs(bs))); - /* Set bclk to lrlck ratio */ - snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, - AIU_CODEC_DAC_LRCLK_CTRL_DIV, - FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, - 64 - 1)); + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, + AIU_CLK_CTRL_MORE_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, + 0)); - /* Use CLK_MORE for mclk to bclk divider */ - snd_soc_component_update_bits(component, AIU_CLK_CTRL, - AIU_CLK_CTRL_I2S_DIV, 0); + return 0; +} +static int aiu_encoder_i2s_set_more_div(struct snd_soc_component *component, + struct snd_pcm_hw_params *params, + unsigned int bs) +{ /* * NOTE: this HW is odd. * In most configuration, the i2s divider is 'mclk / blck'. @@ -146,7 +152,6 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, * increased by 50% to get the correct output rate. * No idea why ! */ - bs = fs / 64; if (params_width(params) == 16 && params_channels(params) == 8) { if (bs % 2) { dev_err(component->dev, @@ -156,11 +161,54 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, bs += bs / 2; } + /* Use CLK_MORE for mclk to bclk divider */ + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, 0)); + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, AIU_CLK_CTRL_MORE_I2S_DIV, FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, bs - 1)); + return 0; +} + +static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(component); + unsigned int srate = params_rate(params); + unsigned int fs, bs; + int ret; + + /* Get the oversampling factor */ + fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); + + if (fs % 64) + return -EINVAL; + + /* Send data MSB first */ + snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, + AIU_I2S_DAC_CFG_MSB_FIRST, + AIU_I2S_DAC_CFG_MSB_FIRST); + + /* Set bclk to lrlck ratio */ + snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, + AIU_CODEC_DAC_LRCLK_CTRL_DIV, + FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, + 64 - 1)); + + bs = fs / 64; + + if (aiu->platform->has_clk_ctrl_more_i2s_div) + ret = aiu_encoder_i2s_set_more_div(component, params, bs); + else + ret = aiu_encoder_i2s_set_legacy_div(component, params, bs); + + if (ret) + return ret; + /* Make sure amclk is used for HDMI i2s as well */ snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, AIU_CLK_CTRL_MORE_HDMI_AMCLK, diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 38209312a8c31a..dc35ca79021c57 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -351,15 +351,24 @@ static int aiu_remove(struct platform_device *pdev) static const struct aiu_platform_data aiu_gxbb_pdata = { .has_acodec = false, + .has_clk_ctrl_more_i2s_div = true, }; static const struct aiu_platform_data aiu_gxl_pdata = { .has_acodec = true, + .has_clk_ctrl_more_i2s_div = true, +}; + +static const struct aiu_platform_data aiu_meson8_pdata = { + .has_acodec = false, + .has_clk_ctrl_more_i2s_div = false, }; static const struct of_device_id aiu_of_match[] = { { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, + { .compatible = "amlogic,aiu-meson8", .data = &aiu_meson8_pdata }, + { .compatible = "amlogic,aiu-meson8b", .data = &aiu_meson8_pdata }, {} }; MODULE_DEVICE_TABLE(of, aiu_of_match); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index ab003638d5e593..87aa19ac4af3af 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -29,6 +29,7 @@ struct aiu_interface { struct aiu_platform_data { bool has_acodec; + bool has_clk_ctrl_more_i2s_div; }; struct aiu { From 150cbf8e66ec86966c13fd7a0e3de8813bca03d8 Mon Sep 17 00:00:00 2001 From: Samuel Holland Date: Mon, 17 Feb 2020 00:42:20 -0600 Subject: [PATCH 162/415] ASoC: sun8i-codec: Remove unused dev from codec struct This field is not used anywhere in the driver, so remove it. Fixes: 36c684936fae ("ASoC: Add sun8i digital audio codec") Signed-off-by: Samuel Holland Acked-by: Chen-Yu Tsai Link: https://lore.kernel.org/r/20200217064250.15516-5-samuel@sholland.org Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 55798bc8eae29d..41471bd010424d 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -85,7 +85,6 @@ #define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK GENMASK(12, 9) struct sun8i_codec { - struct device *dev; struct regmap *regmap; struct clk *clk_module; struct clk *clk_bus; @@ -541,8 +540,6 @@ static int sun8i_codec_probe(struct platform_device *pdev) if (!scodec) return -ENOMEM; - scodec->dev = &pdev->dev; - scodec->clk_module = devm_clk_get(&pdev->dev, "mod"); if (IS_ERR(scodec->clk_module)) { dev_err(&pdev->dev, "Failed to get the module clock\n"); From bf726b1c86f2caab70ad614cdf7da3b81ad08e69 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 21 Feb 2020 06:41:51 -0600 Subject: [PATCH 163/415] ASoC: tas2562: Add support for digital volume control Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200221124151.8774-1-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2562.c | 78 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 6 ++- 2 files changed, 82 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b517ada7e8092a..561ac0ac079596 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,7 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -413,6 +432,50 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; + return 0; +} + +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -426,6 +489,17 @@ static const struct snd_kcontrol_new vsense_switch = static const struct snd_kcontrol_new tas2562_snd_controls[] = { SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -516,6 +590,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 6f55ebcf19ea7c..28e75fc431d0d8 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,8 +35,10 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) From bd56e593da19de22284951c33ce5c419258171bf Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 21 Feb 2020 16:36:05 +0100 Subject: [PATCH 164/415] ASoC: meson: g12a: add toacodec dt-binding documentation Add the DT bindings and documentation of the internal audio DAC glue found on Amlogic g12a and sm1 SoC families Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200221153607.1585499-2-jbrunet@baylibre.com Signed-off-by: Mark Brown --- .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 +++++++++++++++++++ .../dt-bindings/sound/meson-g12a-toacodec.h | 10 ++++ 2 files changed, 61 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml new file mode 100644 index 00000000000000..f778d3371fde67 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-toacodec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a Internal DAC Control Glue + +maintainers: + - Jerome Brunet + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 1 + + compatible: + oneOf: + - items: + - const: + amlogic,g12a-toacodec + - items: + - enum: + - amlogic,sm1-toacodec + - const: + amlogic,g12a-toacodec + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - resets + +examples: + - | + #include + + toacodec: audio-controller@740 { + compatible = "amlogic,g12a-toacodec"; + reg = <0x0 0x740 0x0 0x4>; + #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOACODEC>; + }; diff --git a/include/dt-bindings/sound/meson-g12a-toacodec.h b/include/dt-bindings/sound/meson-g12a-toacodec.h new file mode 100644 index 00000000000000..69d7a75592a2a6 --- /dev/null +++ b/include/dt-bindings/sound/meson-g12a-toacodec.h @@ -0,0 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_MESON_G12A_TOACODEC_H +#define __DT_MESON_G12A_TOACODEC_H + +#define TOACODEC_IN_A 0 +#define TOACODEC_IN_B 1 +#define TOACODEC_IN_C 2 +#define TOACODEC_OUT 3 + +#endif /* __DT_MESON_G12A_TOACODEC_H */ From af2618a2eee8531e134c42194143c2f4faef94ba Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 21 Feb 2020 16:36:06 +0100 Subject: [PATCH 165/415] ASoC: meson: g12a: add internal DAC glue driver Add support for the internal audio DAC glue found on the Amlogic g12a and sm1 SoC families. This allows to connect the TDM outputs of the SoC to the internal t9015 audio DAC. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200221153607.1585499-3-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 9 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/g12a-toacodec.c | 252 ++++++++++++++++++++++++++++++++ 3 files changed, 263 insertions(+) create mode 100644 sound/soc/meson/g12a-toacodec.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index d27e9180b45377..8b6295283989dc 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -109,6 +109,15 @@ config SND_MESON_GX_SOUND_CARD help Select Y or M to add support for the GXBB/GXL SoC sound card +config SND_MESON_G12A_TOACODEC + tristate "Amlogic G12A To Internal DAC Control Support" + select SND_MESON_CODEC_GLUE + select REGMAP_MMIO + imply SND_SOC_MESON_T9015 + help + Select Y or M to add support for the internal audio DAC on the + g12a SoC family + config SND_MESON_G12A_TOHDMITX tristate "Amlogic G12A To HDMI TX Control Support" select REGMAP_MMIO diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 3c9d4884681638..e446bc98048188 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -22,6 +22,7 @@ snd-soc-meson-axg-pdm-objs := axg-pdm.o snd-soc-meson-card-utils-objs := meson-card-utils.o snd-soc-meson-codec-glue-objs := meson-codec-glue.o snd-soc-meson-gx-sound-card-objs := gx-card.o +snd-soc-meson-g12a-toacodec-objs := g12a-toacodec.o snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o snd-soc-meson-t9015-objs := t9015.o @@ -40,5 +41,6 @@ obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o obj-$(CONFIG_SND_MESON_CARD_UTILS) += snd-soc-meson-card-utils.o obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o obj-$(CONFIG_SND_MESON_GX_SOUND_CARD) += snd-soc-meson-gx-sound-card.o +obj-$(CONFIG_SND_MESON_G12A_TOACODEC) += snd-soc-meson-g12a-toacodec.o obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o obj-$(CONFIG_SND_SOC_MESON_T9015) += snd-soc-meson-t9015.o diff --git a/sound/soc/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c new file mode 100644 index 00000000000000..9339fabccb7963 --- /dev/null +++ b/sound/soc/meson/g12a-toacodec.c @@ -0,0 +1,252 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "axg-tdm.h" +#include "meson-codec-glue.h" + +#define G12A_TOACODEC_DRV_NAME "g12a-toacodec" + +#define TOACODEC_CTRL0 0x0 +#define CTRL0_ENABLE_SHIFT 31 +#define CTRL0_DAT_SEL_SHIFT 14 +#define CTRL0_DAT_SEL (0x3 << CTRL0_DAT_SEL_SHIFT) +#define CTRL0_LANE_SEL 12 +#define CTRL0_LRCLK_SEL GENMASK(9, 8) +#define CTRL0_BLK_CAP_INV BIT(7) +#define CTRL0_BCLK_O_INV BIT(6) +#define CTRL0_BCLK_SEL GENMASK(5, 4) +#define CTRL0_MCLK_SEL GENMASK(2, 0) + +#define TOACODEC_OUT_CHMAX 2 + +static const char * const g12a_toacodec_mux_texts[] = { + "I2S A", "I2S B", "I2S C", +}; + +static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL0_DAT_SEL, + FIELD_PREP(CTRL0_DAT_SEL, mux)); + + if (!changed) + return 0; + + /* Force disconnect of the mux while updating */ + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + snd_soc_component_update_bits(component, e->reg, + CTRL0_DAT_SEL | + CTRL0_LRCLK_SEL | + CTRL0_BCLK_SEL, + FIELD_PREP(CTRL0_DAT_SEL, mux) | + FIELD_PREP(CTRL0_LRCLK_SEL, mux) | + FIELD_PREP(CTRL0_BCLK_SEL, mux)); + + /* + * FIXME: + * On this soc, the glue gets the MCLK directly from the clock + * controller instead of going the through the TDM interface. + * + * Here we assume interface A uses clock A, etc ... While it is + * true for now, it could be different. Instead the glue should + * find out the clock used by the interface and select the same + * source. For that, we will need regmap backed clock mux which + * is a work in progress + */ + snd_soc_component_update_bits(component, e->reg, + CTRL0_MCLK_SEL, + FIELD_PREP(CTRL0_MCLK_SEL, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + +static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0, + CTRL0_DAT_SEL_SHIFT, + g12a_toacodec_mux_texts); + +static const struct snd_kcontrol_new g12a_toacodec_mux = + SOC_DAPM_ENUM_EXT("Source", g12a_toacodec_mux_enum, + snd_soc_dapm_get_enum_double, + g12a_toacodec_mux_put_enum); + +static const struct snd_kcontrol_new g12a_toacodec_out_enable = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", TOACODEC_CTRL0, + CTRL0_ENABLE_SHIFT, 1, 0); + +static const struct snd_soc_dapm_widget g12a_toacodec_widgets[] = { + SND_SOC_DAPM_MUX("SRC", SND_SOC_NOPM, 0, 0, + &g12a_toacodec_mux), + SND_SOC_DAPM_SWITCH("OUT EN", SND_SOC_NOPM, 0, 0, + &g12a_toacodec_out_enable), +}; + +static int g12a_toacodec_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data; + int ret; + + ret = meson_codec_glue_input_hw_params(substream, params, dai); + if (ret) + return ret; + + /* The glue will provide 1 lane out of the 4 to the output */ + data = meson_codec_glue_input_get_data(dai); + data->params.channels_min = min_t(unsigned int, TOACODEC_OUT_CHMAX, + data->params.channels_min); + data->params.channels_max = min_t(unsigned int, TOACODEC_OUT_CHMAX, + data->params.channels_max); + + return 0; +} + +static const struct snd_soc_dai_ops g12a_toacodec_input_ops = { + .hw_params = g12a_toacodec_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, +}; + +static const struct snd_soc_dai_ops g12a_toacodec_output_ops = { + .startup = meson_codec_glue_output_startup, +}; + +#define TOACODEC_STREAM(xname, xsuffix, xchmax) \ +{ \ + .stream_name = xname " " xsuffix, \ + .channels_min = 1, \ + .channels_max = (xchmax), \ + .rate_min = 5512, \ + .rate_max = 192000, \ + .formats = AXG_TDM_FORMATS, \ +} + +#define TOACODEC_INPUT(xname, xid) { \ + .name = xname, \ + .id = (xid), \ + .playback = TOACODEC_STREAM(xname, "Playback", 8), \ + .ops = &g12a_toacodec_input_ops, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ +} + +#define TOACODEC_OUTPUT(xname, xid) { \ + .name = xname, \ + .id = (xid), \ + .capture = TOACODEC_STREAM(xname, "Capture", TOACODEC_OUT_CHMAX), \ + .ops = &g12a_toacodec_output_ops, \ +} + +static struct snd_soc_dai_driver g12a_toacodec_dai_drv[] = { + TOACODEC_INPUT("IN A", TOACODEC_IN_A), + TOACODEC_INPUT("IN B", TOACODEC_IN_B), + TOACODEC_INPUT("IN C", TOACODEC_IN_C), + TOACODEC_OUTPUT("OUT", TOACODEC_OUT), +}; + +static int g12a_toacodec_component_probe(struct snd_soc_component *c) +{ + /* Initialize the static clock parameters */ + return snd_soc_component_write(c, TOACODEC_CTRL0, + CTRL0_BLK_CAP_INV); +} + +static const struct snd_soc_dapm_route g12a_toacodec_routes[] = { + { "SRC", "I2S A", "IN A Playback" }, + { "SRC", "I2S B", "IN B Playback" }, + { "SRC", "I2S C", "IN C Playback" }, + { "OUT EN", "Switch", "SRC" }, + { "OUT Capture", NULL, "OUT EN" }, +}; + +static const struct snd_kcontrol_new g12a_toacodec_controls[] = { + SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL, 3, 0), +}; + +static const struct snd_soc_component_driver g12a_toacodec_component_drv = { + .probe = g12a_toacodec_component_probe, + .controls = g12a_toacodec_controls, + .num_controls = ARRAY_SIZE(g12a_toacodec_controls), + .dapm_widgets = g12a_toacodec_widgets, + .num_dapm_widgets = ARRAY_SIZE(g12a_toacodec_widgets), + .dapm_routes = g12a_toacodec_routes, + .num_dapm_routes = ARRAY_SIZE(g12a_toacodec_routes), + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config g12a_toacodec_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, +}; + +static const struct of_device_id g12a_toacodec_of_match[] = { + { .compatible = "amlogic,g12a-toacodec", }, + {} +}; +MODULE_DEVICE_TABLE(of, g12a_toacodec_of_match); + +static int g12a_toacodec_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + void __iomem *regs; + struct regmap *map; + int ret; + + ret = device_reset(dev); + if (ret) + return ret; + + regs = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + map = devm_regmap_init_mmio(dev, regs, &g12a_toacodec_regmap_cfg); + if (IS_ERR(map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(map)); + return PTR_ERR(map); + } + + return devm_snd_soc_register_component(dev, + &g12a_toacodec_component_drv, g12a_toacodec_dai_drv, + ARRAY_SIZE(g12a_toacodec_dai_drv)); +} + +static struct platform_driver g12a_toacodec_pdrv = { + .driver = { + .name = G12A_TOACODEC_DRV_NAME, + .of_match_table = g12a_toacodec_of_match, + }, + .probe = g12a_toacodec_probe, +}; +module_platform_driver(g12a_toacodec_pdrv); + +MODULE_AUTHOR("Jerome Brunet "); +MODULE_DESCRIPTION("Amlogic G12a To Internal DAC Codec Driver"); +MODULE_LICENSE("GPL v2"); From b38c4a8a0291c31a660cd77761202ebb18332fb7 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 21 Feb 2020 16:36:07 +0100 Subject: [PATCH 166/415] ASoC: meson: axg-card: add toacodec support Make sure the axg audio card driver recognise the dai_link as a codec-to-codec link if the cpu dai is the internal dac glue. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20200221153607.1585499-4-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 372dc696cc8e9b..48651631bdcfa3 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -303,7 +303,8 @@ static int axg_card_cpu_is_tdm_iface(struct device_node *np) static int axg_card_cpu_is_codec(struct device_node *np) { - return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx"); + return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx") || + of_device_is_compatible(np, DT_PREFIX "g12a-toacodec"); } static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, From 8a329dbd4a02dc4e4ff78b006c33676f867f2726 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 21 Feb 2020 12:13:57 -0600 Subject: [PATCH 167/415] ASoC: tlv320adcx140: Add DRE and AGC support The TLV320ADCx140 parts support Dynamic Range Enhancer (DRE) as defined in Section 8.3.2 of the data sheets. The DRE achieves a complete-channel dynamic range as high as 120 dB. At a system level, the DRE scheme enables far-field, high-fidelity recording of audio signals in very quiet environments and low-distortion recording in loud environments. There are 2 enables for DRE. The first is a global setting that enables the DRE engine in the device and the other enable is per channel. If the DRE is enabled globally then either DRE or AGC can be used per each configured channel. If global DRE is disabled then even setting the DRE enable bit in the channel config register will have no effect. Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200221181358.22526-1-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adcx140.c | 55 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 1 + 2 files changed, 56 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 8182c584de9ccd..105e51be6fe68a 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -108,6 +108,7 @@ static const struct reg_default adcx140_reg_defaults[] = { { ADCX140_DSP_CFG0, 0x01 }, { ADCX140_DSP_CFG1, 0x40 }, { ADCX140_DRE_CFG0, 0x7b }, + { ADCX140_AGC_CFG0, 0xe7 }, { ADCX140_IN_CH_EN, 0xf0 }, { ADCX140_ASI_OUT_CH_EN, 0x00 }, { ADCX140_PWR_CFG, 0x00 }, @@ -158,6 +159,16 @@ static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); /* ADC gain. From 0 to 42 dB in 1 dB steps */ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); +/* DRE Level. From -12 dB to -66 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_thresh_tlv, -6600, 100, 0); +/* DRE Max Gain. From 2 dB to 26 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_gain_tlv, 200, 200, 0); + +/* AGC Level. From -6 dB to -36 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_thresh_tlv, -3600, 200, 0); +/* AGC Max Gain. From 3 dB to 42 dB in 3 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_gain_tlv, 300, 300, 0); + static const char * const resistor_text[] = { "2.5 kOhm", "10 kOhm", "20 kOhm" }; @@ -281,6 +292,18 @@ static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch = static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch2_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH2_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch3_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH3_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0); + +static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0); + /* Output Mixer */ static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = { SOC_DAPM_SINGLE("Digital CH1 Switch", 0, 0, 0, 0), @@ -361,6 +384,18 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0, &adcx140_dapm_ch4_en_switch), + SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_dre_en_switch), + + SND_SOC_DAPM_SWITCH("CH1_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch1_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH2_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch2_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH3_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch3_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH4_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch4_dre_en_switch), + SND_SOC_DAPM_MUX("IN1 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, in1_resistor_controls), SND_SOC_DAPM_MUX("IN2 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, @@ -383,6 +418,16 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"DRE_ENABLE", "Switch", "CH1_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH2_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH3_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH4_DRE_EN"}, + + {"CH1_DRE_EN", "Switch", "CH1_ADC"}, + {"CH2_DRE_EN", "Switch", "CH2_ADC"}, + {"CH3_DRE_EN", "Switch", "CH3_ADC"}, + {"CH4_DRE_EN", "Switch", "CH4_ADC"}, + /* Mic input */ {"CH1_ADC", NULL, "MIC_GAIN_CTL_CH1"}, {"CH2_ADC", NULL, "MIC_GAIN_CTL_CH2"}, @@ -455,6 +500,16 @@ static const struct snd_kcontrol_new adcx140_snd_controls[] = { SOC_SINGLE_TLV("Analog CH4 Mic Gain Volume", ADCX140_CH1_CFG4, 2, 42, 0, adc_tlv), + SOC_SINGLE_TLV("DRE Threshold", ADCX140_DRE_CFG0, 4, 9, 0, + dre_thresh_tlv), + SOC_SINGLE_TLV("DRE Max Gain", ADCX140_DRE_CFG0, 0, 12, 0, + dre_gain_tlv), + + SOC_SINGLE_TLV("AGC Threshold", ADCX140_AGC_CFG0, 4, 15, 0, + agc_thresh_tlv), + SOC_SINGLE_TLV("AGC Max Gain", ADCX140_AGC_CFG0, 0, 13, 0, + agc_gain_tlv), + SOC_SINGLE_TLV("Digital CH1 Out Volume", ADCX140_CH1_CFG2, 0, 0xff, 0, dig_vol_tlv), SOC_SINGLE_TLV("Digital CH2 Out Volume", ADCX140_CH2_CFG2, diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index 66b1c3b33f1eed..6d055e55909e68 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -84,6 +84,7 @@ #define ADCX140_DSP_CFG0 0x6b #define ADCX140_DSP_CFG1 0x6c #define ADCX140_DRE_CFG0 0x6d +#define ADCX140_AGC_CFG0 0x70 #define ADCX140_IN_CH_EN 0x73 #define ADCX140_ASI_OUT_CH_EN 0x74 #define ADCX140_PWR_CFG 0x75 From 8101d76253f6d1032ca79e937e45b837cb4bf0e0 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 21 Feb 2020 12:13:58 -0600 Subject: [PATCH 168/415] ASoC: tlv320adcx140: Add decimation filter support Add decimation filter selection support. Per Section 8.3.6.7 the Digital Decimation Filter is selectable between a Linear Phase, Low Latency, and Ultra Low Latency filer. Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200221181358.22526-2-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adcx140.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 105e51be6fe68a..93a0cb8e662ccf 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -169,6 +169,17 @@ static DECLARE_TLV_DB_SCALE(agc_thresh_tlv, -3600, 200, 0); /* AGC Max Gain. From 3 dB to 42 dB in 3 dB steps */ static DECLARE_TLV_DB_SCALE(agc_gain_tlv, 300, 300, 0); +static const char * const decimation_filter_text[] = { + "Linear Phase", "Low Latency", "Ultra-low Latency" +}; + +static SOC_ENUM_SINGLE_DECL(decimation_filter_enum, ADCX140_DSP_CFG0, 4, + decimation_filter_text); + +static const struct snd_kcontrol_new decimation_filter_controls[] = { + SOC_DAPM_ENUM("Decimation Filter", decimation_filter_enum), +}; + static const char * const resistor_text[] = { "2.5 kOhm", "10 kOhm", "20 kOhm" }; @@ -404,6 +415,9 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { in3_resistor_controls), SND_SOC_DAPM_MUX("IN4 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, in4_resistor_controls), + + SND_SOC_DAPM_MUX("Decimation Filter", SND_SOC_NOPM, 0, 0, + decimation_filter_controls), }; static const struct snd_soc_dapm_route adcx140_audio_map[] = { @@ -418,6 +432,10 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"Decimation Filter", "Linear Phase", "DRE_ENABLE"}, + {"Decimation Filter", "Low Latency", "DRE_ENABLE"}, + {"Decimation Filter", "Ultra-low Latency", "DRE_ENABLE"}, + {"DRE_ENABLE", "Switch", "CH1_DRE_EN"}, {"DRE_ENABLE", "Switch", "CH2_DRE_EN"}, {"DRE_ENABLE", "Switch", "CH3_DRE_EN"}, From 5a309875787db47d69610e45f00a727ef9e62aa0 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 24 Feb 2020 12:25:37 +0100 Subject: [PATCH 169/415] ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout On i386 randconfig: sound/soc/codecs/wm9705.o: In function `wm9705_soc_resume': wm9705.c:(.text+0x128): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9712.o: In function `wm9712_soc_resume': wm9712.c:(.text+0x2d1): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9713.o: In function `wm9713_soc_resume': wm9713.c:(.text+0x820): undefined reference to `snd_ac97_reset' Fix this by adding the missing dependencies on SND_SOC_AC97_BUS. Reported-by: Randy Dunlap Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/20200224112537.14483-1-geert@linux-m68k.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e9d54e4576ce5..a7e89567edbe8b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1610,16 +1610,19 @@ config SND_SOC_WM9090 config SND_SOC_WM9705 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9712 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9713 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW From 1640c8df0bbac9e5156132e24c0f0a932c2b2865 Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Sun, 23 Feb 2020 01:01:54 +0800 Subject: [PATCH 170/415] ASoC: meson: aiu: fix semicolon.cocci warnings sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon Remove unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Fixes: 3e25c44598aa ("ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families") Signed-off-by: kbuild test robot CC: Martin Blumenstingl Link: https://lore.kernel.org/r/20200222170154.GA119396@e50d7db646c3 Signed-off-by: Mark Brown --- sound/soc/meson/aiu-encoder-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index cc73b5d5c2b704..832e22d275fe08 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -126,7 +126,7 @@ static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, default: dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); return -EINVAL; - }; + } snd_soc_component_update_bits(component, AIU_CLK_CTRL, AIU_CLK_CTRL_I2S_DIV, From acf253c11329caa6be6d2abc14dfc8c0ec83718a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:30 +0900 Subject: [PATCH 171/415] ASoC: soc-pcm: add snd_soc_dai_get_pcm_stream() DAI driver has playback/capture stream. OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. Because of this kind of implementation, ALSA SoC needs to have many verbose code. To solve this issue, this patch adds snd_soc_dai_get_pcm_stream() macro to get playback/capture stream pointer from stream. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87ftf7jcab.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 ++++++ sound/soc/soc-dai.c | 7 +----- sound/soc/soc-pcm.c | 49 ++++++++--------------------------------- 3 files changed, 17 insertions(+), 46 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 7481e468be3916..c1089194ddf167 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -352,6 +352,13 @@ struct snd_soc_dai { unsigned int started:1; }; +static inline struct snd_soc_pcm_stream * +snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) +{ + return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &dai->driver->playback : &dai->driver->capture; +} + static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, const struct snd_pcm_substream *ss) { diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 73a829393652ac..19142f6e533c8c 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -390,12 +390,7 @@ int snd_soc_dai_compress_new(struct snd_soc_dai *dai, */ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir) { - struct snd_soc_pcm_stream *stream; - - if (dir == SNDRV_PCM_STREAM_PLAYBACK) - stream = &dai->driver->playback; - else - stream = &dai->driver->capture; + struct snd_soc_pcm_stream *stream = snd_soc_dai_get_pcm_stream(dai, dir); /* If the codec specifies any channels at all, it supports the stream */ return stream->channels_min; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aff27c8599ef98..7cb445bb1b541b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -396,20 +396,16 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) struct snd_pcm_hardware *hw = &runtime->hw; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; unsigned int chan_min = 0, chan_max = UINT_MAX; unsigned int rate_min = 0, rate_max = UINT_MAX; unsigned int rates = UINT_MAX; u64 formats = ULLONG_MAX; + int stream = substream->stream; int i; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_stream = &cpu_dai_drv->playback; - else - cpu_stream = &cpu_dai_drv->capture; + cpu_stream = snd_soc_dai_get_pcm_stream(rtd->cpu_dai, stream); /* first calculate min/max only for CODECs in the DAI link */ for_each_rtd_codec_dai(rtd, i, codec_dai) { @@ -427,11 +423,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) substream->stream)) continue; - codec_dai_drv = codec_dai->driver; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &codec_dai_drv->playback; - else - codec_stream = &codec_dai_drv->capture; + codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream); + chan_min = max(chan_min, codec_stream->channels_min); chan_max = min(chan_max, codec_stream->channels_max); rate_min = max(rate_min, codec_stream->rate_min); @@ -1600,7 +1593,6 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; int i; @@ -1612,11 +1604,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, if (!snd_soc_dai_stream_valid(dai, stream)) continue; - codec_dai_drv = dai->driver; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &codec_dai_drv->playback; - else - codec_stream = &codec_dai_drv->capture; + codec_stream = snd_soc_dai_get_pcm_stream(dai, stream); *formats &= codec_stream->formats; } @@ -1641,15 +1629,10 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_stream = &cpu_dai_drv->playback; - else - cpu_stream = &cpu_dai_drv->capture; + cpu_stream = snd_soc_dai_get_pcm_stream(be->cpu_dai, stream); *channels_min = max(*channels_min, cpu_stream->channels_min); *channels_max = min(*channels_max, cpu_stream->channels_max); @@ -1659,12 +1642,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, * DAIs connected to a single CPU DAI, use CPU DAI's directly */ if (be->num_codecs == 1) { - codec_dai_drv = be->codec_dais[0]->driver; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &codec_dai_drv->playback; - else - codec_stream = &codec_dai_drv->capture; + codec_stream = snd_soc_dai_get_pcm_stream(be->codec_dais[0], stream); *channels_min = max(*channels_min, codec_stream->channels_min); @@ -1693,17 +1671,12 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; struct snd_soc_dai *dai; int i; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_stream = &cpu_dai_drv->playback; - else - cpu_stream = &cpu_dai_drv->capture; + cpu_stream = snd_soc_dai_get_pcm_stream(be->cpu_dai, stream); *rate_min = max(*rate_min, cpu_stream->rate_min); *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); @@ -1717,11 +1690,7 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, if (!snd_soc_dai_stream_valid(dai, stream)) continue; - codec_dai_drv = dai->driver; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &codec_dai_drv->playback; - else - codec_stream = &codec_dai_drv->capture; + codec_stream = snd_soc_dai_get_pcm_stream(dai, stream); *rate_min = max(*rate_min, codec_stream->rate_min); *rate_max = min_not_zero(*rate_max, From 57be92066f68e63bd4a72a65d45c3407c0cb552a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:36 +0900 Subject: [PATCH 172/415] ASoC: soc-pcm: cleanup soc_pcm_apply_msb() soc_pcm_apply_msb() apply msb for CPU/Codec, but, it has duplicate code. The difference is only SNDRV_PCM_STREAM_PLAYBACK and SNDRV_PCM_STEAM_CAPTURE. It is very verbose and duplicate code. This patch simplify code by using snd_soc_dai_get_pcm_stream(). Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87eeurjca6.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7cb445bb1b541b..6f56526bbb2622 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -363,29 +363,24 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; + struct snd_soc_pcm_stream *pcm_codec, *pcm_cpu; + int stream = substream->stream; int i; unsigned int bits = 0, cpu_bits; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->playback.sig_bits == 0) { - bits = 0; - break; - } - bits = max(codec_dai->driver->playback.sig_bits, bits); - } - cpu_bits = cpu_dai->driver->playback.sig_bits; - } else { - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->capture.sig_bits == 0) { - bits = 0; - break; - } - bits = max(codec_dai->driver->capture.sig_bits, bits); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + pcm_codec = snd_soc_dai_get_pcm_stream(codec_dai, stream); + + if (pcm_codec->sig_bits == 0) { + bits = 0; + break; } - cpu_bits = cpu_dai->driver->capture.sig_bits; + bits = max(pcm_codec->sig_bits, bits); } + pcm_cpu = snd_soc_dai_get_pcm_stream(cpu_dai, stream); + cpu_bits = pcm_cpu->sig_bits; + soc_pcm_set_msb(substream, bits); soc_pcm_set_msb(substream, cpu_bits); } From 0c01f6ca8e4cc1e5505bf4657cf77fbfaa7b0bc2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:41 +0900 Subject: [PATCH 173/415] ASoC: soc-pcm: add snd_soc_dai_get_widget() soc-pcm.c has dai_get_widget(), but it can be more generic. This patch renames it to snd_soc_dai_get_widget(), and use it. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87d0abjca1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 8 ++++++++ sound/soc/intel/skylake/skl-pcm.c | 10 ++-------- sound/soc/soc-dapm.c | 10 ++-------- sound/soc/soc-pcm.c | 17 ++++------------- 4 files changed, 16 insertions(+), 29 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index c1089194ddf167..92c382690930e4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -359,6 +359,14 @@ snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) &dai->driver->playback : &dai->driver->capture; } +static inline +struct snd_soc_dapm_widget *snd_soc_dai_get_widget( + struct snd_soc_dai *dai, int stream) +{ + return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dai->playback_widget : dai->capture_widget; +} + static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, const struct snd_pcm_substream *ss) { diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b99509675d29a2..05a9677c5a53f2 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -112,10 +112,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w; struct skl_dev *skl = bus_to_skl(bus); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + w = snd_soc_dai_get_widget(dai, substream->stream); if (w->ignore_suspend && enable) skl->supend_active++; @@ -475,10 +472,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, if (!mconfig) return -EIO; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + w = snd_soc_dai_get_widget(dai, substream->stream); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 69eff234b26fd2..f2e6788654803b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2620,10 +2620,7 @@ static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w; int ret; - if (dir == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + w = snd_soc_dai_get_widget(dai, dir); if (!w) return 0; @@ -4389,10 +4386,7 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget *w; unsigned int ep; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + w = snd_soc_dai_get_widget(dai, stream); if (w) { dapm_mark_dirty(w, "stream event"); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6f56526bbb2622..e183fabc5b6f2e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -82,15 +82,6 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd, return 0; } -static inline -struct snd_soc_dapm_widget *dai_get_widget(struct snd_soc_dai *dai, int stream) -{ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - return dai->playback_widget; - else - return dai->capture_widget; -} - static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { @@ -1242,7 +1233,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; - w = dai_get_widget(be->cpu_dai, stream); + w = snd_soc_dai_get_widget(be->cpu_dai, stream); dev_dbg(card->dev, "ASoC: try BE : %s\n", w ? w->name : "(not set)"); @@ -1251,7 +1242,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, return be; for_each_rtd_codec_dai(be, i, dai) { - w = dai_get_widget(dai, stream); + w = snd_soc_dai_get_widget(dai, stream); if (w == widget) return be; @@ -1326,7 +1317,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, unsigned int i; /* is there a valid CPU DAI widget for this BE */ - widget = dai_get_widget(dpcm->be->cpu_dai, stream); + widget = snd_soc_dai_get_widget(dpcm->be->cpu_dai, stream); /* prune the BE if it's no longer in our active list */ if (widget && widget_in_list(list, widget)) @@ -1335,7 +1326,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, /* is there a valid CODEC DAI widget for this BE */ do_prune = 1; for_each_rtd_codec_dai(dpcm->be, i, dai) { - widget = dai_get_widget(dai, stream); + widget = snd_soc_dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ if (widget && widget_in_list(list, widget)) From 580dff3636d08ed12cb5d5db2fd895cbeffd0fd5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:46 +0900 Subject: [PATCH 174/415] ASoC: soc-pcm: merge dpcm_run_new/old_update() into dpcm_fe_runtime_update() soc-pcm has dpcm_run_new/old_update(), but these are used from dpcm_fe_runtime_update() only, and are very verbose functions. This patch disassembles these. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87blpvjc9v.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 35 +++++++---------------------------- 1 file changed, 7 insertions(+), 28 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e183fabc5b6f2e..1bf2db1732bf50 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2536,37 +2536,12 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream) -{ - int ret; - - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); - ret = dpcm_run_update_startup(fe, stream); - if (ret < 0) - dev_err(fe->dev, "ASoC: failed to startup some BEs\n"); - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - - return ret; -} - -static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) -{ - int ret; - - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); - ret = dpcm_run_update_shutdown(fe, stream); - if (ret < 0) - dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n"); - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - - return ret; -} - static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { struct snd_soc_dapm_widget_list *list; int stream; int count, paths; + int ret; if (!fe->dai_link->dynamic) return 0; @@ -2603,10 +2578,14 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) /* update any playback/capture paths */ count = dpcm_process_paths(fe, stream, &list, new); if (count) { + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); if (new) - dpcm_run_new_update(fe, stream); + ret = dpcm_run_update_startup(fe, stream); else - dpcm_run_old_update(fe, stream); + ret = dpcm_run_update_shutdown(fe, stream); + if (ret < 0) + dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n"); + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); dpcm_clear_pending_state(fe, stream); dpcm_be_disconnect(fe, stream); From 52645e332d227a3d3cd345e97a10d99b7e80fae4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:52 +0900 Subject: [PATCH 175/415] ASoC: soc-pcm: move dpcm_path_put() to soc-pcm.c dpcm_path_put() (A) is calling kfree(*list). The freed list is created by dapm_widget_list_create() (B) which is called from snd_soc_dapm_dai_get_connected_widgets() (C) which is called from dpcm_path_get() (D). (B) dapm_widget_list_create(**list, ...) { ... => *list = kzalloc(); ... } (C) snd_soc_dapm_dai_get_connected_widgets(..., **list, ...) { ... dapm_widget_list_create(list, ...); ... } (D) dpcm_path_get(..., **list) { ... snd_soc_dapm_dai_get_connected_widgets(..., list, ...); ... } (A) dpcm_path_put(**list) { => kfree(*list); } This kind of unbalance code is very difficult to read/understand. To avoid this issue, this patch adds each missing paired function dapm_widget_list_free() for dapm_widget_list_create() (B), and snd_soc_dapm_dai_free_widgets() for snd_soc_dapm_dai_get_connected_widgets() (C). This patch uses these, and moves dpcm_path_put() next to dpcm_path_get(). Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87a75fjc9q.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + include/sound/soc-dpcm.h | 7 +------ sound/soc/soc-dapm.c | 10 ++++++++++ sound/soc/soc-pcm.c | 5 +++++ 4 files changed, 17 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9439e75945f6ce..464b20acd7204f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -484,6 +484,7 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list, bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, enum snd_soc_dapm_direction)); +void snd_soc_dapm_dai_free_widgets(struct snd_soc_dapm_widget_list **list); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 3e7819d2a6aab4..40223577ec4a70 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -145,6 +145,7 @@ static inline void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_); +void dpcm_path_put(struct snd_soc_dapm_widget_list **list); int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new); int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); @@ -158,10 +159,4 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event); -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - - #endif diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f2e6788654803b..8a7d700a0fda3b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1105,6 +1105,11 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +static void dapm_widget_list_free(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, struct list_head *widgets) { @@ -1310,6 +1315,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, return paths; } +void snd_soc_dapm_dai_free_widgets(struct snd_soc_dapm_widget_list **list) +{ + dapm_widget_list_free(list); +} + /* * Handler for regulator supply widget. */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1bf2db1732bf50..3b3b32923783f1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1302,6 +1302,11 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } +void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + snd_soc_dapm_dai_free_widgets(list); +} + static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { From c3212829f812a4ac0c6078978c109c6f1ff882c2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:56:57 +0900 Subject: [PATCH 176/415] ASoC: soc-pcm: move CONFIG_DEBUG_FS functions to top side This is prepare for CONFIG_DEBUG_FS cleanup Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/878skzjc9k.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 276 ++++++++++++++++++++++---------------------- 1 file changed, 138 insertions(+), 138 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3b3b32923783f1..fc98ab87fa45ab 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -28,6 +28,144 @@ #define DPCM_MAX_BE_USERS 8 +#ifdef CONFIG_DEBUG_FS +static const char *dpcm_state_string(enum snd_soc_dpcm_state state) +{ + switch (state) { + case SND_SOC_DPCM_STATE_NEW: + return "new"; + case SND_SOC_DPCM_STATE_OPEN: + return "open"; + case SND_SOC_DPCM_STATE_HW_PARAMS: + return "hw_params"; + case SND_SOC_DPCM_STATE_PREPARE: + return "prepare"; + case SND_SOC_DPCM_STATE_START: + return "start"; + case SND_SOC_DPCM_STATE_STOP: + return "stop"; + case SND_SOC_DPCM_STATE_SUSPEND: + return "suspend"; + case SND_SOC_DPCM_STATE_PAUSED: + return "paused"; + case SND_SOC_DPCM_STATE_HW_FREE: + return "hw_free"; + case SND_SOC_DPCM_STATE_CLOSE: + return "close"; + } + + return "unknown"; +} + +static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, + int stream, char *buf, size_t size) +{ + struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; + struct snd_soc_dpcm *dpcm; + ssize_t offset = 0; + unsigned long flags; + + /* FE state */ + offset += snprintf(buf + offset, size - offset, + "[%s - %s]\n", fe->dai_link->name, + stream ? "Capture" : "Playback"); + + offset += snprintf(buf + offset, size - offset, "State: %s\n", + dpcm_state_string(fe->dpcm[stream].state)); + + if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && + (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) + offset += snprintf(buf + offset, size - offset, + "Hardware Params: " + "Format = %s, Channels = %d, Rate = %d\n", + snd_pcm_format_name(params_format(params)), + params_channels(params), + params_rate(params)); + + /* BEs state */ + offset += snprintf(buf + offset, size - offset, "Backends:\n"); + + if (list_empty(&fe->dpcm[stream].be_clients)) { + offset += snprintf(buf + offset, size - offset, + " No active DSP links\n"); + goto out; + } + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + for_each_dpcm_be(fe, stream, dpcm) { + struct snd_soc_pcm_runtime *be = dpcm->be; + params = &dpcm->hw_params; + + offset += snprintf(buf + offset, size - offset, + "- %s\n", be->dai_link->name); + + offset += snprintf(buf + offset, size - offset, + " State: %s\n", + dpcm_state_string(be->dpcm[stream].state)); + + if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) + offset += snprintf(buf + offset, size - offset, + " Hardware Params: " + "Format = %s, Channels = %d, Rate = %d\n", + snd_pcm_format_name(params_format(params)), + params_channels(params), + params_rate(params)); + } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); +out: + return offset; +} + +static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_pcm_runtime *fe = file->private_data; + ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0; + int stream; + char *buf; + + buf = kmalloc(out_count, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + for_each_pcm_streams(stream) + if (snd_soc_dai_stream_valid(fe->cpu_dai, stream)) + offset += dpcm_show_state(fe, stream, + buf + offset, + out_count - offset); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); + + kfree(buf); + return ret; +} + +static const struct file_operations dpcm_state_fops = { + .open = simple_open, + .read = dpcm_state_read_file, + .llseek = default_llseek, +}; + +void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) +{ + if (!rtd->dai_link) + return; + + if (!rtd->dai_link->dynamic) + return; + + if (!rtd->card->debugfs_card_root) + return; + + rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, + rtd->card->debugfs_card_root); + + debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, + rtd, &dpcm_state_fops); +} +#endif + static int soc_rtd_startup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream) { @@ -2930,141 +3068,3 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); - -#ifdef CONFIG_DEBUG_FS -static const char *dpcm_state_string(enum snd_soc_dpcm_state state) -{ - switch (state) { - case SND_SOC_DPCM_STATE_NEW: - return "new"; - case SND_SOC_DPCM_STATE_OPEN: - return "open"; - case SND_SOC_DPCM_STATE_HW_PARAMS: - return "hw_params"; - case SND_SOC_DPCM_STATE_PREPARE: - return "prepare"; - case SND_SOC_DPCM_STATE_START: - return "start"; - case SND_SOC_DPCM_STATE_STOP: - return "stop"; - case SND_SOC_DPCM_STATE_SUSPEND: - return "suspend"; - case SND_SOC_DPCM_STATE_PAUSED: - return "paused"; - case SND_SOC_DPCM_STATE_HW_FREE: - return "hw_free"; - case SND_SOC_DPCM_STATE_CLOSE: - return "close"; - } - - return "unknown"; -} - -static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, - int stream, char *buf, size_t size) -{ - struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; - struct snd_soc_dpcm *dpcm; - ssize_t offset = 0; - unsigned long flags; - - /* FE state */ - offset += snprintf(buf + offset, size - offset, - "[%s - %s]\n", fe->dai_link->name, - stream ? "Capture" : "Playback"); - - offset += snprintf(buf + offset, size - offset, "State: %s\n", - dpcm_state_string(fe->dpcm[stream].state)); - - if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && - (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, - "Hardware Params: " - "Format = %s, Channels = %d, Rate = %d\n", - snd_pcm_format_name(params_format(params)), - params_channels(params), - params_rate(params)); - - /* BEs state */ - offset += snprintf(buf + offset, size - offset, "Backends:\n"); - - if (list_empty(&fe->dpcm[stream].be_clients)) { - offset += snprintf(buf + offset, size - offset, - " No active DSP links\n"); - goto out; - } - - spin_lock_irqsave(&fe->card->dpcm_lock, flags); - for_each_dpcm_be(fe, stream, dpcm) { - struct snd_soc_pcm_runtime *be = dpcm->be; - params = &dpcm->hw_params; - - offset += snprintf(buf + offset, size - offset, - "- %s\n", be->dai_link->name); - - offset += snprintf(buf + offset, size - offset, - " State: %s\n", - dpcm_state_string(be->dpcm[stream].state)); - - if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && - (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, - " Hardware Params: " - "Format = %s, Channels = %d, Rate = %d\n", - snd_pcm_format_name(params_format(params)), - params_channels(params), - params_rate(params)); - } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); -out: - return offset; -} - -static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) -{ - struct snd_soc_pcm_runtime *fe = file->private_data; - ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0; - int stream; - char *buf; - - buf = kmalloc(out_count, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - for_each_pcm_streams(stream) - if (snd_soc_dai_stream_valid(fe->cpu_dai, stream)) - offset += dpcm_show_state(fe, stream, - buf + offset, - out_count - offset); - - ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); - - kfree(buf); - return ret; -} - -static const struct file_operations dpcm_state_fops = { - .open = simple_open, - .read = dpcm_state_read_file, - .llseek = default_llseek, -}; - -void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) -{ - if (!rtd->dai_link) - return; - - if (!rtd->dai_link->dynamic) - return; - - if (!rtd->card->debugfs_card_root) - return; - - rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, - rtd->card->debugfs_card_root); - - debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, - rtd, &dpcm_state_fops); -} -#endif From 154dae87e73faa6d56265f22cae16dcdcea3dbb0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Feb 2020 15:57:06 +0900 Subject: [PATCH 177/415] ASoC: soc-pcm: add dpcm_create/remove_debugfs_state() soc-pcm.c has implementation which depends on CONFIG_DEBUG_FS. But, we don't want to have random #ifdef. This patch adds dpcm_create/remove_debugfs_state() and care it. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/877e0jjc9b.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 51 ++++++++++++++++++++++++++++++--------------- 1 file changed, 34 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fc98ab87fa45ab..c5cfd88720c21a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -164,6 +164,36 @@ void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, rtd, &dpcm_state_fops); } + +static void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm, int stream) +{ + char *name; + + name = kasprintf(GFP_KERNEL, "%s:%s", dpcm->be->dai_link->name, + stream ? "capture" : "playback"); + if (name) { + dpcm->debugfs_state = debugfs_create_dir( + name, dpcm->fe->debugfs_dpcm_root); + debugfs_create_u32("state", 0644, dpcm->debugfs_state, + &dpcm->state); + kfree(name); + } +} + +static void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm) +{ + debugfs_remove_recursive(dpcm->debugfs_state); +} + +#else +static inline void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm, + int stream) +{ +} + +static inline void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm) +{ +} #endif static int soc_rtd_startup(struct snd_soc_pcm_runtime *rtd, @@ -1254,9 +1284,6 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; -#ifdef CONFIG_DEBUG_FS - char *name; -#endif /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1281,17 +1308,8 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); -#ifdef CONFIG_DEBUG_FS - name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name, - stream ? "capture" : "playback"); - if (name) { - dpcm->debugfs_state = debugfs_create_dir(name, - fe->debugfs_dpcm_root); - debugfs_create_u32("state", 0644, dpcm->debugfs_state, - &dpcm->state); - kfree(name); - } -#endif + dpcm_create_debugfs_state(dpcm, stream); + return 1; } @@ -1344,9 +1362,8 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) /* BEs still alive need new FE */ dpcm_be_reparent(fe, dpcm->be, stream); -#ifdef CONFIG_DEBUG_FS - debugfs_remove_recursive(dpcm->debugfs_state); -#endif + dpcm_remove_debugfs_state(dpcm); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); From d2aaa8d8bfba93237ac944ee058fb98e2c2ef983 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 20 Feb 2020 11:49:55 +0200 Subject: [PATCH 178/415] ASoC: soc-pcm: fix state tracking error in snd_soc_component_open/close() ASoC component open/close and snd_soc_component_module_get/put are called independently for each component-substream pair, so the logic added in commit dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") was not sufficient and led to PCM playback and module unload errors. Implement handling of failures directly in soc_pcm_components_open(), so that any successfully opened components are closed upon error with other components. This allows to clean up error handling in soc_pcm_open() without adding more state tracking. Fixes: dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") Signed-off-by: Kai Vehmanen Tested-by: Dmitry Osipenko Link: https://lore.kernel.org/r/20200220094955.16968-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 7 ++----- sound/soc/soc-component.c | 35 +++++++---------------------------- sound/soc/soc-pcm.c | 27 +++++++++++++++++++++------ 3 files changed, 30 insertions(+), 39 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 1866ecc8e94b25..154d02fbbfed69 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -147,6 +147,8 @@ struct snd_soc_component { unsigned int active; + unsigned int suspended:1; /* is in suspend PM state */ + struct list_head list; struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; @@ -180,11 +182,6 @@ struct snd_soc_component { struct dentry *debugfs_root; const char *debugfs_prefix; #endif - - /* bit field */ - unsigned int suspended:1; /* is in suspend PM state */ - unsigned int opened:1; - unsigned int module:1; }; #define for_each_component_dais(component, dai)\ diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index ee00c09df5e791..14e175cdeeb84b 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -297,55 +297,34 @@ EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); int snd_soc_component_module_get(struct snd_soc_component *component, int upon_open) { - if (component->module) - return 0; - if (component->driver->module_get_upon_open == !!upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; - component->module = 1; - return 0; } void snd_soc_component_module_put(struct snd_soc_component *component, int upon_open) { - if (component->module && - component->driver->module_get_upon_open == !!upon_open) + if (component->driver->module_get_upon_open == !!upon_open) module_put(component->dev->driver->owner); - - component->module = 0; } int snd_soc_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (!component->opened && - component->driver->open) - ret = component->driver->open(component, substream); - - if (ret == 0) - component->opened = 1; - - return ret; + if (component->driver->open) + return component->driver->open(component, substream); + return 0; } int snd_soc_component_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (component->opened && - component->driver->close) - ret = component->driver->close(component, substream); - - component->opened = 0; - - return ret; + if (component->driver->close) + return component->driver->close(component, substream); + return 0; } int snd_soc_component_prepare(struct snd_soc_component *component, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aff27c8599ef98..235baeb2d56ab6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -469,28 +469,43 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) static int soc_pcm_components_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *last = NULL; struct snd_soc_component *component; int i, ret = 0; for_each_rtd_components(rtd, i, component) { + last = component; + ret = snd_soc_component_module_get_when_open(component); if (ret < 0) { dev_err(component->dev, "ASoC: can't get module %s\n", component->name); - return ret; + break; } ret = snd_soc_component_open(component, substream); if (ret < 0) { + snd_soc_component_module_put_when_close(component); dev_err(component->dev, "ASoC: can't open component %s: %d\n", component->name, ret); - return ret; + break; } } - return 0; + if (ret < 0) { + /* rollback on error */ + for_each_rtd_components(rtd, i, component) { + if (component == last) + break; + + snd_soc_component_close(component, substream); + snd_soc_component_module_put_when_close(component); + } + } + + return ret; } static int soc_pcm_components_close(struct snd_pcm_substream *substream) @@ -585,7 +600,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) { pr_err("ASoC: %s startup failed: %d\n", rtd->dai_link->name, ret); - goto component_err; + goto rtd_startup_err; } /* startup the audio subsystem */ @@ -681,9 +696,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_dai_shutdown(cpu_dai, substream); soc_rtd_shutdown(rtd, substream); -component_err: +rtd_startup_err: soc_pcm_components_close(substream); - +component_err: mutex_unlock(&rtd->card->pcm_mutex); for_each_rtd_components(rtd, i, component) { From f5e056e1e46fcbb5f74ce560792aeb7d57ce79e6 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 19 Nov 2019 11:36:40 +0000 Subject: [PATCH 179/415] ASoC: Intel: mrfld: fix incorrect check on p->sink The check on p->sink looks bogus, I believe it should be p->source since the following code blocks are related to p->source. Fix this by replacing p->sink with p->source. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Colin Ian King Addresses-Coverity: ("Copy-paste error") Link: https://lore.kernel.org/r/20191119113640.166940-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f196..f883c9340eeee6 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1333,7 +1333,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dai->capture_widget->name); w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect && p->source->power && From 76afa64374a79c22b2dab61aebef99a967783bf0 Mon Sep 17 00:00:00 2001 From: Shreyas NC Date: Tue, 25 Feb 2020 21:39:12 +0800 Subject: [PATCH 180/415] ASoC: Add initial support for multiple CPU DAIs ASoC core supports multiple codec DAIs but supports only a CPU DAI. To support multiple cpu DAIs, add cpu_dai and num_cpu_dai in snd_soc_dai_link and snd_soc_pcm_runtime structures similar to support for codec_dai. This is intended as a preparatory patch to eventually support the unification of the Codec and CPU DAI. Inline with multiple codec DAI approach, add support to allocate, init, bind and probe multiple cpu_dai on init if driver specifies that. Also add support to loop over multiple cpu_dai during suspend and resume. This is intended as a preparatory patch to eventually unify the CPU and Codec DAI into DAI components. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Signed-off-by: Kuninori Morimoto Signed-off-by: Shreyas NC Link: https://lore.kernel.org/r/20200225133917.21314-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc.h | 15 ++++ sound/soc/soc-core.c | 168 +++++++++++++++++++++++-------------------- 2 files changed, 106 insertions(+), 77 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8a2266676b2dd2..81e5d17be9351d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -855,6 +855,11 @@ struct snd_soc_dai_link { ((platform) = &link->platforms[i]); \ (i)++) +#define for_each_link_cpus(link, i, cpu) \ + for ((i) = 0; \ + ((i) < link->num_cpus) && ((cpu) = &link->cpus[i]); \ + (i)++) + /* * Sample 1 : Single CPU/Codec/Platform * @@ -1132,6 +1137,9 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai **codec_dais; unsigned int num_codecs; + struct snd_soc_dai **cpu_dais; + unsigned int num_cpus; + struct delayed_work delayed_work; void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_DEBUG_FS @@ -1159,6 +1167,13 @@ struct snd_soc_pcm_runtime { #define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) +#define for_each_rtd_cpu_dai(rtd, i, dai)\ + for ((i) = 0; \ + ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]); \ + (i)++) +#define for_each_rtd_cpu_dai_rollback(rtd, i, dai) \ + for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);) + void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); /* mixer control */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 518b652cf872d0..f2cfbf182f496c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -483,6 +483,14 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( if (!rtd->codec_dais) goto free_rtd; + /* + * for rtd->cpu_dais + */ + rtd->cpu_dais = devm_kcalloc(dev, dai_link->num_cpus, + sizeof(struct snd_soc_dai *), + GFP_KERNEL); + if (!rtd->cpu_dais) + goto free_rtd; /* * rtd remaining settings */ @@ -833,7 +841,7 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, struct snd_soc_dai_link *link) { int i; - struct snd_soc_dai_link_component *codec, *platform; + struct snd_soc_dai_link_component *cpu, *codec, *platform; for_each_link_codecs(link, i, codec) { /* @@ -882,44 +890,38 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, return -EPROBE_DEFER; } - /* FIXME */ - if (link->num_cpus > 1) { - dev_err(card->dev, - "ASoC: multi cpu is not yet supported %s\n", - link->name); - return -EINVAL; - } - - /* - * CPU device may be specified by either name or OF node, but - * can be left unspecified, and will be matched based on DAI - * name alone.. - */ - if (link->cpus->name && link->cpus->of_node) { - dev_err(card->dev, - "ASoC: Neither/both cpu name/of_node are set for %s\n", - link->name); - return -EINVAL; - } + for_each_link_cpus(link, i, cpu) { + /* + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (cpu->name && cpu->of_node) { + dev_err(card->dev, + "ASoC: Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } - /* - * Defer card registration if cpu dai component is not added to - * component list. - */ - if ((link->cpus->of_node || link->cpus->name) && - !soc_find_component(link->cpus)) - return -EPROBE_DEFER; + /* + * Defer card registration if cpu dai component is not added to + * component list. + */ + if ((cpu->of_node || cpu->name) && + !soc_find_component(cpu)) + return -EPROBE_DEFER; - /* - * At least one of CPU DAI name or CPU device name/node must be - * specified - */ - if (!link->cpus->dai_name && - !(link->cpus->name || link->cpus->of_node)) { - dev_err(card->dev, - "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", - link->name); - return -EINVAL; + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified + */ + if (!cpu->dai_name && + !(cpu->name || cpu->of_node)) { + dev_err(card->dev, + "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", + link->name); + return -EINVAL; + } } return 0; @@ -962,7 +964,7 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai_link_component *codec, *platform; + struct snd_soc_dai_link_component *codec, *platform, *cpu; struct snd_soc_component *component; int i, ret; @@ -987,14 +989,19 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, if (!rtd) return -ENOMEM; - /* FIXME: we need multi CPU support in the future */ - rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); - if (!rtd->cpu_dai) { - dev_info(card->dev, "ASoC: CPU DAI %s not registered\n", - dai_link->cpus->dai_name); - goto _err_defer; + rtd->num_cpus = dai_link->num_cpus; + for_each_link_cpus(dai_link, i, cpu) { + rtd->cpu_dais[i] = snd_soc_find_dai(cpu); + if (!rtd->cpu_dais[i]) { + dev_info(card->dev, "ASoC: CPU DAI %s not registered\n", + cpu->dai_name); + goto _err_defer; + } + snd_soc_rtd_add_component(rtd, rtd->cpu_dais[i]->component); } - snd_soc_rtd_add_component(rtd, rtd->cpu_dai->component); + + /* Single cpu links expect cpu and cpu_dai in runtime data */ + rtd->cpu_dai = rtd->cpu_dais[0]; /* Find CODEC from registered CODECs */ rtd->num_codecs = dai_link->num_codecs; @@ -1114,7 +1121,8 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, dai_link->stream_name, ret); return ret; } - ret = soc_dai_pcm_new(&cpu_dai, 1, rtd); + ret = soc_dai_pcm_new(rtd->cpu_dais, + rtd->num_cpus, rtd); if (ret < 0) return ret; ret = soc_dai_pcm_new(rtd->codec_dais, @@ -1306,6 +1314,7 @@ static void soc_remove_link_dais(struct snd_soc_card *card) { int i; struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_pcm_runtime *rtd; int order; @@ -1315,14 +1324,15 @@ static void soc_remove_link_dais(struct snd_soc_card *card) for_each_rtd_codec_dai(rtd, i, codec_dai) soc_remove_dai(codec_dai, order); - soc_remove_dai(rtd->cpu_dai, order); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + soc_remove_dai(cpu_dai, order); } } } static int soc_probe_link_dais(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *codec_dai, *cpu_dai; struct snd_soc_pcm_runtime *rtd; int i, order, ret; @@ -1333,9 +1343,12 @@ static int soc_probe_link_dais(struct snd_soc_card *card) "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); - ret = soc_probe_dai(rtd->cpu_dai, order); - if (ret) - return ret; + /* probe the CPU DAI */ + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + ret = soc_probe_dai(cpu_dai, order); + if (ret) + return ret; + } /* probe the CODEC DAI */ for_each_rtd_codec_dai(rtd, i, codec_dai) { @@ -1467,8 +1480,9 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; + unsigned int inv_dai_fmt; unsigned int i; int ret; @@ -1485,33 +1499,33 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, * Flip the polarity for the "CPU" end of a CODEC<->CODEC link * the component which has non_legacy_dai_naming is Codec */ - if (cpu_dai->component->driver->non_legacy_dai_naming) { - unsigned int inv_dai_fmt; - - inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; - switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - case SND_SOC_DAIFMT_CBM_CFS: - inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - case SND_SOC_DAIFMT_CBS_CFM: - inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - case SND_SOC_DAIFMT_CBS_CFS: - inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - } - - dai_fmt = inv_dai_fmt; + inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; + switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + break; } + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + unsigned int fmt = dai_fmt; - ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) { - dev_warn(cpu_dai->dev, - "ASoC: Failed to set DAI format: %d\n", ret); - return ret; + if (cpu_dai->component->driver->non_legacy_dai_naming) + fmt = inv_dai_fmt; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret != 0 && ret != -ENOTSUPP) { + dev_warn(cpu_dai->dev, + "ASoC: Failed to set DAI format: %d\n", ret); + return ret; + } } return 0; From 19bdcc7aeed4169820be6a683c422fc06d030136 Mon Sep 17 00:00:00 2001 From: Shreyas NC Date: Tue, 25 Feb 2020 21:39:13 +0800 Subject: [PATCH 181/415] ASoC: Add multiple CPU DAI support for PCM ops Add support in PCM operations to invoke multiple cpu dais as we do for multiple codec dais. Also the symmetry calculations are updated to reflect multiple cpu dais. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Signed-off-by: Kuninori Morimoto Signed-off-by: Shreyas NC Signed-off-by: Vinod Koul Link: https://lore.kernel.org/r/20200225133917.21314-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 396 +++++++++++++++++++++++++++++--------------- 1 file changed, 261 insertions(+), 135 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4c27c77206f118..44694e65fc4ad7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -253,18 +253,22 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd, static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i; lockdep_assert_held(&rtd->card->pcm_mutex); - cpu_dai->stream_active[stream] += action; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + cpu_dai->stream_active[stream] += action; + for_each_rtd_codec_dai(rtd, i, codec_dai) codec_dai->stream_active[stream] += action; - cpu_dai->active += action; - cpu_dai->component->active += action; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + cpu_dai->active += action; + cpu_dai->component->active += action; + } for_each_rtd_codec_dai(rtd, i, codec_dai) { codec_dai->active += action; codec_dai->component->active += action; @@ -434,7 +438,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; unsigned int rate, channels, sample_bits, symmetry, i; @@ -443,40 +447,60 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, sample_bits = snd_pcm_format_physical_width(params_format(params)); /* reject unmatched parameters when applying symmetry */ - symmetry = cpu_dai->driver->symmetric_rates || - rtd->dai_link->symmetric_rates; + symmetry = rtd->dai_link->symmetric_rates; + + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + symmetry |= cpu_dai->driver->symmetric_rates; for_each_rtd_codec_dai(rtd, i, codec_dai) symmetry |= codec_dai->driver->symmetric_rates; - if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { - dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", - cpu_dai->rate, rate); - return -EINVAL; + if (symmetry) { + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + if (cpu_dai->rate && cpu_dai->rate != rate) { + dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", + cpu_dai->rate, rate); + return -EINVAL; + } + } } - symmetry = cpu_dai->driver->symmetric_channels || - rtd->dai_link->symmetric_channels; + symmetry = rtd->dai_link->symmetric_channels; + + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + symmetry |= cpu_dai->driver->symmetric_channels; for_each_rtd_codec_dai(rtd, i, codec_dai) symmetry |= codec_dai->driver->symmetric_channels; - if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) { - dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", - cpu_dai->channels, channels); - return -EINVAL; + if (symmetry) { + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + if (cpu_dai->channels && + cpu_dai->channels != channels) { + dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", + cpu_dai->channels, channels); + return -EINVAL; + } + } } - symmetry = cpu_dai->driver->symmetric_samplebits || - rtd->dai_link->symmetric_samplebits; + symmetry = rtd->dai_link->symmetric_samplebits; + + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + symmetry |= cpu_dai->driver->symmetric_samplebits; for_each_rtd_codec_dai(rtd, i, codec_dai) symmetry |= codec_dai->driver->symmetric_samplebits; - if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) { - dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", - cpu_dai->sample_bits, sample_bits); - return -EINVAL; + if (symmetry) { + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + if (cpu_dai->sample_bits && + cpu_dai->sample_bits != sample_bits) { + dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", + cpu_dai->sample_bits, sample_bits); + return -EINVAL; + } + } } return 0; @@ -485,14 +509,20 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; struct snd_soc_dai_link *link = rtd->dai_link; struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; unsigned int symmetry, i; - symmetry = cpu_driver->symmetric_rates || link->symmetric_rates || - cpu_driver->symmetric_channels || link->symmetric_channels || - cpu_driver->symmetric_samplebits || link->symmetric_samplebits; + symmetry = link->symmetric_rates || + link->symmetric_channels || + link->symmetric_samplebits; + + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + symmetry = symmetry || + cpu_dai->driver->symmetric_rates || + cpu_dai->driver->symmetric_channels || + cpu_dai->driver->symmetric_samplebits; for_each_rtd_codec_dai(rtd, i, codec_dai) symmetry = symmetry || @@ -520,12 +550,12 @@ static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits) static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; struct snd_soc_pcm_stream *pcm_codec, *pcm_cpu; int stream = substream->stream; int i; - unsigned int bits = 0, cpu_bits; + unsigned int bits = 0, cpu_bits = 0; for_each_rtd_codec_dai(rtd, i, codec_dai) { pcm_codec = snd_soc_dai_get_pcm_stream(codec_dai, stream); @@ -537,8 +567,15 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) bits = max(pcm_codec->sig_bits, bits); } - pcm_cpu = snd_soc_dai_get_pcm_stream(cpu_dai, stream); - cpu_bits = pcm_cpu->sig_bits; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + pcm_cpu = snd_soc_dai_get_pcm_stream(cpu_dai, stream); + + if (pcm_cpu->sig_bits == 0) { + cpu_bits = 0; + break; + } + cpu_bits = max(pcm_cpu->sig_bits, cpu_bits); + } soc_pcm_set_msb(substream, bits); soc_pcm_set_msb(substream, cpu_bits); @@ -550,18 +587,32 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) struct snd_pcm_hardware *hw = &runtime->hw; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; unsigned int chan_min = 0, chan_max = UINT_MAX; + unsigned int cpu_chan_min = 0, cpu_chan_max = UINT_MAX; unsigned int rate_min = 0, rate_max = UINT_MAX; - unsigned int rates = UINT_MAX; + unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX; + unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX; u64 formats = ULLONG_MAX; int stream = substream->stream; int i; - cpu_stream = snd_soc_dai_get_pcm_stream(rtd->cpu_dai, stream); + /* first calculate min/max only for CPUs in the DAI link */ + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream); + + cpu_chan_min = max(cpu_chan_min, cpu_stream->channels_min); + cpu_chan_max = min(cpu_chan_max, cpu_stream->channels_max); + cpu_rate_min = max(cpu_rate_min, cpu_stream->rate_min); + cpu_rate_max = min_not_zero(cpu_rate_max, cpu_stream->rate_max); + formats &= cpu_stream->formats; + cpu_rates = snd_pcm_rate_mask_intersect(cpu_stream->rates, + cpu_rates); + } - /* first calculate min/max only for CODECs in the DAI link */ + /* second calculate min/max only for CODECs in the DAI link */ for_each_rtd_codec_dai(rtd, i, codec_dai) { /* @@ -589,27 +640,28 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* * chan min/max cannot be enforced if there are multiple CODEC DAIs - * connected to a single CPU DAI, use CPU DAI's directly and let + * connected to CPU DAI(s), use CPU DAI's directly and let * channel allocation be fixed up later */ if (rtd->num_codecs > 1) { - chan_min = cpu_stream->channels_min; - chan_max = cpu_stream->channels_max; + chan_min = cpu_chan_min; + chan_max = cpu_chan_max; } - hw->channels_min = max(chan_min, cpu_stream->channels_min); - hw->channels_max = min(chan_max, cpu_stream->channels_max); + /* finally find a intersection between CODECs and CPUs */ + hw->channels_min = max(chan_min, cpu_chan_min); + hw->channels_max = min(chan_max, cpu_chan_max); if (hw->formats) - hw->formats &= formats & cpu_stream->formats; + hw->formats &= formats; else - hw->formats = formats & cpu_stream->formats; - hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates); + hw->formats = formats; + hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates); snd_pcm_limit_hw_rates(runtime); - hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); + hw->rate_min = max(hw->rate_min, cpu_rate_min); hw->rate_min = max(hw->rate_min, rate_min); - hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max); hw->rate_max = min_not_zero(hw->rate_max, rate_max); } @@ -681,7 +733,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i; @@ -689,9 +741,11 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) snd_soc_runtime_deactivate(rtd, substream->stream); - snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream); - snd_soc_dai_shutdown(cpu_dai, substream); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + snd_soc_dai_shutdown(cpu_dai, substream); for_each_rtd_codec_dai(rtd, i, codec_dai) snd_soc_dai_shutdown(codec_dai, substream); @@ -726,9 +780,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; const char *codec_dai_name = "multicodec"; + const char *cpu_dai_name = "multicpu"; int i, ret = 0; for_each_rtd_components(rtd, i, component) @@ -751,11 +806,13 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } /* startup the audio subsystem */ - ret = snd_soc_dai_startup(cpu_dai, substream); - if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", - cpu_dai->name, ret); - goto cpu_dai_err; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + ret = snd_soc_dai_startup(cpu_dai, substream); + if (ret < 0) { + dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", + cpu_dai->name, ret); + goto cpu_dai_err; + } } for_each_rtd_codec_dai(rtd, i, codec_dai) { @@ -783,34 +840,39 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (rtd->num_codecs == 1) codec_dai_name = rtd->codec_dai->name; + if (rtd->num_cpus == 1) + cpu_dai_name = rtd->cpu_dai->name; + if (soc_pcm_has_symmetry(substream)) runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", - codec_dai_name, cpu_dai->name); + codec_dai_name, cpu_dai_name); goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n", - codec_dai_name, cpu_dai->name); + codec_dai_name, cpu_dai_name); goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max || runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n", - codec_dai_name, cpu_dai->name); + codec_dai_name, cpu_dai_name); goto config_err; } soc_pcm_apply_msb(substream); /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active) { - ret = soc_pcm_apply_symmetry(substream, cpu_dai); - if (ret != 0) - goto config_err; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + if (cpu_dai->active) { + ret = soc_pcm_apply_symmetry(substream, cpu_dai); + if (ret != 0) + goto config_err; + } } for_each_rtd_codec_dai(rtd, i, codec_dai) { @@ -822,7 +884,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } pr_debug("ASoC: %s <-> %s info:\n", - codec_dai_name, cpu_dai->name); + codec_dai_name, cpu_dai_name); pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates); pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -840,7 +902,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) for_each_rtd_codec_dai(rtd, i, codec_dai) snd_soc_dai_shutdown(codec_dai, substream); cpu_dai_err: - snd_soc_dai_shutdown(cpu_dai, substream); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + snd_soc_dai_shutdown(cpu_dai, substream); soc_rtd_shutdown(rtd, substream); rtd_startup_err: @@ -879,7 +942,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i, ret = 0; @@ -911,11 +974,13 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - ret = snd_soc_dai_prepare(cpu_dai, substream); - if (ret < 0) { - dev_err(cpu_dai->dev, - "ASoC: cpu DAI prepare error: %d\n", ret); - goto out; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + ret = snd_soc_dai_prepare(cpu_dai, substream); + if (ret < 0) { + dev_err(cpu_dai->dev, + "ASoC: cpu DAI prepare error: %d\n", ret); + goto out; + } } /* cancel any delayed stream shutdown that is pending */ @@ -931,7 +996,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) for_each_rtd_codec_dai(rtd, i, codec_dai) snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); - snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); out: mutex_unlock(&rtd->card->pcm_mutex); @@ -978,7 +1044,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i, ret = 0; @@ -1042,17 +1108,19 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); } - ret = snd_soc_dai_hw_params(cpu_dai, substream, params); - if (ret < 0) - goto interface_err; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + ret = snd_soc_dai_hw_params(cpu_dai, substream, params); + if (ret < 0) + goto interface_err; - /* store the parameters for each DAIs */ - cpu_dai->rate = params_rate(params); - cpu_dai->channels = params_channels(params); - cpu_dai->sample_bits = - snd_pcm_format_physical_width(params_format(params)); + /* store the parameters for each DAI */ + cpu_dai->rate = params_rate(params); + cpu_dai->channels = params_channels(params); + cpu_dai->sample_bits = + snd_pcm_format_physical_width(params_format(params)); - snd_soc_dapm_update_dai(substream, params, cpu_dai); + snd_soc_dapm_update_dai(substream, params, cpu_dai); + } for_each_rtd_components(rtd, i, component) { ret = snd_soc_component_hw_params(component, substream, params); @@ -1072,10 +1140,14 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, component_err: soc_pcm_components_hw_free(substream, component); - snd_soc_dai_hw_free(cpu_dai, substream); - cpu_dai->rate = 0; + i = rtd->num_cpus; interface_err: + for_each_rtd_cpu_dai_rollback(rtd, i, cpu_dai) { + snd_soc_dai_hw_free(cpu_dai, substream); + cpu_dai->rate = 0; + } + i = rtd->num_codecs; codec_err: @@ -1099,7 +1171,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; @@ -1107,10 +1179,12 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* clear the corresponding DAIs parameters when going to be inactive */ - if (cpu_dai->active == 1) { - cpu_dai->rate = 0; - cpu_dai->channels = 0; - cpu_dai->sample_bits = 0; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + if (cpu_dai->active == 1) { + cpu_dai->rate = 0; + cpu_dai->channels = 0; + cpu_dai->sample_bits = 0; + } } for_each_rtd_codec_dai(rtd, i, codec_dai) { @@ -1146,7 +1220,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_dai_hw_free(codec_dai, substream); } - snd_soc_dai_hw_free(cpu_dai, substream); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + snd_soc_dai_hw_free(cpu_dai, substream); mutex_unlock(&rtd->card->pcm_mutex); return 0; @@ -1156,7 +1231,7 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i, ret; @@ -1170,9 +1245,11 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) return ret; } - ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); - if (ret < 0) - return ret; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + } for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_trigger(codec_dai, substream, cmd); @@ -1187,7 +1264,7 @@ static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i, ret; @@ -1197,9 +1274,11 @@ static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) return ret; } - ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); - if (ret < 0) - return ret; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + } for_each_rtd_components(rtd, i, component) { ret = snd_soc_component_trigger(component, substream, cmd); @@ -1240,7 +1319,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i, ret; @@ -1250,9 +1329,11 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); - if (ret < 0) - return ret; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + } return 0; } @@ -1264,12 +1345,13 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; snd_pcm_sframes_t codec_delay = 0; + snd_pcm_sframes_t cpu_delay = 0; int i; /* clearing the previous total delay */ @@ -1280,7 +1362,11 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) /* base delay if assigned in pointer callback */ delay = runtime->delay; - delay += snd_soc_dai_delay(cpu_dai, substream); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + cpu_delay = max(cpu_delay, + snd_soc_dai_delay(cpu_dai, substream)); + } + delay += cpu_delay; for_each_rtd_codec_dai(rtd, i, codec_dai) { codec_delay = max(codec_delay, @@ -1403,13 +1489,15 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; - w = snd_soc_dai_get_widget(be->cpu_dai, stream); + for_each_rtd_cpu_dai(be, i, dai) { + w = snd_soc_dai_get_widget(dai, stream); - dev_dbg(card->dev, "ASoC: try BE : %s\n", - w ? w->name : "(not set)"); + dev_dbg(card->dev, "ASoC: try BE : %s\n", + w ? w->name : "(not set)"); - if (w == widget) - return be; + if (w == widget) + return be; + } for_each_rtd_codec_dai(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); @@ -1492,10 +1580,18 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, unsigned int i; /* is there a valid CPU DAI widget for this BE */ - widget = snd_soc_dai_get_widget(dpcm->be->cpu_dai, stream); + do_prune = 1; + for_each_rtd_cpu_dai(dpcm->be, i, dai) { + widget = snd_soc_dai_get_widget(dai, stream); - /* prune the BE if it's no longer in our active list */ - if (widget && widget_in_list(list, widget)) + /* + * The BE is pruned only if none of the cpu_dai + * widgets are in the active list. + */ + if (widget && widget_in_list(list, widget)) + do_prune = 0; + } + if (!do_prune) continue; /* is there a valid CODEC DAI widget for this BE */ @@ -1792,11 +1888,17 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; + struct snd_soc_dai *dai; + int i; - cpu_stream = snd_soc_dai_get_pcm_stream(be->cpu_dai, stream); + for_each_rtd_cpu_dai(be, i, dai) { + cpu_stream = snd_soc_dai_get_pcm_stream(dai, stream); - *channels_min = max(*channels_min, cpu_stream->channels_min); - *channels_max = min(*channels_max, cpu_stream->channels_max); + *channels_min = max(*channels_min, + cpu_stream->channels_min); + *channels_max = min(*channels_max, + cpu_stream->channels_max); + } /* * chan min/max cannot be enforced if there are multiple CODEC @@ -1837,11 +1939,15 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, struct snd_soc_dai *dai; int i; - cpu_stream = snd_soc_dai_get_pcm_stream(be->cpu_dai, stream); + for_each_rtd_cpu_dai(be, i, dai) { + cpu_stream = snd_soc_dai_get_pcm_stream(dai, stream); - *rate_min = max(*rate_min, cpu_stream->rate_min); - *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); - *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates); + *rate_min = max(*rate_min, cpu_stream->rate_min); + *rate_max = min_not_zero(*rate_max, + cpu_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, + cpu_stream->rates); + } for_each_rtd_codec_dai(be, i, dai) { /* @@ -1866,13 +1972,17 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai *cpu_dai; + struct snd_soc_dai_driver *cpu_dai_drv; + int i; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); - else - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + cpu_dai_drv = cpu_dai->driver; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); + else + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); + } dpcm_runtime_merge_format(substream, &runtime->hw.formats); dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min, @@ -1909,18 +2019,21 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, { struct snd_soc_dpcm *dpcm; struct snd_soc_pcm_runtime *fe = fe_substream->private_data; - struct snd_soc_dai *fe_cpu_dai = fe->cpu_dai; + struct snd_soc_dai *fe_cpu_dai; int err; + int i; /* apply symmetry for FE */ if (soc_pcm_has_symmetry(fe_substream)) fe_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; - /* Symmetry only applies if we've got an active stream. */ - if (fe_cpu_dai->active) { - err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai); - if (err < 0) - return err; + for_each_rtd_cpu_dai (fe, i, fe_cpu_dai) { + /* Symmetry only applies if we've got an active stream. */ + if (fe_cpu_dai->active) { + err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai); + if (err < 0) + return err; + } } /* apply symmetry for BE */ @@ -1930,6 +2043,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, snd_soc_dpcm_get_substream(be, stream); struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int i; /* A backend may not have the requested substream */ @@ -1944,11 +2058,13 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; /* Symmetry only applies if we've got an active stream. */ - if (rtd->cpu_dai->active) { - err = soc_pcm_apply_symmetry(fe_substream, - rtd->cpu_dai); - if (err < 0) - return err; + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + if (cpu_dai->active) { + err = soc_pcm_apply_symmetry(fe_substream, + cpu_dai); + if (err < 0) + return err; + } } for_each_rtd_codec_dai(rtd, i, codec_dai) { @@ -2863,7 +2979,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_component *component; struct snd_pcm *pcm; char new_name[64]; @@ -2881,6 +2997,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (rtd->num_cpus == 1) { + cpu_dai = rtd->cpu_dais[0]; + } else if (rtd->num_cpus == rtd->num_codecs) { + cpu_dai = rtd->cpu_dais[i]; + } else { + dev_err(rtd->card->dev, + "N cpus to M codecs link is not supported yet\n"); + return -EINVAL; + } + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && snd_soc_dai_stream_valid(cpu_dai, cpu_playback)) playback = 1; @@ -3001,7 +3127,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, - cpu_dai->name); + (rtd->num_cpus > 1) ? "multicpu" : rtd->cpu_dai->name); return ret; } From 6c4b13b51aa36aab023dd0bf24bf5582c9ba091e Mon Sep 17 00:00:00 2001 From: Shreyas NC Date: Tue, 25 Feb 2020 21:39:14 +0800 Subject: [PATCH 182/415] ASoC: Add dapm_add_valid_dai_widget helper Adding a helper to connect widget for a specific cpu and codec dai The helper will help dapm_connect_dai_link_widgets() to reduce indents. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Signed-off-by: Kuninori Morimoto Signed-off-by: Shreyas NC Signed-off-by: Vinod Koul Link: https://lore.kernel.org/r/20200225133917.21314-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 119 +++++++++++++++++++++++-------------------- 1 file changed, 63 insertions(+), 56 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 58c318c9debb1c..539a1eaebeacd5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4277,16 +4277,15 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd) +static void dapm_add_valid_dai_widget(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; struct snd_pcm_substream *substream; struct snd_pcm_str *streams = rtd->pcm->streams; - int i; if (rtd->dai_link->params) { playback_cpu = cpu_dai->capture_widget; @@ -4298,67 +4297,75 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, capture_cpu = capture; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - /* connect BE DAI playback if widgets are valid */ - codec = codec_dai->playback_widget; - - if (playback_cpu && codec) { - if (!playback) { - substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - playback = snd_soc_dapm_new_dai(card, substream, - "playback"); - if (IS_ERR(playback)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(playback)); - continue; - } - - snd_soc_dapm_add_path(&card->dapm, playback_cpu, - playback, NULL, NULL); + /* connect BE DAI playback if widgets are valid */ + codec = codec_dai->playback_widget; + + if (playback_cpu && codec) { + if (!playback) { + substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + playback = snd_soc_dapm_new_dai(card, substream, + "playback"); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + goto capture; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, playback_cpu->name, - codec_dai->component->name, codec->name); - - snd_soc_dapm_add_path(&card->dapm, playback, codec, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, playback_cpu, + playback, NULL, NULL); } - } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - /* connect BE DAI capture if widgets are valid */ - codec = codec_dai->capture_widget; - - if (codec && capture_cpu) { - if (!capture) { - substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; - capture = snd_soc_dapm_new_dai(card, substream, - "capture"); - if (IS_ERR(capture)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(capture)); - continue; - } - - snd_soc_dapm_add_path(&card->dapm, capture, - capture_cpu, NULL, NULL); + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + cpu_dai->component->name, playback_cpu->name, + codec_dai->component->name, codec->name); + + snd_soc_dapm_add_path(&card->dapm, playback, codec, + NULL, NULL); + } + +capture: + /* connect BE DAI capture if widgets are valid */ + codec = codec_dai->capture_widget; + + if (codec && capture_cpu) { + if (!capture) { + substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; + capture = snd_soc_dapm_new_dai(card, substream, + "capture"); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + return; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, codec->name, - cpu_dai->component->name, capture_cpu->name); - - snd_soc_dapm_add_path(&card->dapm, codec, capture, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, capture, + capture_cpu, NULL, NULL); } + + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + codec_dai->component->name, codec->name, + cpu_dai->component->name, capture_cpu->name); + + snd_soc_dapm_add_path(&card->dapm, codec, capture, + NULL, NULL); } } +static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, + codec_dai, rtd->cpu_dais[0]); +} + static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, int event) { From de6214a33633d8ce1c1490336f8e798e75ccd004 Mon Sep 17 00:00:00 2001 From: Shreyas NC Date: Tue, 25 Feb 2020 21:39:15 +0800 Subject: [PATCH 183/415] ASoC: Add multiple CPU DAI support in DAPM DAPM handles DAIs during soc_dapm_stream_event() and during addition and creation of DAI widgets i.e., dapm_add_valid_dai_widget() and dapm_connect_dai_link_widgets(). Extend these functions to handle multiple cpu dai. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Signed-off-by: Kuninori Morimoto Signed-off-by: Shreyas NC Signed-off-by: Vinod Koul Link: https://lore.kernel.org/r/20200225133917.21314-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 539a1eaebeacd5..6ce024d52170e5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4361,9 +4361,19 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, struct snd_soc_dai *codec_dai; int i; - for_each_rtd_codec_dai(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, - codec_dai, rtd->cpu_dais[0]); + if (rtd->num_cpus == 1) { + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, codec_dai, + rtd->cpu_dais[0]); + } else if (rtd->num_codecs == rtd->num_cpus) { + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, codec_dai, + rtd->cpu_dais[i]); + } else { + dev_err(card->dev, + "N cpus to M codecs link is not supported yet\n"); + } + } static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, @@ -4424,9 +4434,11 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int i; - soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); for_each_rtd_codec_dai(rtd, i, codec_dai) soc_dapm_dai_stream_event(codec_dai, stream, event); From 6e1276a5e613d25af71e3494b2dcb331d24f06ce Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 25 Feb 2020 21:39:16 +0800 Subject: [PATCH 184/415] ASoC: Return error if the function does not support multi-cpu Multi cpu is not supported by all functions yet. Add an error message and return. Suggested-by: Kuninori Morimoto Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20200225133917.21314-6-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 5 +++-- sound/soc/soc-generic-dmaengine-pcm.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 18 ++++++++++++++++++ 3 files changed, 39 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 392a1c5b15d324..50062eb79adbf5 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -810,9 +810,10 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) int playback = 0, capture = 0; int i; - if (rtd->num_codecs > 1) { + if (rtd->num_cpus > 1 || + rtd->num_codecs > 1) { dev_err(rtd->card->dev, - "Compress ASoC: Multicodec not supported\n"); + "Compress ASoC: Multi CPU/Codec not supported\n"); return -EINVAL; } diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index d6b4831e8aec9e..facf1922a714d9 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -62,6 +62,12 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_dmaengine_dai_dma_data *dma_data; int ret; + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); @@ -118,6 +124,12 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_hardware hw; + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + if (pcm->config && pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); @@ -185,6 +197,12 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_dmaengine_dai_dma_data *dma_data; dma_filter_fn fn = NULL; + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return NULL; + } + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 44694e65fc4ad7..adbceaff07b8d7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -125,6 +125,12 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, int stream; char *buf; + if (fe->num_cpus > 1) { + dev_err(fe->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + buf = kmalloc(out_count, GFP_KERNEL); if (!buf) return -ENOMEM; @@ -1550,6 +1556,12 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, struct snd_soc_dai *cpu_dai = fe->cpu_dai; int paths; + if (fe->num_cpus > 1) { + dev_err(fe->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + /* get number of valid DAI paths and their widgets */ paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, dpcm_end_walk_at_be); @@ -2834,6 +2846,12 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) int count, paths; int ret; + if (fe->num_cpus > 1) { + dev_err(fe->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + if (!fe->dai_link->dynamic) return 0; From 0e9cf4c452ad7e2776441cbac0b9983abaf17ff0 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 25 Feb 2020 21:39:17 +0800 Subject: [PATCH 185/415] ASoC: pcm: check if cpu-dai supports a given stream Now multi-cpu-dais are supported, we can skip cpi-dais which don't support the current stream, following the example of multi-codec-dais. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20200225133917.21314-7-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 51 ++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 50 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index adbceaff07b8d7..90857138c823d5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -607,6 +607,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CPUs in the DAI link */ for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + + /* + * Skip CPUs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + * At least one CPU should match, otherwise we should have + * bailed out on a higher level, since there would be no + * CPU to support the transfer direction in that case. + */ + if (!snd_soc_dai_stream_valid(cpu_dai, + substream->stream)) + continue; + cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream); cpu_chan_min = max(cpu_chan_min, cpu_stream->channels_min); @@ -1115,6 +1129,13 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + /* + * Skip CPUs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) + continue; + ret = snd_soc_dai_hw_params(cpu_dai, substream, params); if (ret < 0) goto interface_err; @@ -1150,6 +1171,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, interface_err: for_each_rtd_cpu_dai_rollback(rtd, i, cpu_dai) { + if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) + continue; + snd_soc_dai_hw_free(cpu_dai, substream); cpu_dai->rate = 0; } @@ -1226,8 +1250,12 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_dai_hw_free(codec_dai, substream); } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) + continue; + snd_soc_dai_hw_free(cpu_dai, substream); + } mutex_unlock(&rtd->card->pcm_mutex); return 0; @@ -1904,6 +1932,13 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, int i; for_each_rtd_cpu_dai(be, i, dai) { + /* + * Skip CPUs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(dai, stream)) + continue; + cpu_stream = snd_soc_dai_get_pcm_stream(dai, stream); *channels_min = max(*channels_min, @@ -1952,6 +1987,13 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, int i; for_each_rtd_cpu_dai(be, i, dai) { + /* + * Skip CPUs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(dai, stream)) + continue; + cpu_stream = snd_soc_dai_get_pcm_stream(dai, stream); *rate_min = max(*rate_min, cpu_stream->rate_min); @@ -1989,6 +2031,13 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) int i; for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + /* + * Skip CPUs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) + continue; + cpu_dai_drv = cpu_dai->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); From 36d73c4a9ed7c8a0988cfb9d1282c62d8c422a3b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 25 Feb 2020 11:00:40 -0600 Subject: [PATCH 186/415] ASoC: soc-dai: add get_sdw_stream() callback We only have a set() operation, provide the dual get() operation to retrieve the stream information. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200225170041.23644-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 92c382690930e4..7f70db149b812a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -202,6 +202,8 @@ struct snd_soc_dai_ops { int (*set_sdw_stream)(struct snd_soc_dai *dai, void *stream, int direction); + void *(*get_sdw_stream)(struct snd_soc_dai *dai, int direction); + /* * DAI digital mute - optional. * Called by soc-core to minimise any pops. @@ -423,4 +425,23 @@ static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai, return -ENOTSUPP; } +/** + * snd_soc_dai_get_sdw_stream() - Retrieves SDW stream from DAI + * @dai: DAI + * @direction: Stream direction(Playback/Capture) + * + * This routine only retrieves that was previously configured + * with snd_soc_dai_get_sdw_stream() + * + * Returns pointer to stream or NULL; + */ +static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai, + int direction) +{ + if (dai->driver->ops->get_sdw_stream) + return dai->driver->ops->get_sdw_stream(dai, direction); + else + return NULL; +} + #endif From 6b8e4e7db3cd236a2cbb720360fb135087a2ac1d Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Mon, 17 Feb 2020 10:35:01 +0530 Subject: [PATCH 187/415] ASoC: amd: Add machine driver for Raven based platform Add machine driver for Raven based platform using RT5682 + MAX9836 + CROS_EC codecs Signed-off-by: Akshu Agrawal Link: https://lore.kernel.org/r/20200217050515.3847-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 10 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp3x-rt5682-max9836.c | 334 +++++++++++++++++++++++++++ 3 files changed, 346 insertions(+) create mode 100644 sound/soc/amd/acp3x-rt5682-max9836.c diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 5f40517717c4ca..b29ef1373946cb 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -26,3 +26,13 @@ config SND_SOC_AMD_ACP3x depends on X86 && PCI help This option enables ACP v3.x I2S support on AMD platform + +config SND_SOC_AMD_RV_RT5682_MACH + tristate "AMD RV support for RT5682" + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_CROS_EC_CODEC + select I2C_CROS_EC_TUNNEL + depends on SND_SOC_AMD_ACP3x && I2C + help + This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index c4ddc6adb6f0fe..e6f3d9b469f329 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -2,8 +2,10 @@ acp_audio_dma-objs := acp-pcm-dma.o snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o snd-soc-acp-rt5645-mach-objs := acp-rt5645.o +snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/ +obj-$(CONFIG_SND_SOC_AMD_RV_RT5682_MACH) += snd-soc-acp-rt5682-mach.o diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c new file mode 100644 index 00000000000000..96fbcd29e3edaa --- /dev/null +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -0,0 +1,334 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec. +// +//Copyright 2016 Advanced Micro Devices, Inc. + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "raven/acp3x.h" +#include "../codecs/rt5682.h" + +#define PCO_PLAT_CLK 48000000 +#define RT5682_PLL_FREQ (48000 * 512) +#define DUAL_CHANNEL 2 + +static struct snd_soc_jack pco_jack; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; + +static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = codec_dai->component; + + dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); + + /* set rt5682 dai fmt */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + dev_err(rtd->card->dev, + "Failed to set rt5682 dai fmt: %d\n", ret); + return ret; + } + + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + PCO_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + + /* Set tdm/i2s1 master bclk ratio */ + ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (ret < 0) { + dev_err(rtd->dev, + "Failed to set rt5682 tdm bclk ratio: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = clk_get(component->dev, "rt5682-dai-wclk"); + rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk"); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pco_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &pco_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* RT5682 will support only 48K output with 48M mclk */ + clk_set_rate(rt5682_dai_wclk, 48000); + clk_set_rate(rt5682_dai_bclk, 48000 * 64); + ret = clk_prepare_enable(rt5682_dai_wclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable wclk %d\n", ret); + return ret; + } + + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_wclk); +} + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int acp3x_5682_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + return rt5682_clk_enable(substream); +} + +static int acp3x_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + return rt5682_clk_enable(substream); +} + +static int acp3x_ec_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + snd_soc_dai_set_bclk_ratio(codec_dai, 64); + + return rt5682_clk_enable(substream); +} + +static void rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + +static const struct snd_soc_ops acp3x_5682_ops = { + .startup = acp3x_5682_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_max_play_ops = { + .startup = acp3x_max_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_ec_cap_ops = { + .startup = acp3x_ec_startup, + .shutdown = rt5682_shutdown, +}; + +SND_SOC_DAILINK_DEF(acp3x_i2s, + DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0"))); +SND_SOC_DAILINK_DEF(acp3x_bt, + DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.2"))); + +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); +SND_SOC_DAILINK_DEF(max, + DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(cros_ec, + DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0"))); + +static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { + { + .name = "acp3x-5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = acp3x_5682_init, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &acp3x_5682_ops, + SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform), + }, + { + .name = "acp3x-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &acp3x_max_play_ops, + SND_SOC_DAILINK_REG(acp3x_bt, max, platform), + }, + { + .name = "acp3x-ec-capture", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_capture = 1, + .ops = &acp3x_ec_cap_ops, + SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), + }, +}; + +static const struct snd_soc_dapm_widget acp3x_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_audio_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Spk", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new acp3x_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Spk"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_card acp3x_card = { + .name = "acp3xalc5682m98357", + .owner = THIS_MODULE, + .dai_link = acp3x_dai_5682_98357, + .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), + .dapm_widgets = acp3x_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets), + .dapm_routes = acp3x_audio_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route), + .controls = acp3x_mc_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_controls), +}; + +static int acp3x_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_card *card; + struct acp3x_platform_info *machine; + + machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card = &acp3x_card; + acp3x_card.dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); + if (ret) { + dev_err(&pdev->dev, + "devm_snd_soc_register_card(%s) failed: %d\n", + acp3x_card.name, ret); + return ret; + } + return 0; +} + +static const struct acpi_device_id acp3x_audio_acpi_match[] = { + { "AMDI5682", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); + +static struct platform_driver acp3x_audio = { + .driver = { + .name = "acp3x-alc5682-max98357", + .acpi_match_table = ACPI_PTR(acp3x_audio_acpi_match), + .pm = &snd_soc_pm_ops, + }, + .probe = acp3x_probe, +}; + +module_platform_driver(acp3x_audio); + +MODULE_AUTHOR("akshu.agrawal@amd.com"); +MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support"); +MODULE_LICENSE("GPL v2"); From 03f6fc6de9192f4e4209ceee0e92f5947d44fc0a Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 19 Feb 2020 18:28:57 +0800 Subject: [PATCH 188/415] ASoC: rt5682: Add the soundwire support This patch adds the soundwire support for ALC5682. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200219102858.20166-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5682-sdw.c | 333 +++++++++++++++++++++ sound/soc/codecs/rt5682-sdw.h | 20 ++ sound/soc/codecs/rt5682.c | 526 +++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5682.h | 49 ++++ 6 files changed, 897 insertions(+), 40 deletions(-) create mode 100644 sound/soc/codecs/rt5682-sdw.c create mode 100644 sound/soc/codecs/rt5682-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a7e89567edbe8b..6aee70ed43df5b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -168,6 +168,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT5670 imply SND_SOC_RT5677 imply SND_SOC_RT5682 + imply SND_SOC_RT5682_SDW imply SND_SOC_RT700_SDW imply SND_SOC_RT711_SDW imply SND_SOC_RT715_SDW @@ -1136,6 +1137,12 @@ config SND_SOC_RT5682 tristate depends on I2C +config SND_SOC_RT5682_SDW + tristate "Realtek RT5682 Codec - SDW" + depends on SOUNDWIRE + select SND_SOC_RT5682 + select REGMAP_SOUNDWIRE + config SND_SOC_RT700 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 943ebc93fbc1cf..03533157cda66f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -177,6 +177,7 @@ snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o +snd-soc-rt5682-sdw-objs := rt5682-sdw.o snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-rt715-objs := rt715.o rt715-sdw.o @@ -477,6 +478,7 @@ obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o +obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c new file mode 100644 index 00000000000000..fc31d04b5203c2 --- /dev/null +++ b/sound/soc/codecs/rt5682-sdw.c @@ -0,0 +1,333 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt5682-sdw.c -- RT5682 ALSA SoC audio component driver +// +// Copyright 2019 Realtek Semiconductor Corp. +// Author: Oder Chiou +// + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rt5682.h" +#include "rt5682-sdw.h" + +static bool rt5682_sdw_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e0: + case 0x00f0: + case 0x3000: + case 0x3001: + case 0x3004: + case 0x3005: + case 0x3008: + return true; + default: + return false; + } +} + +const struct regmap_config rt5682_sdw_regmap = { + .name = "sdw", + .reg_bits = 32, + .val_bits = 8, + .max_register = RT5682_I2C_MODE, + .readable_reg = rt5682_sdw_readable_register, + .cache_type = REGCACHE_NONE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt5682_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt5682->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt5682->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt5682->hw_init || rt5682->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt5682_io_init(&slave->dev, slave); +} + +static int rt5682_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports = 1; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = false; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x4; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + num_of_ports += nval; + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + /* wake-up event */ + prop->wake_capable = 1; + + return 0; +} + +/* Bus clock frequency */ +#define RT5682_CLK_FREQ_9600000HZ 9600000 +#define RT5682_CLK_FREQ_12000000HZ 12000000 +#define RT5682_CLK_FREQ_6000000HZ 6000000 +#define RT5682_CLK_FREQ_4800000HZ 4800000 +#define RT5682_CLK_FREQ_2400000HZ 2400000 +#define RT5682_CLK_FREQ_12288000HZ 12288000 + +int rt5682_clock_config(struct device *dev) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (rt5682->params.curr_dr_freq >> 1); + + switch (clk_freq) { + case RT5682_CLK_FREQ_12000000HZ: + value = 0x0; + break; + case RT5682_CLK_FREQ_6000000HZ: + value = 0x1; + break; + case RT5682_CLK_FREQ_9600000HZ: + value = 0x2; + break; + case RT5682_CLK_FREQ_4800000HZ: + value = 0x3; + break; + case RT5682_CLK_FREQ_2400000HZ: + value = 0x4; + break; + case RT5682_CLK_FREQ_12288000HZ: + value = 0x5; + break; + default: + return -EINVAL; + } + + regmap_write(rt5682->sdw_regmap, 0xe0, value); + regmap_write(rt5682->sdw_regmap, 0xf0, value); + + dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq); + + return 0; +} + +static int rt5682_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + int ret; + + memcpy(&rt5682->params, params, sizeof(*params)); + + ret = rt5682_clock_config(&slave->dev); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return ret; +} + +static int rt5682_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + dev_dbg(&slave->dev, + "%s control_port_stat=%x", __func__, status->control_port); + + if (status->control_port & 0x4) { + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + } + + return 0; +} + +static struct sdw_slave_ops rt5682_slave_ops = { + .read_prop = rt5682_read_prop, + .interrupt_callback = rt5682_interrupt_callback, + .update_status = rt5682_update_status, + .bus_config = rt5682_bus_config, +}; + +static int rt5682_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Assign ops */ + slave->ops = &rt5682_slave_ops; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt5682_sdw_regmap); + if (IS_ERR(regmap)) + return -EINVAL; + + rt5682_sdw_init(&slave->dev, regmap, slave); + + return 0; +} + +static int rt5682_sdw_remove(struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + if (rt5682 && rt5682->hw_init) + cancel_delayed_work(&rt5682->jack_detect_work); + + return 0; +} + +static const struct sdw_device_id rt5682_id[] = { + SDW_SLAVE_ENTRY(0x025d, 0x5682, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt5682_id); + +static int rt5682_dev_suspend(struct device *dev) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + + if (!rt5682->hw_init) + return 0; + + regcache_cache_only(rt5682->regmap, true); + regcache_mark_dirty(rt5682->regmap); + + return 0; +} + +static int rt5682_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt5682->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt5682->regmap, false); + regcache_sync(rt5682->regmap); + + return 0; +} + +static const struct dev_pm_ops rt5682_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume) + SET_RUNTIME_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume, NULL) +}; + +static struct sdw_driver rt5682_sdw_driver = { + .driver = { + .name = "rt5682", + .owner = THIS_MODULE, + .pm = &rt5682_pm, + }, + .probe = rt5682_sdw_probe, + .remove = rt5682_sdw_remove, + .ops = &rt5682_slave_ops, + .id_table = rt5682_id, +}; +module_sdw_driver(rt5682_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT5682 driver SDW"); +MODULE_AUTHOR("Oder Chiou "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682-sdw.h b/sound/soc/codecs/rt5682-sdw.h new file mode 100644 index 00000000000000..76e6f607066e7d --- /dev/null +++ b/sound/soc/codecs/rt5682-sdw.h @@ -0,0 +1,20 @@ +/* SPDX-License-Identifier: GPL-2.0-only + * + * rt5682-sdw.h -- RT5682 SDW ALSA SoC audio driver + * + * Copyright 2019 Realtek Semiconductor Corp. + * Author: Oder Chiou + */ + +#ifndef __RT5682_SDW_H__ +#define __RT5682_SDW_H__ + +#define RT5682_SDW_ADDR_L 0x3000 +#define RT5682_SDW_ADDR_H 0x3001 +#define RT5682_SDW_DATA_L 0x3004 +#define RT5682_SDW_DATA_H 0x3005 +#define RT5682_SDW_CMD 0x3008 + +#define RT5682_PROBE_TIMEOUT 2000 + +#endif /* __RT5682_SDW_H__ */ diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 6774813e0eea53..1795a8bbea1a88 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -11,13 +11,13 @@ #include #include #include +#include #include #include #include #include #include #include -#include #include #include #include @@ -27,15 +27,11 @@ #include #include #include -#include -#include -#include #include #include "rl6231.h" #include "rt5682.h" - -#define RT5682_NUM_SUPPLIES 3 +#include "rt5682-sdw.h" static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { "AVDD", @@ -52,37 +48,6 @@ static const struct rt5682_platform_data i2s_default_platform_data = { .dai_clk_names[RT5682_DAI_BCLK_IDX] = "rt5682-dai-bclk", }; -struct rt5682_priv { - struct snd_soc_component *component; - struct rt5682_platform_data pdata; - struct regmap *regmap; - struct snd_soc_jack *hs_jack; - struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; - struct delayed_work jack_detect_work; - struct delayed_work jd_check_work; - struct mutex calibrate_mutex; - bool is_sdw; - -#ifdef CONFIG_COMMON_CLK - struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; - struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; - struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; - struct clk *mclk; -#endif - - int sysclk; - int sysclk_src; - int lrck[RT5682_AIFS]; - int bclk[RT5682_AIFS]; - int master[RT5682_AIFS]; - - int pll_src[RT5682_PLLS]; - int pll_in[RT5682_PLLS]; - int pll_out[RT5682_PLLS]; - - int jack_type; -}; - static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, @@ -819,6 +784,22 @@ static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux = static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux = SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum); +static const char * const rt5682_dac_select[] = { + "IF1", "SOUND" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682_dacl_enum, + RT5682_AD_DA_MIXER, RT5682_DAC1_L_SEL_SFT, rt5682_dac_select); + +static const struct snd_kcontrol_new rt5682_dac_l_mux = + SOC_DAPM_ENUM("DAC L Mux", rt5682_dacl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682_dacr_enum, + RT5682_AD_DA_MIXER, RT5682_DAC1_R_SEL_SFT, rt5682_dac_select); + +static const struct snd_kcontrol_new rt5682_dac_r_mux = + SOC_DAPM_ENUM("DAC R Mux", rt5682_dacr_enum); + static void rt5682_reset(struct rt5682_priv *rt5682) { regmap_write(rt5682->regmap, RT5682_RESET, 0); @@ -1271,6 +1252,9 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w, static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + if (rt5682->is_sdw) + return 0; + val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) & RT5682_GP4_PIN_MASK; if (w->shift == RT5682_PWR_ADC_S1F_BIT && @@ -1743,6 +1727,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SOUND DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SOUND DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), /* Digital Interface Select */ SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, @@ -1759,12 +1745,19 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, &rt5682_adcdat_pin_ctrl), + SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682_dac_l_mux), + SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682_dac_r_mux), + /* Audio Interface */ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1), SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SDWRX", "SDW Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SDWTX", "SDW Capture", 0, SND_SOC_NOPM, 0, 0), /* Output Side */ /* DAC mixer before sound effect */ @@ -1921,8 +1914,8 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, - {"IF1_ADC Mux", NULL, "I2S1"}, {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "I2S1"}, {"AIF1TX", NULL, "ADCDAT Mux"}, {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, @@ -1931,6 +1924,10 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, {"AIF2TX", NULL, "ADCDAT Mux"}, + {"SDWTX", NULL, "PLL2B"}, + {"SDWTX", NULL, "PLL2F"}, + {"SDWTX", NULL, "ADCDAT Mux"}, + {"IF1 DAC1 L", NULL, "AIF1RX"}, {"IF1 DAC1 L", NULL, "I2S1"}, {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, @@ -1938,10 +1935,24 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"IF1 DAC1 R", NULL, "I2S1"}, {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC L", NULL, "SDWRX"}, + {"SOUND DAC L", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC L", NULL, "PLL2B"}, + {"SOUND DAC L", NULL, "PLL2F"}, + {"SOUND DAC R", NULL, "SDWRX"}, + {"SOUND DAC R", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC R", NULL, "PLL2B"}, + {"SOUND DAC R", NULL, "PLL2F"}, + + {"DAC L Mux", "IF1", "IF1 DAC1 L"}, + {"DAC L Mux", "SOUND", "SOUND DAC L"}, + {"DAC R Mux", "IF1", "IF1 DAC1 R"}, + {"DAC R Mux", "SOUND", "SOUND DAC R"}, + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, - {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXL", "DAC1 Switch", "DAC L Mux"}, {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, - {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + {"DAC1 MIXR", "DAC1 Switch", "DAC R Mux"}, {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, @@ -2826,6 +2837,8 @@ static int rt5682_register_dai_clks(struct snd_soc_component *component) static int rt5682_probe(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_slave *slave; + unsigned long time; #ifdef CONFIG_COMMON_CLK int ret; @@ -2852,6 +2865,17 @@ static int rt5682_probe(struct snd_soc_component *component) rt5682->lrck[RT5682_AIF1] = CLK_48; #endif + if (rt5682->is_sdw) { + slave = rt5682->slave; + time = wait_for_completion_timeout( + &slave->initialization_complete, + msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + } + return 0; } @@ -2914,6 +2938,194 @@ static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { .set_bclk_ratio = rt5682_set_bclk2_ratio, }; +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt5682_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + unsigned int val_p = 0, val_c = 0, osr_p = 0, osr_c = 0; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -ENOMEM; + + if (!rt5682->slave) + return -EINVAL; + + /* SoundWire specific configuration */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 2; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.num = port; + + retval = sdw_stream_add_slave(rt5682->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + switch (params_rate(params)) { + case 48000: + val_p = RT5682_SDW_REF_1_48K; + val_c = RT5682_SDW_REF_2_48K; + break; + case 96000: + val_p = RT5682_SDW_REF_1_96K; + val_c = RT5682_SDW_REF_2_96K; + break; + case 192000: + val_p = RT5682_SDW_REF_1_192K; + val_c = RT5682_SDW_REF_2_192K; + break; + case 32000: + val_p = RT5682_SDW_REF_1_32K; + val_c = RT5682_SDW_REF_2_32K; + break; + case 24000: + val_p = RT5682_SDW_REF_1_24K; + val_c = RT5682_SDW_REF_2_24K; + break; + case 16000: + val_p = RT5682_SDW_REF_1_16K; + val_c = RT5682_SDW_REF_2_16K; + break; + case 12000: + val_p = RT5682_SDW_REF_1_12K; + val_c = RT5682_SDW_REF_2_12K; + break; + case 8000: + val_p = RT5682_SDW_REF_1_8K; + val_c = RT5682_SDW_REF_2_8K; + break; + case 44100: + val_p = RT5682_SDW_REF_1_44K; + val_c = RT5682_SDW_REF_2_44K; + break; + case 88200: + val_p = RT5682_SDW_REF_1_88K; + val_c = RT5682_SDW_REF_2_88K; + break; + case 176400: + val_p = RT5682_SDW_REF_1_176K; + val_c = RT5682_SDW_REF_2_176K; + break; + case 22050: + val_p = RT5682_SDW_REF_1_22K; + val_c = RT5682_SDW_REF_2_22K; + break; + case 11025: + val_p = RT5682_SDW_REF_1_11K; + val_c = RT5682_SDW_REF_2_11K; + break; + default: + return -EINVAL; + } + + if (params_rate(params) <= 48000) { + osr_p = RT5682_DAC_OSR_D_8; + osr_c = RT5682_ADC_OSR_D_8; + } else if (params_rate(params) <= 96000) { + osr_p = RT5682_DAC_OSR_D_4; + osr_c = RT5682_ADC_OSR_D_4; + } else { + osr_p = RT5682_DAC_OSR_D_2; + osr_c = RT5682_ADC_OSR_D_2; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK, + RT5682_SDW_REF_1_MASK, val_p); + regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1, + RT5682_DAC_OSR_MASK, osr_p); + } else { + regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK, + RT5682_SDW_REF_2_MASK, val_c); + regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1, + RT5682_ADC_OSR_MASK, osr_c); + } + + return retval; +} + +static int rt5682_sdw_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt5682->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt5682->slave, stream->sdw_stream); + return 0; +} + +static struct snd_soc_dai_ops rt5682_sdw_ops = { + .hw_params = rt5682_sdw_hw_params, + .hw_free = rt5682_sdw_hw_free, + .set_sdw_stream = rt5682_set_sdw_stream, + .shutdown = rt5682_sdw_shutdown, +}; +#endif + static struct snd_soc_dai_driver rt5682_dai[] = { { .name = "rt5682-aif1", @@ -2946,6 +3158,27 @@ static struct snd_soc_dai_driver rt5682_dai[] = { }, .ops = &rt5682_aif2_dai_ops, }, +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) + { + .name = "rt5682-sdw", + .id = RT5682_SDW, + .playback = { + .stream_name = "SDW Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .capture = { + .stream_name = "SDW Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_sdw_ops, + }, +#endif }; static const struct snd_soc_component_driver soc_component_dev_rt5682 = { @@ -3064,6 +3297,219 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) } +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) +static int rt5682_sdw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct device *dev = context; + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned int data_l, data_h; + + regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 0); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff)); + regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_H, &data_h); + regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_L, &data_l); + + *val = (data_h << 8) | data_l; + + dev_vdbg(dev, "[%s] %04x => %04x\n", __func__, reg, *val); + + return 0; +} + +static int rt5682_sdw_write(void *context, unsigned int reg, unsigned int val) +{ + struct device *dev = context; + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + + regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 1); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff)); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_H, (val >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_L, (val & 0xff)); + + dev_vdbg(dev, "[%s] %04x <= %04x\n", __func__, reg, val); + + return 0; +} + +static const struct regmap_config rt5682_sdw_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682_I2C_MODE, + .volatile_reg = rt5682_volatile_register, + .readable_reg = rt5682_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682_reg), + .use_single_read = true, + .use_single_write = true, + .reg_read = rt5682_sdw_read, + .reg_write = rt5682_sdw_write, +}; + +int rt5682_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682; + int ret; + + rt5682 = devm_kzalloc(dev, sizeof(*rt5682), GFP_KERNEL); + if (!rt5682) + return -ENOMEM; + + dev_set_drvdata(dev, rt5682); + rt5682->slave = slave; + rt5682->sdw_regmap = regmap; + rt5682->is_sdw = true; + + rt5682->regmap = devm_regmap_init(dev, NULL, dev, &rt5682_sdw_regmap); + if (IS_ERR(rt5682->regmap)) { + ret = PTR_ERR(rt5682->regmap); + dev_err(dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt5682->hw_init = false; + rt5682->first_hw_init = false; + + mutex_init(&rt5682->calibrate_mutex); + INIT_DELAYED_WORK(&rt5682->jack_detect_work, + rt5682_jack_detect_handler); + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_rt5682, + rt5682_dai, ARRAY_SIZE(rt5682_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} +EXPORT_SYMBOL_GPL(rt5682_sdw_init); + +int rt5682_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + int ret = 0; + unsigned int val; + + if (rt5682->hw_init) + return 0; + + regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5682\n", val); + return -ENODEV; + } + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt5682->first_hw_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + rt5682_reset(rt5682); + + if (rt5682->first_hw_init) { + regcache_cache_only(rt5682->regmap, false); + regcache_cache_bypass(rt5682->regmap, true); + } + + rt5682_calibrate(rt5682); + + if (rt5682->first_hw_init) { + regcache_cache_bypass(rt5682->regmap, false); + regcache_mark_dirty(rt5682->regmap); + regcache_sync(rt5682->regmap); + + /* volatile registers */ + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + + goto reinit; + } + + ret = regmap_multi_reg_write(rt5682->regmap, patch_list, + ARRAY_SIZE(patch_list)); + if (ret != 0) + dev_warn(dev, "Failed to apply regmap patch: %d\n", ret); + + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000); + + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, + RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, + RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); + regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, + RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); + + /* Soundwire */ + regmap_write(rt5682->regmap, RT5682_PLL2_INTERNAL, 0xa266); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_1, 0x1700); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_2, 0x0006); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_3, 0x2600); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_4, 0x0c8f); + regmap_write(rt5682->regmap, RT5682_PLL_TRACK_2, 0x3000); + regmap_write(rt5682->regmap, RT5682_PLL_TRACK_3, 0x4000); + regmap_update_bits(rt5682->regmap, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK | RT5682_PLL2_SRC_MASK, + RT5682_SCLK_SRC_PLL2 | RT5682_PLL2_SRC_SDW); + + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, + RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); + regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, + RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_IRQ | RT5682_POW_JDH | + RT5682_POW_ANA, RT5682_POW_IRQ | + RT5682_POW_JDH | RT5682_POW_ANA); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH, RT5682_PWR_JDH); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_IRQ_MASK, + RT5682_JD1_EN | RT5682_JD1_IRQ_PUL); + +reinit: + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + + /* Mark Slave initialization complete */ + rt5682->hw_init = true; + rt5682->first_hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + + return ret; +} +EXPORT_SYMBOL_GPL(rt5682_io_init); +#endif + static int rt5682_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index f82126a6f2118e..43de6e8023093e 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -10,6 +10,12 @@ #define __RT5682_H__ #include +#include +#include +#include +#include +#include +#include #define DEVICE_ID 0x6530 @@ -1355,6 +1361,7 @@ enum { enum { RT5682_AIF1, RT5682_AIF2, + RT5682_SDW, RT5682_AIFS }; @@ -1370,7 +1377,49 @@ enum { RT5682_CLK_SEL_I2S2_ASRC, }; +#define RT5682_NUM_SUPPLIES 3 + +struct rt5682_priv { + struct snd_soc_component *component; + struct rt5682_platform_data pdata; + struct regmap *regmap; + struct regmap *sdw_regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + struct sdw_slave *slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + bool is_sdw; + +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; + struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; + struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + + int sysclk; + int sysclk_src; + int lrck[RT5682_AIFS]; + int bclk[RT5682_AIFS]; + int master[RT5682_AIFS]; + + int pll_src[RT5682_PLLS]; + int pll_in[RT5682_PLLS]; + int pll_out[RT5682_PLLS]; + + int jack_type; +}; + int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, unsigned int filter_mask, unsigned int clk_src); +int rt5682_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave); +int rt5682_io_init(struct device *dev, struct sdw_slave *slave); #endif /* __RT5682_H__ */ From b2d48dde38d373487503fd36cda05f17c1183b6d Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 19 Feb 2020 18:28:58 +0800 Subject: [PATCH 189/415] ASoC: rt5682: Revise the function name This patch revises the function name. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200219102858.20166-2-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 1795a8bbea1a88..e1df2d0765333a 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1554,7 +1554,7 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static int rt5655_set_verf(struct snd_soc_dapm_widget *w, +static int rt5682_set_verf(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = @@ -1632,7 +1632,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), From 911abf8b050e76591479d35c928f7e72605067ac Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Wed, 26 Feb 2020 16:17:44 +0530 Subject: [PATCH 190/415] ASoC: amd: Allow I2S wake event after ACP is powerd On ACP_PME_EN allows wake interrupt to be generated when I2S wake feature is enabled. On turning ACP On, ACP_PME_EN gets cleared. Setting the bit back ensures that wake event can be received when ACP is On. Signed-off-by: Akshu Agrawal Link: https://lore.kernel.org/r/20200226104746.208656-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/raven/pci-acp3x.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index da60e2ec553517..f25ce50f1a9016 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -38,8 +38,13 @@ static int acp3x_power_on(void __iomem *acp3x_base) timeout = 0; while (++timeout < 500) { val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); - if (!val) + if (!val) { + /* Set PME_EN as after ACP power On, + * PME_EN gets cleared + */ + rv_writel(0x1, acp3x_base + mmACP_PME_EN); return 0; + } udelay(1); } return -ETIMEDOUT; From f87cdb1f9937e6f5234e3300804ac156e639bc00 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Wed, 26 Feb 2020 07:03:03 -0600 Subject: [PATCH 191/415] ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding Add the Texas Instruments TAS2563 audio amplifier to the TAS262 binding. Signed-off-by: Dan Murphy CC: Rob Herring Link: https://lore.kernel.org/r/20200226130305.12043-1-dmurphy@ti.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/tas2562.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt index 658e1fb18a99cd..94796b547184ba 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -8,7 +8,7 @@ real time monitoring of loudspeaker behavior. Required properties: - #address-cells - Should be <1>. - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562". + - compatible: - Should contain "ti,tas2562", "ti,tas2563". - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - ti,imon-slot-no:- TDM TX current sense time slot. From 14f8c8d8fd62207f081549d45099a90dd3717696 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Wed, 26 Feb 2020 07:03:04 -0600 Subject: [PATCH 192/415] ASoC: tas2562: Add entries for the TAS2563 audio amplifier The TAS2563 is register compatible with the TAS2562. The main difference is the TAS2563 has a programmable DSP to manage different audio profiles. Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200226130305.12043-2-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2562.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index d5e04030a0c176..79c3c3d7976618 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -55,6 +55,11 @@ struct tas2562_data { int volume_lvl; }; +enum tas256x_model { + TAS2562, + TAS2563, +}; + static int tas2562_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { @@ -664,13 +669,15 @@ static int tas2562_probe(struct i2c_client *client, } static const struct i2c_device_id tas2562_id[] = { - { "tas2562", 0 }, + { "tas2562", TAS2562 }, + { "tas2563", TAS2563 }, { } }; MODULE_DEVICE_TABLE(i2c, tas2562_id); static const struct of_device_id tas2562_of_match[] = { { .compatible = "ti,tas2562", }, + { .compatible = "ti,tas2563", }, { }, }; MODULE_DEVICE_TABLE(of, tas2562_of_match); From 4a88b7dec331cf1ac661e38d610cd0ff0c073607 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Thu, 27 Feb 2020 10:06:37 +0800 Subject: [PATCH 193/415] ASoC: rt1015: modify some structure to be static. Modify rt1015_aif_dai_ops and rt1015_dai[] to be static. Signed-off-by: Jack Yu Link: https://lore.kernel.org/r/20200227020637.15135-1-jack.yu@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1015.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index d300b417dd508c..c118d030bd2d0f 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -841,12 +841,12 @@ static void rt1015_remove(struct snd_soc_component *component) #define RT1015_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt1015_aif_dai_ops = { +static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, }; -struct snd_soc_dai_driver rt1015_dai[] = { +static struct snd_soc_dai_driver rt1015_dai[] = { { .name = "rt1015-aif", .id = 0, From a3c2e894cdafbfa376a28a89a60df415b6ab6ee6 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Fri, 28 Feb 2020 15:56:09 +0800 Subject: [PATCH 194/415] ASoC: rt5682: Make rt5682_clock_config static Fix sparse warning: sound/soc/codecs/rt5682-sdw.c:163:5: warning: symbol 'rt5682_clock_config' was not declared. Should it be static? Reported-by: Hulk Robot Signed-off-by: YueHaibing Link: https://lore.kernel.org/r/20200228075609.38236-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index fc31d04b5203c2..1d6963dd6403aa 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -160,7 +160,7 @@ static int rt5682_read_prop(struct sdw_slave *slave) #define RT5682_CLK_FREQ_2400000HZ 2400000 #define RT5682_CLK_FREQ_12288000HZ 12288000 -int rt5682_clock_config(struct device *dev) +static int rt5682_clock_config(struct device *dev) { struct rt5682_priv *rt5682 = dev_get_drvdata(dev); unsigned int clk_freq, value; From 1a1b3743487317514f7d5d66dd9d6c9233321eba Mon Sep 17 00:00:00 2001 From: Marek Szyprowski Date: Fri, 28 Feb 2020 11:11:20 +0100 Subject: [PATCH 195/415] ASoC: samsung: Silence warnings during deferred probe Don't confuse user with meaningless warning about the failure in getting resources and registering card in case of deferred probe. Signed-off-by: Marek Szyprowski Reviewed-by: Sylwester Nawrocki Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20200228101120.28819-1-m.szyprowski@samsung.com Signed-off-by: Mark Brown --- sound/soc/samsung/arndale.c | 4 +++- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/lowland.c | 2 +- sound/soc/samsung/odroid.c | 4 +++- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/snow.c | 4 +++- sound/soc/samsung/speyside.c | 2 +- sound/soc/samsung/tm2_wm5110.c | 3 ++- sound/soc/samsung/tobermory.c | 2 +- 10 files changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index d64602950cbdf3..6e6d67d6e0ab4c 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -174,7 +174,9 @@ static int arndale_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(card->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); goto err_put_of_nodes; } return 0; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 59904f44118b9d..2f2f83a8c23a35 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -325,7 +325,7 @@ static int littlemill_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 098eefc764dbaa..fcc7897ee7d05e 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -183,7 +183,7 @@ static int lowland_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index f0f5fa9c27d32a..30c7e1bc2a30b2 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -311,7 +311,9 @@ static int odroid_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { - dev_err(dev, "snd_soc_register_card() failed: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(dev, "snd_soc_register_card() failed: %d\n", + ret); goto err_put_clk_i2s; } diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 28f8be000aa1dc..8fa5f6b387adcf 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -178,7 +178,7 @@ static int smdk_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); return ret; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 2e3dc7320c62e9..6e44f7927852e3 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -118,7 +118,7 @@ static int snd_smdk_probe(struct platform_device *pdev) smdk_pcm.dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); return ret; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index f075aae9561a70..bebcf0a4d608e7 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -216,7 +216,9 @@ static int snow_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card failed (%d)\n", ret); return ret; } diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index ea0d1ec67f0101..8f175f204eb70a 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -330,7 +330,7 @@ static int speyside_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 10ff14b856f236..043a287728b36b 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -611,7 +611,8 @@ static int tm2_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { - dev_err(dev, "Failed to register card: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(dev, "Failed to register card: %d\n", ret); goto dai_node_put; } diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index fdce28cc26c415..1aa3fdb4b15218 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -229,7 +229,7 @@ static int tobermory_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); From ac5bf39e39683c6f06c2e5b4baf27c7208f0c86d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Feb 2020 10:47:02 +0900 Subject: [PATCH 196/415] ASoC: soc-dapm: don't use rtd->cpu_dai on for_each_rtd_cpu_dai() soc_dapm_stream_event() is using for_each_rtd_cpu_dais(). It should use "cpu_dai", instead of "rtd->cpu_dai". This patch fixup it. Fixes: commit de6214a33633d ("ASoC: Add multiple CPU DAI support in DAPM") Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87pne07qeh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6ce024d52170e5..9a809f2caa10dc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4438,7 +4438,7 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int i; for_each_rtd_cpu_dai(rtd, i, cpu_dai) - soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); + soc_dapm_dai_stream_event(cpu_dai, stream, event); for_each_rtd_codec_dai(rtd, i, codec_dai) soc_dapm_dai_stream_event(codec_dai, stream, event); From a57ec83a7104eab6f08215702067fbcbef90c0a0 Mon Sep 17 00:00:00 2001 From: tangbin Date: Thu, 27 Feb 2020 23:07:01 +0800 Subject: [PATCH 197/415] ASoC: zte: zx-spdif: remove redundant dev_err message devm_ioremap_resource has already contains error message, so remove the redundant dev_err message Signed-off-by: tangbin Link: https://lore.kernel.org/r/20200227150701.15652-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/zte/zx-spdif.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c index 60382ec23832d7..a3a07c0730e691 100644 --- a/sound/soc/zte/zx-spdif.c +++ b/sound/soc/zte/zx-spdif.c @@ -322,7 +322,6 @@ static int zx_spdif_probe(struct platform_device *pdev) zx_spdif->mapbase = res->start; zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(zx_spdif->reg_base)) { - dev_err(&pdev->dev, "ioremap failed!\n"); return PTR_ERR(zx_spdif->reg_base); } From e582f4832a9ee27d92502b58f3a1b3331457e8bb Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Fri, 28 Feb 2020 17:18:47 -0600 Subject: [PATCH 198/415] ASoC: SOF: pcm: skip DMA buffer pre-allocation As discussion in ALSA https://patchwork.kernel.org/patch/11336023/, it is suggested to skip DMA buffer pre-allocation with passing size=0 when calling snd_pcm_set_managed_buffer(), to make the full buffer_bytes range configured in topology file selectable from user space, here do the corresponding change in SOF PCM driver to implement it. This change doesn't have dependency to the change that Takashi will do in the ALSA core by adding total_pcm_alloc_bytes limitation to the struct snd_card, it passes tests both with or without Takashi's coming change on SOF CML platform. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Keyon Jie Link: https://lore.kernel.org/r/20200228231850.9226-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index b239bbff4b5cda..f4769e19965a55 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -601,8 +601,7 @@ static int sof_pcm_new(struct snd_soc_component *component, snd_pcm_set_managed_buffer(pcm->streams[stream].substream, SNDRV_DMA_TYPE_DEV_SG, sdev->dev, - le32_to_cpu(caps->buffer_size_min), - le32_to_cpu(caps->buffer_size_max)); + 0, le32_to_cpu(caps->buffer_size_max)); capture: stream = SNDRV_PCM_STREAM_CAPTURE; @@ -624,8 +623,7 @@ static int sof_pcm_new(struct snd_soc_component *component, snd_pcm_set_managed_buffer(pcm->streams[stream].substream, SNDRV_DMA_TYPE_DEV_SG, sdev->dev, - le32_to_cpu(caps->buffer_size_min), - le32_to_cpu(caps->buffer_size_max)); + 0, le32_to_cpu(caps->buffer_size_max)); return 0; } From 1919b42ca4ad75a2397081164661af3ce5a7b8f4 Mon Sep 17 00:00:00 2001 From: Jaska Uimonen Date: Fri, 28 Feb 2020 17:18:48 -0600 Subject: [PATCH 199/415] ASoC: SOF: ipc: check ipc return value before data copy In tx_wait_done the ipc payload is copied before the DSP transaction error code is checked. This might lead to corrupted data in kernel side even though the error would be handled later. It is also pointless to copy the data in case of error. So change the order of error check and copy. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Jaska Uimonen Link: https://lore.kernel.org/r/20200228231850.9226-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 22d296f957610f..cc5762706c9c99 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -214,15 +214,17 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, snd_sof_handle_fw_exception(ipc->sdev); ret = -ETIMEDOUT; } else { - /* copy the data returned from DSP */ ret = msg->reply_error; - if (msg->reply_size) - memcpy(reply_data, msg->reply_data, msg->reply_size); - if (ret < 0) + if (ret < 0) { dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n", hdr->cmd, msg->reply_size); - else + } else { ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); + if (msg->reply_size) + /* copy the data returned from DSP */ + memcpy(reply_data, msg->reply_data, + msg->reply_size); + } } return ret; From 8354d9b44530b5f534146668ae641572dede79a4 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Fri, 28 Feb 2020 17:18:49 -0600 Subject: [PATCH 200/415] ASoC: SOF: Intel: hda-loader: clear the IPC ack bit after FW_PURGE done Set DONE bit after the FW_PURGE IPC is polled successfully, to clear the interrupt and avoid the arrival of the confusing unexpected ipc. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Keyon Jie Link: https://lore.kernel.org/r/20200228231850.9226-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 8852184a256905..03b05d7f66da3d 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -131,6 +131,12 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, const void *fwdata, goto err; } + /* set DONE bit to clear the reply IPC message */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, + chip->ipc_ack, + chip->ipc_ack_mask, + chip->ipc_ack_mask); + /* step 5: power down corex */ ret = hda_dsp_core_power_down(sdev, chip->cores_mask & ~(HDA_DSP_CORE_MASK(0))); From 1a2289fdf678b780b2d68f408523c09b7074982e Mon Sep 17 00:00:00 2001 From: Tomasz Lauda Date: Fri, 28 Feb 2020 17:18:50 -0600 Subject: [PATCH 201/415] ASoC: SOF: add core id to sof_ipc_comp Adds core id to sof_ipc_comp. The intention of this change is to inform FW on which core that particular component should run. Right now core id is only passed when pipeline is created, which is not flexible enough and doesn't allow for FW to handle this the right way. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Tomasz Lauda Link: https://lore.kernel.org/r/20200228231850.9226-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/topology.h | 3 ++- include/uapi/sound/sof/abi.h | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h index 8e76178fedf08a..402e0250c5088d 100644 --- a/include/sound/sof/topology.h +++ b/include/sound/sof/topology.h @@ -53,9 +53,10 @@ struct sof_ipc_comp { uint32_t id; enum sof_comp_type type; uint32_t pipeline_id; + uint32_t core; /* reserved for future use */ - uint32_t reserved[2]; + uint32_t reserved[1]; } __packed; /* diff --git a/include/uapi/sound/sof/abi.h b/include/uapi/sound/sof/abi.h index c0ef1643c75389..5995b79d6df13c 100644 --- a/include/uapi/sound/sof/abi.h +++ b/include/uapi/sound/sof/abi.h @@ -26,7 +26,7 @@ /* SOF ABI version major, minor and patch numbers */ #define SOF_ABI_MAJOR 3 -#define SOF_ABI_MINOR 12 +#define SOF_ABI_MINOR 13 #define SOF_ABI_PATCH 0 /* SOF ABI version number. Format within 32bit word is MMmmmppp */ From 787c5214ea6f6e9b7c75ae670d6b6a7deecb2d45 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Fri, 28 Feb 2020 15:42:25 -0800 Subject: [PATCH 202/415] ASoC: SOF: Intel: hda: use snd_sof_dsp_set_power_state() op Replace the calls to hda_dsp_set_power_state() with the top-level SOF op snd_sof_set_power_state(). Along with this, modify the hda_dsp_resume() function to return the value of snd_sof_set_power_state() directly. Signed-off-by: Ranjani Sridharan Reviewed-by: Jaska Uimonen Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200228234225.6963-1-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 0e61c27785a3ae..79ce52c32ef1a8 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -662,7 +662,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) hda_codec_i915_display_power(sdev, true); /* Set DSP power state */ - ret = hda_dsp_set_power_state(sdev, &target_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_state); if (ret < 0) { dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", target_state.state, target_state.substate); @@ -686,8 +686,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - hda_dsp_set_power_state(sdev, &target_state); - return ret; + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) @@ -702,7 +701,7 @@ int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hda_dsp_set_power_state(sdev, &target_state); + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) @@ -730,7 +729,7 @@ int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hda_dsp_set_power_state(sdev, &target_state); + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) @@ -753,7 +752,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) hda_codec_i915_display_power(sdev, false); /* Set DSP power state */ - ret = hda_dsp_set_power_state(sdev, &target_dsp_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_dsp_state); if (ret < 0) { dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", target_dsp_state.state, @@ -781,7 +780,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) return ret; } - return hda_dsp_set_power_state(sdev, &target_dsp_state); + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); } int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) @@ -849,7 +848,7 @@ void hda_dsp_d0i3_work(struct work_struct *work) return; /* This can fail but error cannot be propagated */ - ret = hda_dsp_set_power_state(sdev, &target_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_state); if (ret < 0) dev_err_ratelimited(sdev->dev, "error: failed to set DSP state %d substate %d\n", From 72c3b2b09fcdaa6a63e17e9a715e2a8236af529a Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Mon, 2 Mar 2020 13:54:36 +0530 Subject: [PATCH 203/415] ASoc: amd: Add DMIC switch capability to machine driver Switch between DMIC0 and DMIC1 based on recording device selected. This is done by toggling the dmic select gpio. Signed-off-by: Akshu Agrawal Link: https://lore.kernel.org/r/20200302082443.51587-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp3x-rt5682-max9836.c | 53 ++++++++++++++++++++++++---- 1 file changed, 47 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 96fbcd29e3edaa..511b8b1722aafb 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include #include @@ -27,6 +28,7 @@ static struct snd_soc_jack pco_jack; static struct clk *rt5682_dai_wclk; static struct clk *rt5682_dai_bclk; +static struct gpio_desc *dmic_sel; static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) { @@ -176,7 +178,7 @@ static int acp3x_max_startup(struct snd_pcm_substream *substream) return rt5682_clk_enable(substream); } -static int acp3x_ec_startup(struct snd_pcm_substream *substream) +static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; @@ -185,6 +187,23 @@ static int acp3x_ec_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_BT_INSTANCE; snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (dmic_sel) + gpiod_set_value(dmic_sel, 0); + + return rt5682_clk_enable(substream); +} + +static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (dmic_sel) + gpiod_set_value(dmic_sel, 1); return rt5682_clk_enable(substream); } @@ -204,8 +223,13 @@ static const struct snd_soc_ops acp3x_max_play_ops = { .shutdown = rt5682_shutdown, }; -static const struct snd_soc_ops acp3x_ec_cap_ops = { - .startup = acp3x_ec_startup, +static const struct snd_soc_ops acp3x_ec_cap0_ops = { + .startup = acp3x_ec_dmic0_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_ec_cap1_ops = { + .startup = acp3x_ec_dmic1_startup, .shutdown = rt5682_shutdown, }; @@ -246,12 +270,21 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { SND_SOC_DAILINK_REG(acp3x_bt, max, platform), }, { - .name = "acp3x-ec-capture", - .stream_name = "Capture", + .name = "acp3x-ec-dmic0-capture", + .stream_name = "Capture DMIC0", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_capture = 1, + .ops = &acp3x_ec_cap0_ops, + SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), + }, + { + .name = "acp3x-ec-dmic1-capture", + .stream_name = "Capture DMIC1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .dpcm_capture = 1, - .ops = &acp3x_ec_cap_ops, + .ops = &acp3x_ec_cap1_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), }, }; @@ -302,6 +335,14 @@ static int acp3x_probe(struct platform_device *pdev) acp3x_card.dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); + + dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); + if (IS_ERR(dmic_sel)) { + dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + PTR_ERR(dmic_sel)); + return PTR_ERR(dmic_sel); + } + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); if (ret) { dev_err(&pdev->dev, From a79ee2e095c0a60c32d5b1fce39d58e0fc4d9ec5 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Mon, 2 Mar 2020 15:05:22 +0800 Subject: [PATCH 204/415] ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver snd_soc_dai_driver should set ops in rt1015_dai driver. Also make the two variable static to fix sparse warnings. Fixes: df31007400c3 ("ASoC: rt1015: add rt1015 amplifier driver") Signed-off-by: YueHaibing Link: https://lore.kernel.org/r/20200302070522.48104-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index c118d030bd2d0f..100b8c89d53776 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -857,6 +857,7 @@ static struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; From e7e2afeacaa6e6b3d428ca8dd0507f1098bafe5d Mon Sep 17 00:00:00 2001 From: Enric Balletbo i Serra Date: Tue, 3 Mar 2020 12:05:14 +0100 Subject: [PATCH 205/415] ASoC: amd: AMD RV RT5682 should depends on CROS_EC If SND_SOC_AMD_RV_RT5682_MACH=y, below kconfig and build errors can be seen: WARNING: unmet direct dependencies detected for SND_SOC_CROS_EC_CODEC WARNING: unmet direct dependencies detected for I2C_CROS_EC_TUNNEL ld: drivers/i2c/busses/i2c-cros-ec-tunnel.o: in function `ec_i2c_xfer': i2c-cros-ec-tunnel.c:(.text+0x2fc): undefined reference to `cros_ec_cmd_xfer_status' ld: sound/soc/codecs/cros_ec_codec.o: in function `wov_host_event': cros_ec_codec.c:(.text+0x4fb): undefined reference to `cros_ec_get_host_event' ld: sound/soc/codecs/cros_ec_codec.o: in function `send_ec_host_command': cros_ec_codec.c:(.text+0x831): undefined reference to `cros_ec_cmd_xfer_status' This is because it will select SND_SOC_CROS_EC_CODEC and I2c_CROS_EC_TUNNEL but both depends on CROS_EC. Fixes: 6b8e4e7db3cd ("ASoC: amd: Add machine driver for Raven based platform") Reported-by: Randy Dunlap Signed-off-by: Enric Balletbo i Serra Link: https://lore.kernel.org/r/20200303110514.3267126-1-enric.balletbo@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index b29ef1373946cb..bce4cee5cb54a0 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -33,6 +33,6 @@ config SND_SOC_AMD_RV_RT5682_MACH select SND_SOC_MAX98357A select SND_SOC_CROS_EC_CODEC select I2C_CROS_EC_TUNNEL - depends on SND_SOC_AMD_ACP3x && I2C + depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC help This option enables machine driver for RT5682 and MAX9835. From d7729c40b376ab7c1ad4b0498a8bbf44bed3cbf4 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Tue, 3 Mar 2020 14:34:37 +0530 Subject: [PATCH 206/415] ASoC: amd: Fix compile warning of argument type Fixes: >> sound/soc//amd/acp3x-rt5682-max9836.c:341:23: warning: format '%d' >> expects argument of type 'int', but argument 3 has type 'long int' >> [-Wformat=] dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", Reported-by: kbuild test robot Signed-off-by: Akshu Agrawal Link: https://lore.kernel.org/r/20200303090444.95805-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1722aafb..521c9ab4c29c40 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -338,7 +338,7 @@ static int acp3x_probe(struct platform_device *pdev) dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); if (IS_ERR(dmic_sel)) { - dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", PTR_ERR(dmic_sel)); return PTR_ERR(dmic_sel); } From 14beaccc36dc9c1afbe6da627b873bf1d6849234 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 4 Mar 2020 16:40:57 +0800 Subject: [PATCH 207/415] ASoc: amd: acp3x: Add missing include gcc 7.4.0 build fails: In file included from sound/soc/amd/acp3x-rt5682-max9836.c:20:0: sound/soc/amd/raven/acp3x.h: In function rv_readl: sound/soc/amd/raven/acp3x.h:113:9: error: implicit declaration of function readl; did you mean rv_readl? [-Werror=implicit-function-declaration] return readl(base_addr - ACP3x_PHY_BASE_ADDRESS); ^~~~~ rv_readl sound/soc/amd/raven/acp3x.h: In function rv_writel: sound/soc/amd/raven/acp3x.h:118:2: error: implicit declaration of function writel; did you mean rv_writel? [-Werror=implicit-function-declaration] writel(val, base_addr - ACP3x_PHY_BASE_ADDRESS); ^~~~~~ rv_writel Add to fix this. Fixes: 6b8e4e7db3cd ("ASoC: amd: Add machine driver for Raven based platform") Reported-by: Hulk Robot Signed-off-by: YueHaibing Message-Id: <20200304084057.44764-1-yuehaibing@huawei.com> Signed-off-by: Mark Brown --- sound/soc/amd/acp3x-rt5682-max9836.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 521c9ab4c29c40..8f71c3f7ef792e 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include "raven/acp3x.h" From 2e4249f58074ec93746df3a902d1835b7edfef49 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Wed, 4 Mar 2020 13:34:27 -0600 Subject: [PATCH 208/415] ASoC: tlv320adcx140: Fix mic_bias and vref device tree verification Fix the range verification check for the mic_bias and vref device tree entries. Fixes 37bde5acf040 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200304193427.16886-1-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adcx140.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 93a0cb8e662ccf..38897568ee9699 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -748,9 +748,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) bias_source = ADCX140_MIC_BIAS_VAL_VREF; - if (bias_source != ADCX140_MIC_BIAS_VAL_VREF && - bias_source != ADCX140_MIC_BIAS_VAL_VREF_1096 && - bias_source != ADCX140_MIC_BIAS_VAL_AVDD) { + if (bias_source < ADCX140_MIC_BIAS_VAL_VREF || + bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } @@ -760,9 +759,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) vref_source = ADCX140_MIC_BIAS_VREF_275V; - if (vref_source != ADCX140_MIC_BIAS_VREF_275V && - vref_source != ADCX140_MIC_BIAS_VREF_25V && - vref_source != ADCX140_MIC_BIAS_VREF_1375V) { + if (vref_source < ADCX140_MIC_BIAS_VREF_275V || + vref_source > ADCX140_MIC_BIAS_VREF_1375V) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } From fd357ec595d36676c239d8d16706a270a961ac32 Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Thu, 5 Mar 2020 14:00:53 +0800 Subject: [PATCH 209/415] ASoC: sprd: Allow the MCDT driver to build into modules Change the config to 'tristate' for MCDT driver to allow it to build into modules, as well as changing to use IS_ENABLED() to validate if need supply dummy functions when building the MCDT driver as a module. Signed-off-by: Baolin Wang Link: https://lore.kernel.org/r/9306f2b99641136653ae4fe6cf9e859b7f698f77.1583387748.git.baolin.wang7@gmail.com Signed-off-by: Mark Brown --- sound/soc/sprd/Kconfig | 2 +- sound/soc/sprd/sprd-mcdt.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig index 5474fd3de8c006..5e0ac827857213 100644 --- a/sound/soc/sprd/Kconfig +++ b/sound/soc/sprd/Kconfig @@ -8,7 +8,7 @@ config SND_SOC_SPRD the Spreadtrum SoCs' Audio interfaces. config SND_SOC_SPRD_MCDT - bool "Spreadtrum multi-channel data transfer support" + tristate "Spreadtrum multi-channel data transfer support" depends on SND_SOC_SPRD help Say y here to enable multi-channel data transfer support. It diff --git a/sound/soc/sprd/sprd-mcdt.h b/sound/soc/sprd/sprd-mcdt.h index 9cc7e207ac7663..679e3af3baad4a 100644 --- a/sound/soc/sprd/sprd-mcdt.h +++ b/sound/soc/sprd/sprd-mcdt.h @@ -48,7 +48,7 @@ struct sprd_mcdt_chan { struct list_head list; }; -#ifdef CONFIG_SND_SOC_SPRD_MCDT +#if IS_ENABLED(CONFIG_SND_SOC_SPRD_MCDT) struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, enum sprd_mcdt_channel_type type); void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan); From 25c2f5156dd57f03aee2de079248c23a56222c92 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Feb 2020 10:54:38 +0900 Subject: [PATCH 210/415] ASoC: soc-pcm: use defined stream Many functions defines "stream = substream->stream", but some of them is using "substream->stream" instead of "stream". It is pointless. This patch uses defined stream. Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/87mu947q1t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 90857138c823d5..8c27eb4d5e4c6d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -644,8 +644,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * bailed out on a higher level, since there would be no * CODEC to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(codec_dai, - substream->stream)) + if (!snd_soc_dai_stream_valid(codec_dai, stream)) continue; codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream); @@ -2149,7 +2148,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - ret = dpcm_be_dai_startup(fe, fe_substream->stream); + ret = dpcm_be_dai_startup(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: failed to start some BEs %d\n", ret); goto be_err; @@ -2180,7 +2179,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) return 0; unwind: - dpcm_be_dai_startup_unwind(fe, fe_substream->stream); + dpcm_be_dai_startup_unwind(fe, stream); be_err: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; @@ -2234,7 +2233,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ - dpcm_be_dai_shutdown(fe, substream->stream); + dpcm_be_dai_shutdown(fe, stream); dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name); @@ -2412,9 +2411,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - memcpy(&fe->dpcm[substream->stream].hw_params, params, + memcpy(&fe->dpcm[stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); - ret = dpcm_be_dai_hw_params(fe, substream->stream); + ret = dpcm_be_dai_hw_params(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret); goto out; @@ -2736,7 +2735,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) goto out; } - ret = dpcm_be_dai_prepare(fe, substream->stream); + ret = dpcm_be_dai_prepare(fe, stream); if (ret < 0) goto out; From 6e02feb0d2663c1b7caa5e271c2a60e219f0ca07 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Feb 2020 10:54:48 +0900 Subject: [PATCH 211/415] ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() dpcm_add_paths() checks returned be from dpcm_get_be() static int dpcm_add_paths(...) { ... for_each_dapm_widgets(list, i, widget) { ... be = dpcm_get_be(...); ... /* make sure BE is a real BE */ => if (!be->dai_link->no_pcm) continue; ... } ... } But, dpcm_get_be() itself is checking it already. dpcm_get_be(...) { ... for_each_card_rtds(card, be) { => if (!be->dai_link->no_pcm) continue; ... if (...) => return be; } return NULL } This patch removes duplicate check Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/87lfoo7q1j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8c27eb4d5e4c6d..e3a2c4f7757bfa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1690,10 +1690,6 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; } - /* make sure BE is a real BE */ - if (!be->dai_link->no_pcm) - continue; - /* don't connect if FE is not running */ if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; From 66db29588dd6c55591b0f93dc8044a018d78d501 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 25 Feb 2020 21:32:49 -0800 Subject: [PATCH 212/415] ALSA: korg1212: fix if-statement empty body warnings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix gcc warnings when -Wextra is used by using an empty do-while block instead of . Fixes these build warnings: ../sound/pci/korg1212/korg1212.c:674:44: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:708:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:730:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:853:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1013:44: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1035:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1052:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1066:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1087:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1094:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1208:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:2360:102: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] Signed-off-by: Randy Dunlap Link: https://lore.kernel.org/r/91fb1e97-a773-5790-3f65-8198403341e1@infradead.org Signed-off-by: Takashi Iwai --- sound/pci/korg1212/korg1212.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 21ab9cc50c7152..65a887b217ee50 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -30,7 +30,7 @@ #if K1212_DEBUG_LEVEL > 0 #define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) #else -#define K1212_DEBUG_PRINTK(fmt,...) +#define K1212_DEBUG_PRINTK(fmt,...) do { } while (0) #endif #if K1212_DEBUG_LEVEL > 1 #define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args) From 2edb84e3047b93da2f2b234219cdc304df042d9e Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Sat, 29 Feb 2020 18:18:15 +0300 Subject: [PATCH 213/415] ALSA: usb-audio: Add support for MOTU MicroBook IIc MicroBook IIc operates in UAC2 mode by default. This patch addresses several issues with it: - MicroBook II and IIc shares the same USB ID. We can distinguish them by interface class. - MaxPacketsOnly attribute is erroneously set in endpoint descriptors. As a result this card produces noise with all sample rates other than 96 KHz. This also causes issues like IOMMU page faults and other problems with host controller. - Sample rate changes takes more than 2 seconds for this device. Clock validity request returns false during that period, so the clock validity quirk is required. Signed-off-by: Alexander Tsoy Link: https://lore.kernel.org/r/20200229151815.14199-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 59 ++++++++++++++++++++++++++++++++-------- sound/usb/pcm.c | 7 ++++- sound/usb/quirks-table.h | 2 +- sound/usb/quirks.c | 18 +++++++++++- 4 files changed, 72 insertions(+), 14 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index a48313dfa967a1..b118cf97607f3d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -151,16 +151,15 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } -/* - * Assume the clock is valid if clock source supports only one single sample - * rate, the terminal is connected directly to it (there is no clock selector) - * and clock type is internal. This is to deal with some Denon DJ controllers - * that always reports that clock is invalid. - */ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, int source_id) { + bool ret = false; + int count; + unsigned char data; + struct usb_device *dev = chip->dev; + if (fmt->protocol == UAC_VERSION_2) { struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); @@ -168,13 +167,51 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, if (!cs_desc) return false; - return (fmt->nr_rates == 1 && - (fmt->clock & 0xff) == cs_desc->bClockID && - (cs_desc->bmAttributes & 0x3) != - UAC_CLOCK_SOURCE_TYPE_EXT); + /* + * Assume the clock is valid if clock source supports only one + * single sample rate, the terminal is connected directly to it + * (there is no clock selector) and clock type is internal. + * This is to deal with some Denon DJ controllers that always + * reports that clock is invalid. + */ + if (fmt->nr_rates == 1 && + (fmt->clock & 0xff) == cs_desc->bClockID && + (cs_desc->bmAttributes & 0x3) != + UAC_CLOCK_SOURCE_TYPE_EXT) + return true; + } + + /* + * MOTU MicroBook IIc + * Sample rate changes takes more than 2 seconds for this device. Clock + * validity request returns false during that period. + */ + if (chip->usb_id == USB_ID(0x07fd, 0x0004)) { + count = 0; + + while ((!ret) && (count < 50)) { + int err; + + msleep(100); + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_CLOCK_VALID << 8, + snd_usb_ctrl_intf(chip) | (source_id << 8), + &data, sizeof(data)); + if (err < 0) { + dev_warn(&dev->dev, + "%s(): cannot get clock validity for id %d\n", + __func__, source_id); + return false; + } + + ret = !!data; + count++; + } } - return false; + return ret; } static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bd258f1ec2dd3c..a4e4064f9aee7b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -357,7 +357,12 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; - case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */ + /* MicroBook IIc */ + if (altsd->bInterfaceClass == USB_CLASS_AUDIO) + return 0; + + /* MicroBook II */ ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db0c..1c8719292eee41 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3472,7 +3472,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), }, /* MOTU Microbook II */ { - USB_DEVICE(0x07fd, 0x0004), + USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "MOTU", .product_name = "MicroBookII", diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 915aea256f65c3..3cc745fe24d8dd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1352,7 +1352,15 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ - return snd_usb_motu_microbookii_boot_quirk(dev); + /* + * For some reason interface 3 with vendor-spec class is + * detected on MicroBook IIc. + */ + if (get_iface_desc(intf->altsetting)->bInterfaceClass == + USB_CLASS_VENDOR_SPEC && + get_iface_desc(intf->altsetting)->bInterfaceNumber < 3) + return snd_usb_motu_microbookii_boot_quirk(dev); + break; } return 0; @@ -1790,5 +1798,13 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, else fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; break; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook IIc */ + /* + * MaxPacketsOnly attribute is erroneously set in endpoint + * descriptors. As a result this card produces noise with + * all sample rates other than 96 KHz. + */ + fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; + break; } } From d0ee674bb5d3787ca0122c5ae8e52680de0e7c52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2020 09:12:31 +0100 Subject: [PATCH 214/415] ALSA: usb-audio: Fix missing braces in some struct inits The struct s1810c_state_packet contains the array in the first field hence zero-initialization requires a more couple of braces. Fix the compile warning pointing it out: sound/usb/mixer_s1810c.c: In function 'snd_sc1810c_get_status_field': sound/usb/mixer_s1810c.c:178:9: warning: missing braces around initializer [-Wmissing-braces] Reported-by: kbuild test robot Fixes: 8dc5efe3d17c ("ALSA: usb-audio: Add support for Presonus Studio 1810c") Link: https://lore.kernel.org/r/202002210251.WgMfvKJP%lkp@intel.com Link: https://lore.kernel.org/r/20200306081231.7940-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer_s1810c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index 816879a07f82f0..6483e47bafd013 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -175,8 +175,8 @@ static int snd_sc1810c_get_status_field(struct usb_device *dev, u32 *field, int field_idx, uint16_t *seqnum) { - struct s1810c_state_packet pkt_out = { 0 }; - struct s1810c_state_packet pkt_in = { 0 }; + struct s1810c_state_packet pkt_out = { { 0 } }; + struct s1810c_state_packet pkt_in = { { 0 } }; int ret = 0; pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; From 820766c1e16651b46bfb771afae8d789da1986cf Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 6 Mar 2020 13:28:05 +0000 Subject: [PATCH 215/415] ASoC: wcd934x: fix High Accuracy Buck enable High Accuracy buck is not applicable when we use RCO Band Gap source, so move it back to correct place. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200306132806.19684-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index aefaadfba8a106..83d643a0777505 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1202,11 +1202,6 @@ static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src) regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO, WCD934X_ANA_RCO_BG_EN_MASK, 0); usleep_range(100, 110); - } else if (sido_src == SIDO_SOURCE_RCO_BG) { - regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO, - WCD934X_ANA_RCO_BG_EN_MASK, - WCD934X_ANA_RCO_BG_ENABLE); - usleep_range(100, 110); regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL, WCD934X_ANA_BUCK_PRE_EN1_MASK, WCD934X_ANA_BUCK_PRE_EN1_ENABLE); @@ -1219,6 +1214,11 @@ static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src) WCD934X_ANA_BUCK_HI_ACCU_EN_MASK, WCD934X_ANA_BUCK_HI_ACCU_ENABLE); usleep_range(100, 110); + } else if (sido_src == SIDO_SOURCE_RCO_BG) { + regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO, + WCD934X_ANA_RCO_BG_EN_MASK, + WCD934X_ANA_RCO_BG_ENABLE); + usleep_range(100, 110); } wcd->sido_input_src = sido_src; From e0e247d593f78f4ac5647a9ef2c6db8f918ecbdc Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 6 Mar 2020 13:28:06 +0000 Subject: [PATCH 216/415] ASoC: wcd934x: remove unused headers Looks like there are some unused headers, remove them. Seems to be missed while moving to mfd. Reported-by: Stephen Boyd Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200306132806.19684-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 83d643a0777505..5269857e27462c 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -3,7 +3,6 @@ #include #include -#include #include #include #include @@ -11,10 +10,7 @@ #include #include #include -#include -#include #include -#include #include #include #include From 4769bfb9dada678b31a2ec275372624dbfeed9d1 Mon Sep 17 00:00:00 2001 From: Samuel Holland Date: Wed, 4 Mar 2020 23:11:41 -0600 Subject: [PATCH 217/415] ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates It can be useful to derive min/max rates of a snd_pcm_hardware without having a snd_pcm_runtime, such as before constructing an ASoC DAI link. Create a new helper that takes a pointer to a snd_pcm_hardware directly, and refactor the original function as a wrapper around it, to avoid needing to update any call sites. Signed-off-by: Samuel Holland Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200305051143.60691-2-samuel@sholland.org Signed-off-by: Mark Brown --- include/sound/pcm.h | 9 ++++++++- sound/core/pcm_misc.c | 18 +++++++++--------- 2 files changed, 17 insertions(+), 10 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2628246b76fa76..f7a95b7111007e 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1127,7 +1127,14 @@ snd_pcm_kernel_readv(struct snd_pcm_substream *substream, return __snd_pcm_lib_xfer(substream, bufs, false, frames, true); } -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw); + +static inline int +snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) +{ + return snd_pcm_hw_limit_rates(&runtime->hw); +} + unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a, diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index a6a5415115343e..5dd2e533590027 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -474,32 +474,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int EXPORT_SYMBOL(snd_pcm_format_set_silence); /** - * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields - * @runtime: the runtime instance + * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields + * @hw: the pcm hw instance * * Determines the rate_min and rate_max fields from the rates bits of - * the given runtime->hw. + * the given hw. * * Return: Zero if successful. */ -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw) { int i; for (i = 0; i < (int)snd_pcm_known_rates.count; i++) { - if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_min = snd_pcm_known_rates.list[i]; + if (hw->rates & (1 << i)) { + hw->rate_min = snd_pcm_known_rates.list[i]; break; } } for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) { - if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_max = snd_pcm_known_rates.list[i]; + if (hw->rates & (1 << i)) { + hw->rate_max = snd_pcm_known_rates.list[i]; break; } } return 0; } -EXPORT_SYMBOL(snd_pcm_limit_hw_rates); +EXPORT_SYMBOL(snd_pcm_hw_limit_rates); /** * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit From 5854a46486ad5b8d73766735fb0d77f05956b22c Mon Sep 17 00:00:00 2001 From: Samuel Holland Date: Wed, 4 Mar 2020 23:11:42 -0600 Subject: [PATCH 218/415] ASoC: pcm: Export parameter intersection logic The logic to calculate the subset of stream parameters supported by all DAIs associated with a PCM stream is nontrivial. Export a helper function so it can be used to set up simple codec2codec DAI links. Signed-off-by: Samuel Holland Link: https://lore.kernel.org/r/20200305051143.60691-3-samuel@sholland.org Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 56 ++++++++++++++++++++++++++++++--------------- 2 files changed, 41 insertions(+), 18 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 81e5d17be9351d..9543d9246ca450 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -471,6 +471,9 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd); void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream); void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream); +int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hardware *hw, int stream); + int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e3a2c4f7757bfa..de4226357e2b05 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -587,11 +587,18 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) soc_pcm_set_msb(substream, cpu_bits); } -static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) +/** + * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream + * @rtd: ASoC PCM runtime + * @hw: PCM hardware parameters (output) + * @stream: Direction of the PCM stream + * + * Calculates the subset of stream parameters supported by all DAIs + * associated with the PCM stream. + */ +int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hardware *hw, int stream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_pcm_hardware *hw = &runtime->hw; - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; struct snd_soc_pcm_stream *codec_stream; @@ -602,7 +609,6 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX; unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX; u64 formats = ULLONG_MAX; - int stream = substream->stream; int i; /* first calculate min/max only for CPUs in the DAI link */ @@ -613,12 +619,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * Otherwise, since the rate, channel, and format values will * zero in that case, we would have no usable settings left, * causing the resulting setup to fail. - * At least one CPU should match, otherwise we should have - * bailed out on a higher level, since there would be no - * CPU to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(cpu_dai, - substream->stream)) + if (!snd_soc_dai_stream_valid(cpu_dai, stream)) continue; cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream); @@ -640,9 +642,6 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * Otherwise, since the rate, channel, and format values will * zero in that case, we would have no usable settings left, * causing the resulting setup to fail. - * At least one CODEC should match, otherwise we should have - * bailed out on a higher level, since there would be no - * CODEC to support the transfer direction in that case. */ if (!snd_soc_dai_stream_valid(codec_dai, stream)) continue; @@ -657,6 +656,10 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates); } + /* Verify both a valid CPU DAI and a valid CODEC DAI were found */ + if (!chan_min || !cpu_chan_min) + return -EINVAL; + /* * chan min/max cannot be enforced if there are multiple CODEC DAIs * connected to CPU DAI(s), use CPU DAI's directly and let @@ -670,18 +673,35 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* finally find a intersection between CODECs and CPUs */ hw->channels_min = max(chan_min, cpu_chan_min); hw->channels_max = min(chan_max, cpu_chan_max); - if (hw->formats) - hw->formats &= formats; - else - hw->formats = formats; + hw->formats = formats; hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates); - snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_limit_rates(hw); hw->rate_min = max(hw->rate_min, cpu_rate_min); hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max); hw->rate_max = min_not_zero(hw->rate_max, rate_max); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_runtime_calc_hw); + +static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) +{ + struct snd_pcm_hardware *hw = &substream->runtime->hw; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + u64 formats = hw->formats; + + /* + * At least one CPU and one CODEC should match. Otherwise, we should + * have bailed out on a higher level, since there would be no CPU or + * CODEC to support the transfer direction in that case. + */ + snd_soc_runtime_calc_hw(rtd, hw, substream->stream); + + if (formats) + hw->formats &= formats; } static int soc_pcm_components_open(struct snd_pcm_substream *substream) From 95cfc0a0aaf575207152dd7601e782702565a6f1 Mon Sep 17 00:00:00 2001 From: Samuel Holland Date: Wed, 4 Mar 2020 23:11:43 -0600 Subject: [PATCH 219/415] ASoC: simple-card: Add support for codec2codec DAI links Following the example in cb2cf0de1174 ("ASoC: soc-core: care Codec <-> Codec case by non_legacy_dai_naming"), determine if a DAI link contains only codec DAIs by examining the non_legacy_dai_naming flag in each DAI's component. For now, we assume there is only one or a small set of valid PCM stream parameters, so num_params == 1 is good enough. We also assume that the same params are valid for all supported streams. params is set to the subset of parameters common among all DAIs, and then the existing code automatically chooses the highest quality of the remaining values when the link is brought up. Signed-off-by: Samuel Holland Link: https://lore.kernel.org/r/20200305051143.60691-4-samuel@sholland.org Signed-off-by: Mark Brown --- Documentation/sound/soc/codec-to-codec.rst | 9 +++- sound/soc/generic/simple-card-utils.c | 48 ++++++++++++++++++++++ 2 files changed, 55 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst index 810109d7500dfe..4eaa9a0c41fc55 100644 --- a/Documentation/sound/soc/codec-to-codec.rst +++ b/Documentation/sound/soc/codec-to-codec.rst @@ -104,5 +104,10 @@ Make sure to name your corresponding cpu and codec playback and capture dai names ending with "Playback" and "Capture" respectively as dapm core will link and power those dais based on the name. -Note that in current device tree there is no way to mark a dai_link -as codec to codec. However, it may change in future. +A dai_link in a "simple-audio-card" will automatically be detected as +codec to codec when all DAIs on the link belong to codec components. +The dai_link will be initialized with the subset of stream parameters +(channels, format, sample rate) supported by all DAIs on the link. Since +there is no way to provide these parameters in the device tree, this is +mostly useful for communication with simple fixed-function codecs, such +as a Bluetooth controller or cellular modem. diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 9b794775df537a..320e648f749914 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -331,6 +331,50 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; } +static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, + struct simple_dai_props *dai_props) +{ + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_component *component; + struct snd_soc_pcm_stream *params; + struct snd_pcm_hardware hw; + int i, ret, stream; + + /* Only codecs should have non_legacy_dai_naming set. */ + for_each_rtd_components(rtd, i, component) { + if (!component->driver->non_legacy_dai_naming) + return 0; + } + + /* Assumes the capabilities are the same for all supported streams */ + for (stream = 0; stream < 2; stream++) { + ret = snd_soc_runtime_calc_hw(rtd, &hw, stream); + if (ret == 0) + break; + } + + if (ret < 0) { + dev_err(rtd->dev, "simple-card: no valid dai_link params\n"); + return ret; + } + + params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + + params->formats = hw.formats; + params->rates = hw.rates; + params->rate_min = hw.rate_min; + params->rate_max = hw.rate_max; + params->channels_min = hw.channels_min; + params->channels_max = hw.channels_max; + + dai_link->params = params; + dai_link->num_params = 1; + + return 0; +} + int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) { struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); @@ -347,6 +391,10 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + ret = asoc_simple_init_dai_link_params(rtd, dai_props); + if (ret < 0) + return ret; + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_dai_init); From 30fca26f8e2277ccd14fe3277a330b4f21cadca7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 6 Mar 2020 10:09:44 +0900 Subject: [PATCH 220/415] ASoC: soc-pcm: move dpcm_fe_dai_close() move dpcm_fe_dai_close() next to dpcm_fe_dai_open(). This is prepare for dpcm_fe_dai_open() cleanup Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/87pndqp9uv.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e3a2c4f7757bfa..3686dda097e273 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2978,6 +2978,26 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) return ret; } +static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = fe_substream->stream, ret; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + ret = dpcm_fe_dai_shutdown(fe_substream); + + /* mark FE's links ready to prune */ + for_each_dpcm_be(fe, stream, dpcm) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + + fe->dpcm[stream].runtime = NULL; + mutex_unlock(&fe->card->mutex); + return ret; +} + static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; @@ -3017,26 +3037,6 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) return ret; } -static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) -{ - struct snd_soc_pcm_runtime *fe = fe_substream->private_data; - struct snd_soc_dpcm *dpcm; - int stream = fe_substream->stream, ret; - - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - ret = dpcm_fe_dai_shutdown(fe_substream); - - /* mark FE's links ready to prune */ - for_each_dpcm_be(fe, stream, dpcm) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - - fe->dpcm[stream].runtime = NULL; - mutex_unlock(&fe->card->mutex); - return ret; -} - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { From 265694b67c13f00384bd0b97549b4681cbcc85af Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 6 Mar 2020 10:09:49 +0900 Subject: [PATCH 221/415] ASoC: soc-pcm: add dpcm_fe_dai_cleanup() dpcm_fe_dai_close() and error case of dpcm_fe_dai_open() need to do same cleanup operation. To avoid duplicate code, this patch adds dpcm_fe_dai_cleanup() and use it. Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/87o8tap9uq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 30 ++++++++++++++++-------------- 1 file changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3686dda097e273..b405fb3a181b92 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2978,14 +2978,11 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) return ret; } -static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +static void dpcm_fe_dai_cleanup(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_soc_dpcm *dpcm; - int stream = fe_substream->stream, ret; - - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - ret = dpcm_fe_dai_shutdown(fe_substream); + int stream = fe_substream->stream; /* mark FE's links ready to prune */ for_each_dpcm_be(fe, stream, dpcm) @@ -2994,6 +2991,18 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) dpcm_be_disconnect(fe, stream); fe->dpcm[stream].runtime = NULL; +} + +static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + int ret; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + ret = dpcm_fe_dai_shutdown(fe_substream); + + dpcm_fe_dai_cleanup(fe_substream); + mutex_unlock(&fe->card->mutex); return ret; } @@ -3001,7 +3010,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; - struct snd_soc_dpcm *dpcm; struct snd_soc_dapm_widget_list *list; int ret; int stream = fe_substream->stream; @@ -3021,14 +3029,8 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) dpcm_process_paths(fe, stream, &list, 1); ret = dpcm_fe_dai_startup(fe_substream); - if (ret < 0) { - /* clean up all links */ - for_each_dpcm_be(fe, stream, dpcm) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - fe->dpcm[stream].runtime = NULL; - } + if (ret < 0) + dpcm_fe_dai_cleanup(fe_substream); dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); From 0c9ba720f0be457443ba89b09a5198616cd3e811 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 6 Mar 2020 10:09:54 +0900 Subject: [PATCH 222/415] ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() We already have snd_soc_dai_get_pcm_stream(), let's use it Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/87mu8up9ul.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b405fb3a181b92..3a30776858bf35 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2022,7 +2022,6 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai; - struct snd_soc_dai_driver *cpu_dai_drv; int i; for_each_rtd_cpu_dai(rtd, i, cpu_dai) { @@ -2033,11 +2032,9 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) continue; - cpu_dai_drv = cpu_dai->driver; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); - else - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); + dpcm_init_runtime_hw(runtime, + snd_soc_dai_get_pcm_stream(cpu_dai, + substream->stream)); } dpcm_runtime_merge_format(substream, &runtime->hw.formats); From 8a01fbf0ac115268293d8764850edc0628a58e4f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 6 Mar 2020 10:09:59 +0900 Subject: [PATCH 223/415] ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() error handling at dpcm_fe_dai_startup() has duplicate code. This patch tidyup it. Signed-off-by: Kuninori Morimoto Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/87lfoep9ug.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3a30776858bf35..7d787e0966f335 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2162,17 +2162,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) snd_pcm_limit_hw_rates(runtime); ret = dpcm_apply_symmetry(fe_substream, stream); - if (ret < 0) { + if (ret < 0) dev_err(fe->dev, "ASoC: failed to apply dpcm symmetry %d\n", ret); - goto unwind; - } - - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - return 0; unwind: - dpcm_be_dai_startup_unwind(fe, stream); + if (ret < 0) + dpcm_be_dai_startup_unwind(fe, stream); be_err: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; From 67ad877757cea329f74c1e169ec54131c3f223ce Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 6 Mar 2020 10:10:04 +0900 Subject: [PATCH 224/415] ASoC: soc-pcm: check DAI's activity more simply soc_pcm_hw_free() want to call snd_soc_dai_digital_mute() if it was last user of Playback or Capture. bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int playback_active = dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]; int capture_active = dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]; if ((playback && playback_active == 1) || (!playback && capture_active == 1)) snd_soc_dai_digital_mute(...) But it is same as int active = dai->stream_active[substream->stream]; if (active == 1) snd_soc_dai_digital_mute(...) This patch simplify the code. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87k13yp9ub.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7d787e0966f335..af0e17bfeeab4b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1202,7 +1202,6 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; - bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -1226,11 +1225,9 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* apply codec digital mute */ for_each_rtd_codec_dai(rtd, i, codec_dai) { - int playback_active = codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]; - int capture_active = codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]; + int active = codec_dai->stream_active[substream->stream]; - if ((playback && playback_active == 1) || - (!playback && capture_active == 1)) + if (active == 1) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); } From a9ee331b537a3dfe6778fa4e07c0801f33e474f5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 6 Mar 2020 10:10:17 +0900 Subject: [PATCH 225/415] ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. Digital Mute for CPU is done at soc_pcm_close(), and Digital Mute for Codec is done at soc_pcm_hw_free(). It is just confusable. This patch do Digital Mute for both CPU/Codec in same timing. Then, it cares DAI activity Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87imjip9ty.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index af0e17bfeeab4b..90d26fccb0da32 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -760,9 +760,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) snd_soc_runtime_deactivate(rtd, substream->stream); - for_each_rtd_cpu_dai(rtd, i, cpu_dai) - snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream); - for_each_rtd_cpu_dai(rtd, i, cpu_dai) snd_soc_dai_shutdown(cpu_dai, substream); @@ -1232,6 +1229,14 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) substream->stream); } + for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + int active = cpu_dai->stream_active[substream->stream]; + + if (active == 1) + snd_soc_dai_digital_mute(cpu_dai, 1, + substream->stream); + } + /* free any machine hw params */ soc_rtd_hw_free(rtd, substream); From 9c0d16ac059148fc7647f5f9e90df6f34d3439f0 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 6 Mar 2020 22:52:29 +0900 Subject: [PATCH 226/415] ALSA: firewire: use KBUILD_MODNAME for struct driver.name instead of string KBUILD_MODNAME is available to name kernel modules according to its object name. This commit uses the macro instead of string for name field of struct driver since drivers in ALSA firewire stack have the same name of each object name. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20200306135229.11659-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/digi00x/digi00x.c | 2 +- sound/firewire/fireface/ff.c | 2 +- sound/firewire/fireworks/fireworks.c | 2 +- sound/firewire/tascam/tascam.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 976d8cb9a34f07..2c8e3392a4903a 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -509,7 +509,7 @@ MODULE_DEVICE_TABLE(ieee1394, bebob_id_table); static struct fw_driver bebob_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-bebob", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = bebob_probe, diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 1f5fc0e7c02439..c84b913a9fe010 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -192,7 +192,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table); static struct fw_driver dg00x_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-firewire-digi00x", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_dg00x_probe, diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c index f5a016560eb8e7..b62a4fd2240762 100644 --- a/sound/firewire/fireface/ff.c +++ b/sound/firewire/fireface/ff.c @@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_ff_id_table); static struct fw_driver ff_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-fireface", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_ff_probe, diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 134fc9ee26b90f..b1cc013a354094 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -362,7 +362,7 @@ MODULE_DEVICE_TABLE(ieee1394, efw_id_table); static struct fw_driver efw_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-fireworks", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = efw_probe, diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index addc464503bcf7..5dac0d9fc58e51 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table); static struct fw_driver tscm_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-firewire-tascam", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_tscm_probe, From f9c23615c688270d2a383bd752f7a54a7137d596 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 27 Feb 2020 11:35:44 +0200 Subject: [PATCH 227/415] ALSA: dmaengine_pcm: No need to take runtime reference twice in pcm_pointer The runtime pointer has been taken in functional level so there is no need to take it again under the if () case. Fixes: 9d789dc047e3 ("ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback") Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20200227093544.27723-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/core/pcm_dmaengine.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6852bb670b4ef5..9d4f48cfe47f12 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -248,8 +248,6 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { - struct snd_pcm_runtime *runtime = substream->runtime; - buf_size = snd_pcm_lib_buffer_bytes(substream); if (state.residue > 0 && state.residue <= buf_size) pos = buf_size - state.residue; From d902e7856d2a3b5da7acab90e5faec22e395e57a Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 6 Mar 2020 15:26:33 +0000 Subject: [PATCH 228/415] ASoC: wcd9335: fix address map representation slimbus addresses are 16 bit wide, masking page numbers to wcd register at offset of 12 will limit the number for pages. So it becomes impossible to write to page 0x10 registers. Remove masking 0x800 (slimbus address range) from register address and making use of window parameters in regmap config should fix it and also will represent the registers exactly inline with Datasheet. Remove this unnessary masking and make the registers be inline with datasheet. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200306152633.25836-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 18 +++++++++--------- sound/soc/codecs/wcd9335.h | 7 ++++--- 2 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f11ffa28683b11..700cc12127705d 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4926,11 +4926,11 @@ static const struct regmap_range_cfg wcd9335_ranges[] = { .name = "WCD9335", .range_min = 0x0, .range_max = WCD9335_MAX_REGISTER, - .selector_reg = WCD9335_REG(0x0, 0), + .selector_reg = WCD9335_SEL_REGISTER, .selector_mask = 0xff, .selector_shift = 0, - .window_start = 0x0, - .window_len = 0x1000, + .window_start = 0x800, + .window_len = 0x100, }, }; @@ -4968,12 +4968,12 @@ static const struct regmap_range_cfg wcd9335_ifc_ranges[] = { { .name = "WCD9335-IFC-DEV", .range_min = 0x0, - .range_max = WCD9335_REG(0, 0x7ff), - .selector_reg = WCD9335_REG(0, 0x0), - .selector_mask = 0xff, + .range_max = WCD9335_MAX_REGISTER, + .selector_reg = WCD9335_SEL_REGISTER, + .selector_mask = 0xfff, .selector_shift = 0, - .window_start = 0x0, - .window_len = 0x1000, + .window_start = 0x800, + .window_len = 0x400, }, }; @@ -4981,7 +4981,7 @@ static struct regmap_config wcd9335_ifc_regmap_config = { .reg_bits = 16, .val_bits = 8, .can_multi_write = true, - .max_register = WCD9335_REG(0, 0x7FF), + .max_register = WCD9335_MAX_REGISTER, .ranges = wcd9335_ifc_ranges, .num_ranges = ARRAY_SIZE(wcd9335_ifc_ranges), }; diff --git a/sound/soc/codecs/wcd9335.h b/sound/soc/codecs/wcd9335.h index 4d9be2496c3050..72060824c7436f 100644 --- a/sound/soc/codecs/wcd9335.h +++ b/sound/soc/codecs/wcd9335.h @@ -8,9 +8,9 @@ * in slimbus mode the reg base starts from 0x800 * in i2s/i2c mode the reg base is 0x0 */ -#define WCD9335_REG(pg, r) ((pg << 12) | (r) | 0x800) +#define WCD9335_REG(pg, r) ((pg << 8) | (r)) #define WCD9335_REG_OFFSET(r) (r & 0xFF) -#define WCD9335_PAGE_OFFSET(r) ((r >> 12) & 0xFF) +#define WCD9335_PAGE_OFFSET(r) ((r >> 8) & 0xFF) /* Page-0 Registers */ #define WCD9335_PAGE0_PAGE_REGISTER WCD9335_REG(0x00, 0x000) @@ -600,7 +600,8 @@ #define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE BIT(0) #define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_DISABLE 0 #define WCD9335_CDC_TOP_TOP_CFG1 WCD9335_REG(0x0d, 0x082) -#define WCD9335_MAX_REGISTER WCD9335_REG(0x80, 0x0FF) +#define WCD9335_MAX_REGISTER 0xffff +#define WCD9335_SEL_REGISTER 0x800 /* SLIMBUS Slave Registers */ #define WCD9335_SLIM_PGD_PORT_INT_EN0 WCD9335_REG(0, 0x30) From 9401d5aa328e64617d87abd59af1c91cace4c3e4 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 6 Mar 2020 23:29:27 +0100 Subject: [PATCH 229/415] ASoC: jz4740-i2s: Fix divider written at incorrect offset in register The 4-bit divider value was written at offset 8, while the jz4740 programming manual locates it at offset 0. Fixes: 26b0aad80a86 ("ASoC: jz4740: Add dynamic sampling rate support to jz4740-i2s") Signed-off-by: Paul Cercueil Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20200306222931.39664-2-paul@crapouillou.net Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 9d540588120918..434737b2b2b2e2 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -83,7 +83,7 @@ #define JZ_AIC_I2S_STATUS_BUSY BIT(2) #define JZ_AIC_CLK_DIV_MASK 0xf -#define I2SDIV_DV_SHIFT 8 +#define I2SDIV_DV_SHIFT 0 #define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT) #define I2SDIV_IDV_SHIFT 8 #define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT) From 40a92dbcbc32d7dfbf186dfb1e27ee55d1df2f64 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:02:37 +0900 Subject: [PATCH 230/415] ASoC: simple-card-utils: use for_each_pcm_streams() We already have for_each_pcm_streams() macro. Let's use it. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/875zfei3aa.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 320e648f749914..abbdf1054f6fa2 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -347,7 +347,7 @@ static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, } /* Assumes the capabilities are the same for all supported streams */ - for (stream = 0; stream < 2; stream++) { + for_each_pcm_streams(stream) { ret = snd_soc_runtime_calc_hw(rtd, &hw, stream); if (ret == 0) break; From ab985be95da1d68a0cdba1ed702961aae047bf89 Mon Sep 17 00:00:00 2001 From: Ravulapati Vishnu vardhan rao Date: Mon, 9 Mar 2020 16:20:10 +0530 Subject: [PATCH 231/415] ASoC: amd: Adding TDM support in hw_params. TDM related settings for ACP registers in hw_params. When TDM mode is enabled, Hw_params needs to read and write from/to respective TX/RX (ACP_(I2S/BT)TDM_(TX/RX)FRMT) registers. Signed-off-by: Ravulapati Vishnu vardhan rao Link: https://lore.kernel.org/r/1583751029-2850-1-git-send-email-Vishnuvardhanrao.Ravulapati@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/raven/acp3x-i2s.c | 44 ++++++++++----------------------- 1 file changed, 13 insertions(+), 31 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 91a388184e525d..3a3c47e820ab9b 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -42,7 +42,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, u32 rx_mask, int slots, int slot_width) { struct i2s_dev_data *adata; - u32 val, reg_val, frmt_reg, frm_len; + u32 frm_len; u16 slot_len; adata = snd_soc_dai_get_drvdata(cpu_dai); @@ -64,36 +64,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } - - /* Enable I2S/BT channels TDM, respective TX/RX frame lengths.*/ - frm_len = FRM_LEN | (slots << 15) | (slot_len << 18); - if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) { - switch (adata->i2s_instance) { - case I2S_BT_INSTANCE: - reg_val = mmACP_BTTDM_ITER; - frmt_reg = mmACP_BTTDM_TXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = mmACP_I2STDM_ITER; - frmt_reg = mmACP_I2STDM_TXFRMT; - } - } else { - switch (adata->i2s_instance) { - case I2S_BT_INSTANCE: - reg_val = mmACP_BTTDM_IRER; - frmt_reg = mmACP_BTTDM_RXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = mmACP_I2STDM_IRER; - frmt_reg = mmACP_I2STDM_RXFRMT; - } - } - val = rv_readl(adata->acp3x_base + reg_val); - rv_writel(val | 0x2, adata->acp3x_base + reg_val); - rv_writel(frm_len, adata->acp3x_base + frmt_reg); adata->tdm_fmt = frm_len; return 0; } @@ -105,12 +76,14 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *prtd; struct snd_soc_card *card; struct acp3x_platform_info *pinfo; + struct i2s_dev_data *adata; u32 val; - u32 reg_val; + u32 reg_val, frmt_reg; prtd = substream->private_data; rtd = substream->runtime->private_data; card = prtd->card; + adata = snd_soc_dai_get_drvdata(dai); pinfo = snd_soc_card_get_drvdata(card); if (pinfo) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -141,21 +114,30 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_val = mmACP_BTTDM_ITER; + frmt_reg = mmACP_BTTDM_TXFRMT; break; case I2S_SP_INSTANCE: default: reg_val = mmACP_I2STDM_ITER; + frmt_reg = mmACP_I2STDM_TXFRMT; } } else { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_val = mmACP_BTTDM_IRER; + frmt_reg = mmACP_BTTDM_RXFRMT; break; case I2S_SP_INSTANCE: default: reg_val = mmACP_I2STDM_IRER; + frmt_reg = mmACP_I2STDM_RXFRMT; } } + if (adata->tdm_mode) { + val = rv_readl(rtd->acp3x_base + reg_val); + rv_writel(val | 0x2, rtd->acp3x_base + reg_val); + rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg); + } val = rv_readl(rtd->acp3x_base + reg_val); val = val | (rtd->xfer_resolution << 3); rv_writel(val, rtd->acp3x_base + reg_val); From a42d9ba15cbf3e84307906db65fc598a8b73e2f1 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 6 Mar 2020 23:29:28 +0100 Subject: [PATCH 232/415] ASoC: jz4740-i2s: Add local dev variable in probe function Make the code cleaner by using a "struct device *dev" variable instead of dereferencing it everytime from within the struct platform_device. Signed-off-by: Paul Cercueil Link: https://lore.kernel.org/r/20200306222931.39664-3-paul@crapouillou.net Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 9d540588120918..2572aba843e3b2 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -492,45 +492,45 @@ MODULE_DEVICE_TABLE(of, jz4740_of_matches); static int jz4740_i2s_dev_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct jz4740_i2s *i2s; struct resource *mem; int ret; - i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + i2s = devm_kzalloc(dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; - i2s->version = - (enum jz47xx_i2s_version)of_device_get_match_data(&pdev->dev); + i2s->version = (enum jz47xx_i2s_version)of_device_get_match_data(dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - i2s->base = devm_ioremap_resource(&pdev->dev, mem); + i2s->base = devm_ioremap_resource(dev, mem); if (IS_ERR(i2s->base)) return PTR_ERR(i2s->base); i2s->phys_base = mem->start; - i2s->clk_aic = devm_clk_get(&pdev->dev, "aic"); + i2s->clk_aic = devm_clk_get(dev, "aic"); if (IS_ERR(i2s->clk_aic)) return PTR_ERR(i2s->clk_aic); - i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s"); + i2s->clk_i2s = devm_clk_get(dev, "i2s"); if (IS_ERR(i2s->clk_i2s)) return PTR_ERR(i2s->clk_i2s); platform_set_drvdata(pdev, i2s); if (i2s->version == JZ_I2S_JZ4780) - ret = devm_snd_soc_register_component(&pdev->dev, + ret = devm_snd_soc_register_component(dev, &jz4740_i2s_component, &jz4780_i2s_dai, 1); else - ret = devm_snd_soc_register_component(&pdev->dev, + ret = devm_snd_soc_register_component(dev, &jz4740_i2s_component, &jz4740_i2s_dai, 1); if (ret) return ret; - return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + return devm_snd_dmaengine_pcm_register(dev, NULL, SND_DMAENGINE_PCM_FLAG_COMPAT); } From 62f9ed5f8768d2425461737d77b83f888b525c06 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 6 Mar 2020 23:29:29 +0100 Subject: [PATCH 233/415] ASoC: jz4740-i2s: Avoid passing enum as match data Instead of passing an enum as match data, and checking its value in the probe to register one or the other dai, pass a pointer to a struct i2s_soc_info, which contains all the information relative to one SoC. Signed-off-by: Paul Cercueil Link: https://lore.kernel.org/r/20200306222931.39664-4-paul@crapouillou.net Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 36 ++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 2572aba843e3b2..d1512d483cda67 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -93,6 +93,11 @@ enum jz47xx_i2s_version { JZ_I2S_JZ4780, }; +struct i2s_soc_info { + enum jz47xx_i2s_version version; + struct snd_soc_dai_driver *dai; +}; + struct jz4740_i2s { struct resource *mem; void __iomem *base; @@ -104,7 +109,7 @@ struct jz4740_i2s { struct snd_dmaengine_dai_dma_data playback_dma_data; struct snd_dmaengine_dai_dma_data capture_dma_data; - enum jz47xx_i2s_version version; + const struct i2s_soc_info *soc_info; }; static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s, @@ -284,7 +289,7 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; - if (i2s->version >= JZ_I2S_JZ4780) { + if (i2s->soc_info->version >= JZ_I2S_JZ4780) { div_reg &= ~I2SDIV_IDV_MASK; div_reg |= (div - 1) << I2SDIV_IDV_SHIFT; } else { @@ -398,7 +403,7 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, &i2s->capture_dma_data); - if (i2s->version >= JZ_I2S_JZ4780) { + if (i2s->soc_info->version >= JZ_I2S_JZ4780) { conf = (7 << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | (8 << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | JZ_AIC_CONF_OVERFLOW_PLAY_LAST | @@ -457,6 +462,11 @@ static struct snd_soc_dai_driver jz4740_i2s_dai = { .ops = &jz4740_i2s_dai_ops, }; +static const struct i2s_soc_info jz4740_i2s_soc_info = { + .version = JZ_I2S_JZ4740, + .dai = &jz4740_i2s_dai, +}; + static struct snd_soc_dai_driver jz4780_i2s_dai = { .probe = jz4740_i2s_dai_probe, .remove = jz4740_i2s_dai_remove, @@ -475,6 +485,11 @@ static struct snd_soc_dai_driver jz4780_i2s_dai = { .ops = &jz4740_i2s_dai_ops, }; +static const struct i2s_soc_info jz4780_i2s_soc_info = { + .version = JZ_I2S_JZ4780, + .dai = &jz4780_i2s_dai, +}; + static const struct snd_soc_component_driver jz4740_i2s_component = { .name = "jz4740-i2s", .suspend = jz4740_i2s_suspend, @@ -483,8 +498,8 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { #ifdef CONFIG_OF static const struct of_device_id jz4740_of_matches[] = { - { .compatible = "ingenic,jz4740-i2s", .data = (void *)JZ_I2S_JZ4740 }, - { .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 }, + { .compatible = "ingenic,jz4740-i2s", .data = &jz4740_i2s_soc_info }, + { .compatible = "ingenic,jz4780-i2s", .data = &jz4780_i2s_soc_info }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, jz4740_of_matches); @@ -501,7 +516,7 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) if (!i2s) return -ENOMEM; - i2s->version = (enum jz47xx_i2s_version)of_device_get_match_data(dev); + i2s->soc_info = device_get_match_data(dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); i2s->base = devm_ioremap_resource(dev, mem); @@ -520,13 +535,8 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); - if (i2s->version == JZ_I2S_JZ4780) - ret = devm_snd_soc_register_component(dev, - &jz4740_i2s_component, &jz4780_i2s_dai, 1); - else - ret = devm_snd_soc_register_component(dev, - &jz4740_i2s_component, &jz4740_i2s_dai, 1); - + ret = devm_snd_soc_register_component(dev, &jz4740_i2s_component, + i2s->soc_info->dai, 1); if (ret) return ret; From 3bbf9e2f8624432d1f436bf13ddce1cf9db5238d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Mar 2020 19:58:55 +0100 Subject: [PATCH 234/415] ALSA: pcm: oss: Simplify plugin frame size calculations Both snd_pcm_plug_client_size() and snd_pcm_plug_slave_size() do the almost same calculations of calling src_frames() and dst_frames() in the chain, but just to the different directions with each other. This patch simplifies those functions. Now they return -EINVAL for the invalid direction, but practically seen, there is no functional changes at all. Link: https://lore.kernel.org/r/20200309185855.15693-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 120 +++++++++++++++++------------------- 1 file changed, 56 insertions(+), 64 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index c9401832967c3c..58642fecd15c7c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -196,82 +196,74 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) return 0; } -snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, + snd_pcm_sframes_t frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - int stream; + struct snd_pcm_plugin *plugin, *plugin_next; - if (snd_BUG_ON(!plug)) - return -ENXIO; - if (drv_frames == 0) - return 0; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_last(plug); - while (plugin && drv_frames > 0) { - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) - drv_frames = plugin->src_frames(plugin, drv_frames); - plugin = plugin_prev; + plugin = snd_pcm_plug_first(plug); + while (plugin && frames > 0) { + plugin_next = plugin->next; + if (plugin->dst_frames) { + frames = plugin->dst_frames(plugin, frames); + if (frames < 0) + return frames; } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_first(plug); - while (plugin && drv_frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) - drv_frames = plugin->dst_frames(plugin, drv_frames); - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin = plugin_next; + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin = plugin_next; + } + return frames; +} + +static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, + snd_pcm_sframes_t frames) +{ + struct snd_pcm_plugin *plugin, *plugin_prev; + + plugin = snd_pcm_plug_last(plug); + while (plugin && frames > 0) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin_prev = plugin->prev; + if (plugin->src_frames) { + frames = plugin->src_frames(plugin, frames); + if (frames < 0) + return frames; } - } else + plugin = plugin_prev; + } + return frames; +} + +snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +{ + if (snd_BUG_ON(!plug)) + return -ENXIO; + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_src_frames(plug, drv_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_dst_frames(plug, drv_frames); + default: snd_BUG(); - return drv_frames; + return -EINVAL; + } } snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - snd_pcm_sframes_t frames; - int stream; - if (snd_BUG_ON(!plug)) return -ENXIO; - if (clt_frames == 0) - return 0; - frames = clt_frames; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_first(plug); - while (plugin && frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) { - frames = plugin->dst_frames(plugin, frames); - if (frames < 0) - return frames; - } - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin = plugin_next; - } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_last(plug); - while (plugin) { - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) { - frames = plugin->src_frames(plugin, frames); - if (frames < 0) - return frames; - } - plugin = plugin_prev; - } - } else + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_dst_frames(plug, clt_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_src_frames(plug, clt_frames); + default: snd_BUG(); - return frames; + return -EINVAL; + } } static int snd_pcm_plug_formats(const struct snd_mask *mask, From a22ae72b86a4f754e8d25fbf9ea5a8f77365e531 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 9 Mar 2020 14:27:43 -0500 Subject: [PATCH 235/415] ASoC: soc-core: disable route checks for legacy devices v5.4 changes in soc-core tightened the checks on soc_dapm_add_routes, which results in the ASoC card probe failing. Introduce a flag to be set in machine drivers to prevent the probe from stopping in case of incomplete topologies or missing routes. This flag is for backwards compatibility only and shall not be used for newer machine drivers. Example with an HDaudio card with a bad topology: [ 236.177898] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: Failed to add route iDisp1_out -> direct -> iDisp1 Tx [ 236.177902] skl_hda_dsp_generic skl_hda_dsp_generic: snd_soc_bind_card: snd_soc_dapm_add_routes failed: -19 with the disable_route_checks set: [ 64.031657] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: Failed to add route iDisp1_out -> direct -> iDisp1 Tx [ 64.031661] skl_hda_dsp_generic skl_hda_dsp_generic: snd_soc_bind_card: disable_route_checks set, ignoring errors on add_routes Fixes: daa480bde6b3a9 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") Signed-off-by: Pierre-Louis Bossart Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20200309192744.18380-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 28 ++++++++++++++++++++++++---- 2 files changed, 25 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8a2266676b2dd2..efb8bad7b0fabb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1058,6 +1058,7 @@ struct snd_soc_card { const struct snd_soc_dapm_route *of_dapm_routes; int num_of_dapm_routes; bool fully_routed; + bool disable_route_checks; /* lists of probed devices belonging to this card */ struct list_head component_dev_list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068d809c349a2f..b17366bac84632 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1256,8 +1256,18 @@ static int soc_probe_component(struct snd_soc_card *card, ret = snd_soc_dapm_add_routes(dapm, component->driver->dapm_routes, component->driver->num_dapm_routes); - if (ret < 0) - goto err_probe; + if (ret < 0) { + if (card->disable_route_checks) { + dev_info(card->dev, + "%s: disable_route_checks set, ignoring errors on add_routes\n", + __func__); + } else { + dev_err(card->dev, + "%s: snd_soc_dapm_add_routes failed: %d\n", + __func__, ret); + goto err_probe; + } + } /* see for_each_card_components */ list_add(&component->card_list, &card->component_dev_list); @@ -1938,8 +1948,18 @@ static int snd_soc_bind_card(struct snd_soc_card *card) ret = snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); - if (ret < 0) - goto probe_end; + if (ret < 0) { + if (card->disable_route_checks) { + dev_info(card->dev, + "%s: disable_route_checks set, ignoring errors on add_routes\n", + __func__); + } else { + dev_err(card->dev, + "%s: snd_soc_dapm_add_routes failed: %d\n", + __func__, ret); + goto probe_end; + } + } ret = snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, card->num_of_dapm_routes); From c8061689ffadde941f9c3756f1362bd2b97311c8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 9 Mar 2020 14:27:44 -0500 Subject: [PATCH 236/415] ASoC: Intel: skl_nau88l25_ssm4567: disable route checks Deal with incomplete topologies, this patch restores sound on user devices. Fixes: daa480bde6b3a9 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") Signed-off-by: Pierre-Louis Bossart Tested-by: ojab // Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20200309192744.18380-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index c99c8b23e509b0..b3b835156d776a 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -686,6 +686,7 @@ static struct snd_soc_card skylake_audio_card = { .codec_conf = ssm4567_codec_conf, .num_configs = ARRAY_SIZE(ssm4567_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = skylake_card_late_probe, }; From 995cbc3ca1ab39fb5cf254181dcfba883c5d6d69 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:07:29 +0900 Subject: [PATCH 237/415] ASoC: soc.h: add for_each_rtd_codecs/cpus_dai() macro We are using plural form for for_each_xxx() macro. But, for_each_rtd_codec/cpu_dai() are out of this rule. This patch adds plural form macro. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/8736aii326.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9543d9246ca450..09bc45b8bf0062 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1177,6 +1177,20 @@ struct snd_soc_pcm_runtime { #define for_each_rtd_cpu_dai_rollback(rtd, i, dai) \ for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);) +#define for_each_rtd_cpu_dais(rtd, i, dai) \ + for ((i) = 0; \ + ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]); \ + (i)++) +#define for_each_rtd_cpu_dais_rollback(rtd, i, dai) \ + for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);) +#define for_each_rtd_codec_dais(rtd, i, dai) \ + for ((i) = 0; \ + ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ + (i)++) +#define for_each_rtd_codec_dais_rollback(rtd, i, dai) \ + for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) + + void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); /* mixer control */ From 5dd1677c81c09932afad4dba2759dff7cf33ecbe Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:07:35 +0900 Subject: [PATCH 238/415] ASoC: Intel: use for_each_rtd_codecs/cpus_dai() macro This patch switch to use plural form macro. - for_each_rtd_codec_dai() + for_each_rtd_codec_dais() - for_each_rtd_codec_dai_rollback() + for_each_rtd_codec_dais_rollback() - for_each_rtd_cpu_dai() + for_each_rtd_cpu_dais() - for_each_rtd_cpu_dai_rollback() + for_each_rtd_cpu_dais_rollback() Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/871rq2i320.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index 02aa18d24319cf..2a6e5b12409923 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -164,7 +164,7 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params); - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* 100 Fs to drive 24 bit data */ ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 88f69e3697d21a..0ceb1748a26225 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -179,7 +179,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int ret, j; - for_each_rtd_codec_dai(runtime, j, codec_dai) { + for_each_rtd_codec_dais(runtime, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); @@ -224,7 +224,7 @@ static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int j, ret; - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { const char *name = codec_dai->component->name; struct snd_soc_component *component = codec_dai->component; struct snd_soc_dapm_context *dapm = diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index d8f2ff7139a9de..f65feee1c16603 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -472,7 +472,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int ret = 0, j; - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { /* * Use channel 4 and 5 for the first amp diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 96c814f364581b..341bb47311a682 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -399,7 +399,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int ret = 0, j; - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); if (ret < 0) { From c8654520234192688eefd7b40a66de7cf69c5189 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:07:42 +0900 Subject: [PATCH 239/415] ASoC: mediatek: use for_each_rtd_codecs/cpus_dai() macro This patch switch to use plural form macro. - for_each_rtd_codec_dai() + for_each_rtd_codec_dais() - for_each_rtd_codec_dai_rollback() + for_each_rtd_codec_dais_rollback() - for_each_rtd_cpu_dai() + for_each_rtd_cpu_dais() - for_each_rtd_cpu_dai_rollback() + for_each_rtd_cpu_dais_rollback() Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87zhcqgohd.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 4 ++-- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 2e1e61d8f12779..5d82159f4f2ea0 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -47,7 +47,7 @@ static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* pll from mclk 12.288M */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, params_rate(params) * 512); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index ebcc0b86286b9d..f65e3ebe38b8a4 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -51,7 +51,7 @@ static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* pll from mclk 12.288M */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, params_rate(params) * 512); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 849b050a54d149..bbc4ad749892c9 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -78,7 +78,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, break; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* pll from mclk */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock, params_rate(params) * 512); diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 4a5ef07e956b55..c4e4f1f99ddeef 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -52,7 +52,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) dev_err(rtd->dev, "failed to set cpu dai sysclk\n"); - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_sysclk(codec_dai, @@ -85,7 +85,7 @@ static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai; int ret = 0, j; - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_pll(codec_dai, From b5c52f5801c6e076377f7f411fb61bab86cb9542 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:07:48 +0900 Subject: [PATCH 240/415] ASoC: meson: use for_each_rtd_codecs/cpus_dai() macro This patch switch to use plural form macro. - for_each_rtd_codec_dai() + for_each_rtd_codec_dais() - for_each_rtd_codec_dai_rollback() + for_each_rtd_codec_dais_rollback() - for_each_rtd_cpu_dai() + for_each_rtd_cpu_dais() - for_each_rtd_cpu_dai_rollback() + for_each_rtd_cpu_dais_rollback() Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87y2sagoh7.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 48651631bdcfa3..77a7d5f36ebf61 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -60,7 +60,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai; int ret, i; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, be->codec_masks[i].tx, be->codec_masks[i].rx, diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index a70d244ef88b6b..b5d3c9f56bac0b 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -23,7 +23,7 @@ int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream, mclk = params_rate(params) * mclk_fs; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) From c998ee30e493ea3de0e52f0ec57995905d5ba43d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:07:57 +0900 Subject: [PATCH 241/415] ASoC: qcom: use for_each_rtd_codecs/cpus_dai() macro This patch switch to use plural form macro. - for_each_rtd_codec_dai() + for_each_rtd_codec_dais() - for_each_rtd_codec_dai_rollback() + for_each_rtd_codec_dais_rollback() - for_each_rtd_cpu_dai() + for_each_rtd_cpu_dais() - for_each_rtd_cpu_dai_rollback() + for_each_rtd_cpu_dais_rollback() Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87wo7ugogy.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- drivers/soundwire/qcom.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/sdm845.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb30bbec999ad8..440effed6df6dd 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -603,7 +603,7 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, ctrl->sruntime[dai->id] = sruntime; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_sdw_stream(codec_dai, sruntime, substream->stream); if (ret < 0 && ret != -ENOTSUPP) { diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 2d064f3bc9b6d0..7647af3e51f6b7 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -90,7 +90,7 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { component = codec_dai->component; /* Set default mclk for internal codec */ diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 5a23597261acfc..3ac02204a7069d 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -49,7 +49,7 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream, u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0, i; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); @@ -126,7 +126,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, } } - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { ret = snd_soc_dai_set_tdm_slot( @@ -265,7 +265,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) } break; case SLIMBUS_0_RX...SLIMBUS_6_TX: - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { rval = snd_soc_dai_set_channel_map(codec_dai, ARRAY_SIZE(tx_ch), tx_ch, @@ -344,7 +344,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B; - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { From a4be4187b2bfc66f4be8a6d35f497eb53a2c8d76 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:08:04 +0900 Subject: [PATCH 242/415] ASoC: soc: use for_each_rtd_codecs/cpus_dai() macro This patch switch to use plural form macro. - for_each_rtd_codec_dai() + for_each_rtd_codec_dais() - for_each_rtd_codec_dai_rollback() + for_each_rtd_codec_dais_rollback() - for_each_rtd_cpu_dai() + for_each_rtd_cpu_dais() - for_each_rtd_cpu_dai_rollback() + for_each_rtd_cpu_dais_rollback() Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87v9negogr.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 16 +++--- sound/soc/soc-dapm.c | 10 ++-- sound/soc/soc-pcm.c | 124 +++++++++++++++++++++---------------------- 3 files changed, 75 insertions(+), 75 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f2cfbf182f496c..4e0f55555e374e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -547,7 +547,7 @@ int snd_soc_suspend(struct device *dev) if (rtd->dai_link->ignore_suspend) continue; - for_each_rtd_codec_dai(rtd, i, dai) { + for_each_rtd_codec_dais(rtd, i, dai) { if (dai->stream_active[playback]) snd_soc_dai_digital_mute(dai, 1, playback); } @@ -689,7 +689,7 @@ static void soc_resume_deferred(struct work_struct *work) if (rtd->dai_link->ignore_suspend) continue; - for_each_rtd_codec_dai(rtd, i, dai) { + for_each_rtd_codec_dais(rtd, i, dai) { if (dai->stream_active[playback]) snd_soc_dai_digital_mute(dai, 0, playback); } @@ -1321,10 +1321,10 @@ static void soc_remove_link_dais(struct snd_soc_card *card) for_each_comp_order(order) { for_each_card_rtds(card, rtd) { /* remove the CODEC DAI */ - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) soc_remove_dai(codec_dai, order); - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) soc_remove_dai(cpu_dai, order); } } @@ -1344,14 +1344,14 @@ static int soc_probe_link_dais(struct snd_soc_card *card) card->name, rtd->num, order); /* probe the CPU DAI */ - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { ret = soc_probe_dai(cpu_dai, order); if (ret) return ret; } /* probe the CODEC DAI */ - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = soc_probe_dai(codec_dai, order); if (ret) return ret; @@ -1486,7 +1486,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int i; int ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, @@ -1514,7 +1514,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; break; } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { unsigned int fmt = dai_fmt; if (cpu_dai->component->driver->non_legacy_dai_naming) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 09fa437fc33edf..7374829c667597 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2437,7 +2437,7 @@ static ssize_t dapm_widget_show(struct device *dev, mutex_lock(&rtd->card->dapm_mutex); - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { struct snd_soc_component *cmpnt = codec_dai->component; count += dapm_widget_show_component(cmpnt, buf + count); @@ -4362,11 +4362,11 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, int i; if (rtd->num_cpus == 1) { - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) dapm_add_valid_dai_widget(card, rtd, codec_dai, rtd->cpu_dais[0]); } else if (rtd->num_codecs == rtd->num_cpus) { - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) dapm_add_valid_dai_widget(card, rtd, codec_dai, rtd->cpu_dais[i]); } else { @@ -4437,9 +4437,9 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, struct snd_soc_dai *cpu_dai; int i; - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) soc_dapm_dai_stream_event(cpu_dai, stream, event); - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) soc_dapm_dai_stream_event(codec_dai, stream, event); dapm_power_widgets(rtd->card, event); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7915adaaf0ac5..fbea005043deae 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -265,17 +265,17 @@ static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, lockdep_assert_held(&rtd->card->pcm_mutex); - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) cpu_dai->stream_active[stream] += action; - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) codec_dai->stream_active[stream] += action; - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { cpu_dai->active += action; cpu_dai->component->active += action; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { codec_dai->active += action; codec_dai->component->active += action; } @@ -455,14 +455,14 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, /* reject unmatched parameters when applying symmetry */ symmetry = rtd->dai_link->symmetric_rates; - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) symmetry |= cpu_dai->driver->symmetric_rates; - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) symmetry |= codec_dai->driver->symmetric_rates; if (symmetry) { - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { if (cpu_dai->rate && cpu_dai->rate != rate) { dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", cpu_dai->rate, rate); @@ -473,14 +473,14 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = rtd->dai_link->symmetric_channels; - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) symmetry |= cpu_dai->driver->symmetric_channels; - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) symmetry |= codec_dai->driver->symmetric_channels; if (symmetry) { - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { if (cpu_dai->channels && cpu_dai->channels != channels) { dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", @@ -492,14 +492,14 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = rtd->dai_link->symmetric_samplebits; - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) symmetry |= cpu_dai->driver->symmetric_samplebits; - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) symmetry |= codec_dai->driver->symmetric_samplebits; if (symmetry) { - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { if (cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) { dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", @@ -524,13 +524,13 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) link->symmetric_channels || link->symmetric_samplebits; - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) symmetry = symmetry || cpu_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_channels || cpu_dai->driver->symmetric_samplebits; - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) symmetry = symmetry || codec_dai->driver->symmetric_rates || codec_dai->driver->symmetric_channels || @@ -563,7 +563,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) int i; unsigned int bits = 0, cpu_bits = 0; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { pcm_codec = snd_soc_dai_get_pcm_stream(codec_dai, stream); if (pcm_codec->sig_bits == 0) { @@ -573,7 +573,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) bits = max(pcm_codec->sig_bits, bits); } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { pcm_cpu = snd_soc_dai_get_pcm_stream(cpu_dai, stream); if (pcm_cpu->sig_bits == 0) { @@ -612,7 +612,7 @@ int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, int i; /* first calculate min/max only for CPUs in the DAI link */ - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { /* * Skip CPUs which don't support the current stream type. @@ -635,7 +635,7 @@ int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, } /* second calculate min/max only for CODECs in the DAI link */ - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* * Skip CODECs which don't support the current stream type. @@ -780,10 +780,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) snd_soc_runtime_deactivate(rtd, substream->stream); - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) snd_soc_dai_shutdown(cpu_dai, substream); - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) snd_soc_dai_shutdown(codec_dai, substream); soc_rtd_shutdown(rtd, substream); @@ -842,7 +842,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } /* startup the audio subsystem */ - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { ret = snd_soc_dai_startup(cpu_dai, substream); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", @@ -851,7 +851,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_startup(codec_dai, substream); if (ret < 0) { dev_err(codec_dai->dev, @@ -903,7 +903,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) soc_pcm_apply_msb(substream); /* Symmetry only applies if we've already got an active stream. */ - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { if (cpu_dai->active) { ret = soc_pcm_apply_symmetry(substream, cpu_dai); if (ret != 0) @@ -911,7 +911,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { if (codec_dai->active) { ret = soc_pcm_apply_symmetry(substream, codec_dai); if (ret != 0) @@ -935,10 +935,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) snd_soc_dai_shutdown(codec_dai, substream); cpu_dai_err: - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) snd_soc_dai_shutdown(cpu_dai, substream); soc_rtd_shutdown(rtd, substream); @@ -1000,7 +1000,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_prepare(codec_dai, substream); if (ret < 0) { dev_err(codec_dai->dev, @@ -1010,7 +1010,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { ret = snd_soc_dai_prepare(cpu_dai, substream); if (ret < 0) { dev_err(cpu_dai->dev, @@ -1029,10 +1029,10 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_codec_dais(rtd, i, codec_dai) snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); - for_each_rtd_cpu_dai(rtd, i, cpu_dai) + for_each_rtd_cpu_dais(rtd, i, cpu_dai) snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); out: @@ -1097,7 +1097,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, goto out; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { struct snd_pcm_hw_params codec_params; /* @@ -1144,7 +1144,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { /* * Skip CPUs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details @@ -1186,7 +1186,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, i = rtd->num_cpus; interface_err: - for_each_rtd_cpu_dai_rollback(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais_rollback(rtd, i, cpu_dai) { if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) continue; @@ -1197,7 +1197,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, i = rtd->num_codecs; codec_err: - for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { + for_each_rtd_codec_dais_rollback(rtd, i, codec_dai) { if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue; @@ -1224,7 +1224,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* clear the corresponding DAIs parameters when going to be inactive */ - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { if (cpu_dai->active == 1) { cpu_dai->rate = 0; cpu_dai->channels = 0; @@ -1232,7 +1232,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { if (codec_dai->active == 1) { codec_dai->rate = 0; codec_dai->channels = 0; @@ -1241,7 +1241,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { int active = codec_dai->stream_active[substream->stream]; if (active == 1) @@ -1249,7 +1249,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) substream->stream); } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { int active = cpu_dai->stream_active[substream->stream]; if (active == 1) @@ -1264,14 +1264,14 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue; snd_soc_dai_hw_free(codec_dai, substream); } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) continue; @@ -1300,13 +1300,13 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) return ret; } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_trigger(codec_dai, substream, cmd); if (ret < 0) return ret; @@ -1323,13 +1323,13 @@ static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_trigger(codec_dai, substream, cmd); if (ret < 0) return ret; } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; @@ -1378,13 +1378,13 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd); if (ret < 0) return ret; } - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; @@ -1417,13 +1417,13 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) /* base delay if assigned in pointer callback */ delay = runtime->delay; - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { cpu_delay = max(cpu_delay, snd_soc_dai_delay(cpu_dai, substream)); } delay += cpu_delay; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { codec_delay = max(codec_delay, snd_soc_dai_delay(codec_dai, substream)); } @@ -1544,7 +1544,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; - for_each_rtd_cpu_dai(be, i, dai) { + for_each_rtd_cpu_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); dev_dbg(card->dev, "ASoC: try BE : %s\n", @@ -1554,7 +1554,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, return be; } - for_each_rtd_codec_dai(be, i, dai) { + for_each_rtd_codec_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); if (w == widget) @@ -1642,7 +1642,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, /* is there a valid CPU DAI widget for this BE */ do_prune = 1; - for_each_rtd_cpu_dai(dpcm->be, i, dai) { + for_each_rtd_cpu_dais(dpcm->be, i, dai) { widget = snd_soc_dai_get_widget(dai, stream); /* @@ -1657,7 +1657,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, /* is there a valid CODEC DAI widget for this BE */ do_prune = 1; - for_each_rtd_codec_dai(dpcm->be, i, dai) { + for_each_rtd_codec_dais(dpcm->be, i, dai) { widget = snd_soc_dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ @@ -1910,7 +1910,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, struct snd_soc_pcm_stream *codec_stream; int i; - for_each_rtd_codec_dai(be, i, dai) { + for_each_rtd_codec_dais(be, i, dai) { /* * Skip CODECs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details @@ -1948,7 +1948,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, struct snd_soc_dai *dai; int i; - for_each_rtd_cpu_dai(be, i, dai) { + for_each_rtd_cpu_dais(be, i, dai) { /* * Skip CPUs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details @@ -2003,7 +2003,7 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, struct snd_soc_dai *dai; int i; - for_each_rtd_cpu_dai(be, i, dai) { + for_each_rtd_cpu_dais(be, i, dai) { /* * Skip CPUs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details @@ -2020,7 +2020,7 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, cpu_stream->rates); } - for_each_rtd_codec_dai(be, i, dai) { + for_each_rtd_codec_dais(be, i, dai) { /* * Skip CODECs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details @@ -2046,7 +2046,7 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai; int i; - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { /* * Skip CPUs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details @@ -2102,7 +2102,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, if (soc_pcm_has_symmetry(fe_substream)) fe_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; - for_each_rtd_cpu_dai (fe, i, fe_cpu_dai) { + for_each_rtd_cpu_dais (fe, i, fe_cpu_dai) { /* Symmetry only applies if we've got an active stream. */ if (fe_cpu_dai->active) { err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai); @@ -2133,7 +2133,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; /* Symmetry only applies if we've got an active stream. */ - for_each_rtd_cpu_dai(rtd, i, cpu_dai) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { if (cpu_dai->active) { err = soc_pcm_apply_symmetry(fe_substream, cpu_dai); @@ -2142,7 +2142,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, } } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { if (codec_dai->active) { err = soc_pcm_apply_symmetry(fe_substream, codec_dai); @@ -3075,7 +3075,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int cpu_playback = rtd->dai_link->params ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { if (rtd->num_cpus == 1) { cpu_dai = rtd->cpu_dais[0]; } else if (rtd->num_cpus == rtd->num_codecs) { From 17e6dab5013ddb36997011cf6daea7297dfc215e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:08:10 +0900 Subject: [PATCH 243/415] ASoC: soc.h: remove non plural form for_each_xxx macro Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87tv2ygogl.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 09bc45b8bf0062..5e1b4ef1543ced 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1163,20 +1163,6 @@ struct snd_soc_pcm_runtime { for ((i) = 0; \ ((i) < rtd->num_components) && ((component) = rtd->components[i]);\ (i)++) -#define for_each_rtd_codec_dai(rtd, i, dai)\ - for ((i) = 0; \ - ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ - (i)++) -#define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ - for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) - -#define for_each_rtd_cpu_dai(rtd, i, dai)\ - for ((i) = 0; \ - ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]); \ - (i)++) -#define for_each_rtd_cpu_dai_rollback(rtd, i, dai) \ - for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);) - #define for_each_rtd_cpu_dais(rtd, i, dai) \ for ((i) = 0; \ ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]); \ From df817f8e71e3a0256bd3d2d3a4e5399b409698f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:08:16 +0900 Subject: [PATCH 244/415] ASoC: soc-dapm: add for_each_card_dapms() macro To be more readable code, this patch adds new for_each_card_dapms() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87sgiigogf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-dapm.c | 18 +++++++++--------- 2 files changed, 12 insertions(+), 9 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5e1b4ef1543ced..3aee33c8249ed9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1117,6 +1117,9 @@ struct snd_soc_card { #define for_each_card_components(card, component) \ list_for_each_entry(component, &(card)->component_dev_list, card_list) +#define for_each_card_dapms(card, dapm) \ + list_for_each_entry(dapm, &card->dapm_list, list) + /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_pcm_runtime { struct device *dev; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7374829c667597..ac48303ea26dfa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1716,9 +1716,8 @@ static void dapm_seq_run(struct snd_soc_card *card, i, cur_subseq); } - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) soc_dapm_async_complete(d); - } } static void dapm_widget_update(struct snd_soc_card *card) @@ -1949,7 +1948,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_start(card); - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) { if (dapm_idle_bias_off(d)) d->target_bias_level = SND_SOC_BIAS_OFF; else @@ -2013,10 +2012,10 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) * they're not ground referenced. */ bias = SND_SOC_BIAS_OFF; - list_for_each_entry(d, &card->dapm_list, list) + for_each_card_dapms(card, d) if (d->target_bias_level > bias) bias = d->target_bias_level; - list_for_each_entry(d, &card->dapm_list, list) + for_each_card_dapms(card, d) if (!dapm_idle_bias_off(d)) d->target_bias_level = bias; @@ -2025,7 +2024,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Run card bias changes at first */ dapm_pre_sequence_async(&card->dapm, 0); /* Run other bias changes in parallel */ - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) { if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); @@ -2049,7 +2048,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_seq_run(card, &up_list, event, true); /* Run all the bias changes in parallel */ - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) { if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); @@ -2059,7 +2058,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_post_sequence_async(&card->dapm, 0); /* do we need to notify any clients that DAPM event is complete */ - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) { if (!d->component) continue; @@ -4776,6 +4775,7 @@ void snd_soc_dapm_init(struct snd_soc_dapm_context *dapm, } INIT_LIST_HEAD(&dapm->list); + /* see for_each_card_dapms */ list_add(&dapm->list, &card->dapm_list); } EXPORT_SYMBOL_GPL(snd_soc_dapm_init); @@ -4822,7 +4822,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { struct snd_soc_dapm_context *dapm; - list_for_each_entry(dapm, &card->dapm_list, list) { + for_each_card_dapms(card, dapm) { if (dapm != &card->dapm) { soc_dapm_shutdown_dapm(dapm); if (dapm->bias_level == SND_SOC_BIAS_STANDBY) From 14596692631eadbefba8419698cccfc23bfccd2b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Mar 2020 13:08:21 +0900 Subject: [PATCH 245/415] ASoC: soc-dapm: add for_each_card_widgets() macro To be more readable code, this patch adds new for_each_card_widgets() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87r1y2goga.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/soc-dapm.c | 25 +++++++++++++------------ sound/soc/soc-topology.c | 2 +- 3 files changed, 19 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3aee33c8249ed9..03054bf9cd3776 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1120,6 +1120,11 @@ struct snd_soc_card { #define for_each_card_dapms(card, dapm) \ list_for_each_entry(dapm, &card->dapm_list, list) +#define for_each_card_widgets(card, w)\ + list_for_each_entry(w, &card->widgets, list) +#define for_each_card_widgets_safe(card, w, _w) \ + list_for_each_entry_safe(w, _w, &card->widgets, list) + /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_pcm_runtime { struct device *dev; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ac48303ea26dfa..e00a465a7c32ce 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -302,7 +302,7 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); - list_for_each_entry(w, &card->widgets, list) { + for_each_card_widgets(card, w) { if (w->is_ep) { dapm_mark_dirty(w, "Rechecking endpoints"); if (w->is_ep & SND_SOC_DAPM_EP_SINK) @@ -589,7 +589,7 @@ static void dapm_reset(struct snd_soc_card *card) memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); - list_for_each_entry(w, &card->widgets, list) { + for_each_card_widgets(card, w) { w->new_power = w->power; w->power_checked = false; } @@ -833,7 +833,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, *kcontrol = NULL; - list_for_each_entry(w, &dapm->card->widgets, list) { + for_each_card_widgets(dapm->card, w) { if (w == kcontrolw || w->dapm != kcontrolw->dapm) continue; for (i = 0; i < w->num_kcontrols; i++) { @@ -1967,7 +1967,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_power_one_widget(w, &up_list, &down_list); } - list_for_each_entry(w, &card->widgets, list) { + for_each_card_widgets(card, w) { switch (w->id) { case snd_soc_dapm_pre: case snd_soc_dapm_post: @@ -2376,7 +2376,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, if (!cmpnt->card) return 0; - list_for_each_entry(w, &cmpnt->card->widgets, list) { + for_each_card_widgets(cmpnt->card, w) { if (w->dapm != dapm) continue; @@ -2496,7 +2496,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w, *next_w; - list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { + for_each_card_widgets_safe(dapm->card, w, next_w) { if (w->dapm != dapm) continue; snd_soc_dapm_free_widget(w); @@ -2511,7 +2511,7 @@ static struct snd_soc_dapm_widget *dapm_find_widget( struct snd_soc_dapm_widget *w; struct snd_soc_dapm_widget *fallback = NULL; - list_for_each_entry(w, &dapm->card->widgets, list) { + for_each_card_widgets(dapm->card, w) { if (!strcmp(w->name, pin)) { if (w->dapm == dapm) return w; @@ -2910,7 +2910,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, * find src and dest widgets over all widgets but favor a widget from * current DAPM context */ - list_for_each_entry(w, &dapm->card->widgets, list) { + for_each_card_widgets(dapm->card, w) { if (!wsink && !(strcmp(w->name, sink))) { wtsink = w; if (w->dapm == dapm) { @@ -3189,7 +3189,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); - list_for_each_entry(w, &card->widgets, list) + for_each_card_widgets(card, w) { if (w->new) continue; @@ -3703,6 +3703,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->dapm = dapm; INIT_LIST_HEAD(&w->list); INIT_LIST_HEAD(&w->dirty); + /* see for_each_card_widgets */ list_add_tail(&w->list, &dapm->card->widgets); snd_soc_dapm_for_each_direction(dir) { @@ -4227,7 +4228,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) struct snd_soc_dai *dai; /* For each DAI widget... */ - list_for_each_entry(dai_w, &card->widgets, list) { + for_each_card_widgets(card, dai_w) { switch (dai_w->id) { case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: @@ -4246,7 +4247,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) dai = dai_w->priv; /* ...find all widgets with the same stream and link them */ - list_for_each_entry(w, &card->widgets, list) { + for_each_card_widgets(card, w) { if (w->dapm != dai_w->dapm) continue; @@ -4789,7 +4790,7 @@ static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm) mutex_lock(&card->dapm_mutex); - list_for_each_entry(w, &dapm->card->widgets, list) { + for_each_card_widgets(dapm->card, w) { if (w->dapm != dapm) continue; if (w->power) { diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba80755..33909afd3bbc15 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2774,7 +2774,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w, *next_w; - list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { + for_each_card_widgets_safe(dapm->card, w, next_w) { /* make sure we are a widget with correct context */ if (w->dobj.type != SND_SOC_DOBJ_WIDGET || w->dapm != dapm) From a3b7343e3f8c4c74516df41827b6d81905e346a1 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 9 Mar 2020 15:21:24 +0100 Subject: [PATCH 246/415] ASoC: SOF: Fix probe point getter Firmware API changes which introduced 'num_elems' param in several probe structs such as sof_ipc_probe_dma_add_params also impacted getter for both, DMA and probe points. All struct handlers except for sof_ipc_probe_info_params have been updated. Align said handler too to calculate payload size correctly. Fixes: f3b433e4699f ("ASoC: SOF: Implement Probe IPC API") Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200309142124.29262-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/sof/probe.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/probe.c b/sound/soc/sof/probe.c index 2b2f3dcfc7e9fd..c38169fe00c5d6 100644 --- a/sound/soc/sof/probe.c +++ b/sound/soc/sof/probe.c @@ -95,13 +95,17 @@ static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd, if (!reply->num_elems) goto exit; - bytes = reply->num_elems * sizeof(reply->dma[0]); + if (cmd == SOF_IPC_PROBE_DMA_INFO) + bytes = sizeof(reply->dma[0]); + else + bytes = sizeof(reply->desc[0]); + bytes *= reply->num_elems; *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL); if (!*params) { ret = -ENOMEM; goto exit; } - *num_params = msg.num_elems; + *num_params = reply->num_elems; exit: kfree(reply); From 2ef81057d80456870b97890dd79c8f56a85b1242 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 5 Mar 2020 15:53:08 +0100 Subject: [PATCH 247/415] ASoC: Intel: Skylake: Remove superfluous chip initialization Skylake driver does the controller init operation twice: - first during probe (only to stop it just before scheduling probe_work) - and during said probe_work where the actual correct sequence is executed To properly complete boot sequence when iDisp codec is present, bus initialization has to be called only after _i915_init() finishes. With additional _reset_list preceding _i915_init(), iDisp codec never gets the chance to enumerate on the link. Remove the superfluous initialization to address the issue. Signed-off-by: Cezary Rojewski Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200305145314.32579-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f755ca2484cff7..d662315253560d 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -803,6 +803,9 @@ static void skl_probe_work(struct work_struct *work) return; } + skl_init_pci(skl); + skl_dum_set(bus); + err = skl_init_chip(bus, true); if (err < 0) { dev_err(bus->dev, "Init chip failed with err: %d\n", err); @@ -918,8 +921,6 @@ static int skl_first_init(struct hdac_bus *bus) return -ENXIO; } - snd_hdac_bus_reset_link(bus, true); - snd_hdac_bus_parse_capabilities(bus); /* check if PPCAP exists */ @@ -967,11 +968,7 @@ static int skl_first_init(struct hdac_bus *bus) if (err < 0) return err; - /* initialize chip */ - skl_init_pci(skl); - skl_dum_set(bus); - - return skl_init_chip(bus, true); + return 0; } static int skl_probe(struct pci_dev *pci, @@ -1064,8 +1061,6 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); - snd_hdac_bus_stop_chip(bus); - /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) { From a66f88394a78fec9a05fa6e517e9603e8eca8363 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 5 Mar 2020 15:53:09 +0100 Subject: [PATCH 248/415] ASoC: Intel: Skylake: Select hda configuration permissively With _reset_link removed from the probe sequence, codec_mask at the time skl_find_hda_machine() is invoked will always be 0, so hda machine will never be chosen. Rather than reorganizing boot flow, be permissive about invalid mask. codec_mask will be set to proper value during probe_work - before skl_codec_create() ever gets called. Signed-off-by: Cezary Rojewski Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200305145314.32579-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index d662315253560d..4827fe6bc1cb97 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -481,13 +481,8 @@ static struct skl_ssp_clk skl_ssp_clks[] = { static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl_dev *skl, struct snd_soc_acpi_mach *machines) { - struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach; - /* check if we have any codecs detected on bus */ - if (bus->codec_mask == 0) - return NULL; - /* point to common table */ mach = snd_soc_acpi_intel_hda_machines; From e603f11d5df8997d104ab405ff27640b90baffaa Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 5 Mar 2020 15:53:10 +0100 Subject: [PATCH 249/415] ASoC: Intel: Skylake: Enable codec wakeup during chip init Follow the recommendation set by hda_intel.c and enable HDMI/DP codec wakeup during bus initialization procedure. Disable wakeup once init completes. Signed-off-by: Cezary Rojewski Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200305145314.32579-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 4827fe6bc1cb97..e2e531c96dd109 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -130,6 +130,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) struct hdac_ext_link *hlink; int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); @@ -138,6 +139,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } From 9e6c382f5a6161eb55115fb56614b9827f2e7da3 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 5 Mar 2020 15:53:11 +0100 Subject: [PATCH 250/415] ASoC: Intel: Skylake: Shield against no-NHLT configurations Some configurations expose no NHLT table at all within their /sys/firmware/acpi/tables. To prevent NULL-dereference errors from occurring, adjust probe flow and append additional safety checks in functions involved in NHLT lifecycle. Signed-off-by: Cezary Rojewski Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200305145314.32579-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-nhlt.c | 3 ++- sound/soc/intel/skylake/skl.c | 9 +++++++-- 2 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 19f328d71f2446..d9c8f5cb389e33 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -182,7 +182,8 @@ void skl_nhlt_remove_sysfs(struct skl_dev *skl) { struct device *dev = &skl->pci->dev; - sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); + if (skl->nhlt) + sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); } /* diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index e2e531c96dd109..7ad8a75759bde7 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -633,6 +633,9 @@ static int skl_clock_device_register(struct skl_dev *skl) struct platform_device_info pdevinfo = {NULL}; struct skl_clk_pdata *clk_pdata; + if (!skl->nhlt) + return 0; + clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata), GFP_KERNEL); if (!clk_pdata) @@ -1074,7 +1077,8 @@ static int skl_probe(struct pci_dev *pci, out_clk_free: skl_clock_device_unregister(skl); out_nhlt_free: - intel_nhlt_free(skl->nhlt); + if (skl->nhlt) + intel_nhlt_free(skl->nhlt); out_free: skl_free(bus); @@ -1123,7 +1127,8 @@ static void skl_remove(struct pci_dev *pci) skl_dmic_device_unregister(skl); skl_clock_device_unregister(skl); skl_nhlt_remove_sysfs(skl); - intel_nhlt_free(skl->nhlt); + if (skl->nhlt) + intel_nhlt_free(skl->nhlt); skl_free(bus); dev_set_drvdata(&pci->dev, NULL); } From 024aa45f55ccd40704cfdef61b2a8b6d0de9cdd1 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 5 Mar 2020 15:53:13 +0100 Subject: [PATCH 251/415] ASoC: Intel: Allow for ROM init retry on CNL platforms Due to unconditional initial timeouts, firmware may fail to load during its initialization. This issue cannot be resolved on driver side as it is caused by external sources such as CSME but has to be accounted for nonetheless. Fixes: cb6a55284629 ("ASoC: Intel: cnl: Add sst library functions for cnl platform") Signed-off-by: Cezary Rojewski Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200305145314.32579-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/bxt-sst.c | 2 -- sound/soc/intel/skylake/cnl-sst.c | 15 ++++++++++----- sound/soc/intel/skylake/skl-sst-dsp.h | 1 + 3 files changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 92a82e6b5fe621..cdafade8abd672 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -38,8 +38,6 @@ /* Delay before scheduling D0i3 entry */ #define BXT_D0I3_DELAY 5000 -#define BXT_FW_ROM_INIT_RETRY 3 - static unsigned int bxt_get_errorcode(struct sst_dsp *ctx) { return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE); diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index 4f64f097e9ae36..060e47ae339149 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -109,7 +109,7 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) { struct firmware stripped_fw; struct skl_dev *cnl = ctx->thread_context; - int ret; + int ret, i; if (!ctx->fw) { ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev); @@ -131,12 +131,16 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) stripped_fw.size = ctx->fw->size; skl_dsp_strip_extended_manifest(&stripped_fw); - ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size); - if (ret < 0) { - dev_err(ctx->dev, "prepare firmware failed: %d\n", ret); - goto cnl_load_base_firmware_failed; + for (i = 0; i < BXT_FW_ROM_INIT_RETRY; i++) { + ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size); + if (!ret) + break; + dev_dbg(ctx->dev, "prepare firmware failed: %d\n", ret); } + if (ret < 0) + goto cnl_load_base_firmware_failed; + ret = sst_transfer_fw_host_dma(ctx); if (ret < 0) { dev_err(ctx->dev, "transfer firmware failed: %d\n", ret); @@ -158,6 +162,7 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) return 0; cnl_load_base_firmware_failed: + dev_err(ctx->dev, "firmware load failed: %d\n", ret); release_firmware(ctx->fw); ctx->fw = NULL; diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index cdfec0fca5773d..067d1ea11cde4f 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -67,6 +67,7 @@ struct skl_dev; #define SKL_FW_INIT 0x1 #define SKL_FW_RFW_START 0xf +#define BXT_FW_ROM_INIT_RETRY 3 #define SKL_ADSPIC_IPC 1 #define SKL_ADSPIS_IPC 1 From 7693cadac86548b30389a6e11d78c38db654f393 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 5 Mar 2020 15:53:14 +0100 Subject: [PATCH 252/415] ASoC: Intel: Skylake: Await purge request ack on CNL Each purge request is sent by driver after master core is powered up and unresetted but before it is unstalled. On unstall, ROM begins processing the request and initializing environment for FW load. Host should await ROM's ack before moving forward. Without doing so, ROM init poll may start too early and false timeouts can occur. Fixes: cb6a55284629 ("ASoC: Intel: cnl: Add sst library functions for cnl platform") Signed-off-by: Cezary Rojewski Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200305145314.32579-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/bxt-sst.c | 1 - sound/soc/intel/skylake/cnl-sst.c | 20 ++++++++++++++++++-- sound/soc/intel/skylake/skl-sst-dsp.h | 1 + 3 files changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index cdafade8abd672..38b9d749408359 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -17,7 +17,6 @@ #include "skl.h" #define BXT_BASEFW_TIMEOUT 3000 -#define BXT_INIT_TIMEOUT 300 #define BXT_ROM_INIT_TIMEOUT 70 #define BXT_IPC_PURGE_FW 0x01004000 diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index 060e47ae339149..c6abcd5aa67b96 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -57,18 +57,34 @@ static int cnl_prepare_fw(struct sst_dsp *ctx, const void *fwdata, u32 fwsize) ctx->dsp_ops.stream_tag = stream_tag; memcpy(ctx->dmab.area, fwdata, fwsize); + ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK); + if (ret < 0) { + dev_err(ctx->dev, "dsp core0 power up failed\n"); + ret = -EIO; + goto base_fw_load_failed; + } + /* purge FW request */ sst_dsp_shim_write(ctx, CNL_ADSP_REG_HIPCIDR, CNL_ADSP_REG_HIPCIDR_BUSY | (CNL_IPC_PURGE | ((stream_tag - 1) << CNL_ROM_CTRL_DMA_ID))); - ret = cnl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK); + ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK); if (ret < 0) { - dev_err(ctx->dev, "dsp boot core failed ret: %d\n", ret); + dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret); ret = -EIO; goto base_fw_load_failed; } + ret = sst_dsp_register_poll(ctx, CNL_ADSP_REG_HIPCIDA, + CNL_ADSP_REG_HIPCIDA_DONE, + CNL_ADSP_REG_HIPCIDA_DONE, + BXT_INIT_TIMEOUT, "HIPCIDA Done"); + if (ret < 0) { + dev_err(ctx->dev, "timeout for purge request: %d\n", ret); + goto base_fw_load_failed; + } + /* enable interrupt */ cnl_ipc_int_enable(ctx); cnl_ipc_op_int_enable(ctx); diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index 067d1ea11cde4f..1df9ef422f61df 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -68,6 +68,7 @@ struct skl_dev; #define SKL_FW_INIT 0x1 #define SKL_FW_RFW_START 0xf #define BXT_FW_ROM_INIT_RETRY 3 +#define BXT_INIT_TIMEOUT 300 #define SKL_ADSPIC_IPC 1 #define SKL_ADSPIS_IPC 1 From 5549ea64799784308cc03313a86dea3de56d48ce Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 10 Mar 2020 11:35:07 -0500 Subject: [PATCH 253/415] ASoC: rt5682: fix unmet dependencies The rt5682 code can be used in I2C or SoundWire mode. When I2C is not selected, we have the following issue: WARNING: unmet direct dependencies detected for SND_SOC_RT5682 Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && I2C [=n] Selected by [m]: - SND_SOC_RT5682_SDW [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SOUNDWIRE [=m] Fix by adding SOUNDWIRE as a dependency. Fixes: 03f6fc6de9192f ('ASoC: rt5682: Add the soundwire support') Reported-by: kbuild test robot Signed-off-by: Pierre-Louis Bossart Cc: Oder Chiou Link: https://lore.kernel.org/r/20200310163509.14466-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6aee70ed43df5b..78be69e9b61829 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1135,7 +1135,7 @@ config SND_SOC_RT5677_SPI config SND_SOC_RT5682 tristate - depends on I2C + depends on I2C || SOUNDWIRE config SND_SOC_RT5682_SDW tristate "Realtek RT5682 Codec - SDW" From 724cc62f7a71e3a04112126806c62d9c639ab92c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 10 Mar 2020 11:35:09 -0500 Subject: [PATCH 254/415] ASoC: rt5682-sdw: fix 'defined but not used' pm functions Gcc reports the following warnings: sound/soc/codecs/rt5682-sdw.c:286:12: warning: 'rt5682_dev_resume' defined but not used [-Wunused-function] static int rt5682_dev_resume(struct device *dev) ^~~~~~~~~~~~~~~~~ sound/soc/codecs/rt5682-sdw.c:273:12: warning: 'rt5682_dev_suspend' defined but not used [-Wunused-function] static int rt5682_dev_suspend(struct device *dev) ^~~~~~~~~~~~~~~~~~ Fix by adding maybe_unused as done for other SoundWire codecs Fixes: 03f6fc6de9192f ('ASoC: rt5682: Add the soundwire support') Reported-by: kbuild test robot Signed-off-by: Pierre-Louis Bossart Cc: Oder Chiou Link: https://lore.kernel.org/r/20200310163509.14466-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 1d6963dd6403aa..a2d1d3ae1e312f 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -270,7 +270,7 @@ static const struct sdw_device_id rt5682_id[] = { }; MODULE_DEVICE_TABLE(sdw, rt5682_id); -static int rt5682_dev_suspend(struct device *dev) +static int __maybe_unused rt5682_dev_suspend(struct device *dev) { struct rt5682_priv *rt5682 = dev_get_drvdata(dev); @@ -283,7 +283,7 @@ static int rt5682_dev_suspend(struct device *dev) return 0; } -static int rt5682_dev_resume(struct device *dev) +static int __maybe_unused rt5682_dev_resume(struct device *dev) { struct sdw_slave *slave = dev_to_sdw_dev(dev); struct rt5682_priv *rt5682 = dev_get_drvdata(dev); From d0c9abb8339dfdb5c5fcdfab5aefcba578a4d50d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2020 17:36:25 +0100 Subject: [PATCH 255/415] ASoC: pcm: Fix (again) possible buffer overflow in dpcm state sysfs output This is re-applying the fix that went into 5.6 (commit 6c89ffea60aa) as the changes were wiped out after merging the other code refactoring. Basically the same changes, just replacing the suspicious calls of snprintf() with scnprintf(). Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20200310163625.10838-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fbea005043deae..733d7e8a0e5512 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -66,16 +66,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, unsigned long flags; /* FE state */ - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "[%s - %s]\n", fe->dai_link->name, stream ? "Capture" : "Playback"); - offset += snprintf(buf + offset, size - offset, "State: %s\n", + offset += scnprintf(buf + offset, size - offset, "State: %s\n", dpcm_state_string(fe->dpcm[stream].state)); if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "Hardware Params: " "Format = %s, Channels = %d, Rate = %d\n", snd_pcm_format_name(params_format(params)), @@ -83,10 +83,10 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_rate(params)); /* BEs state */ - offset += snprintf(buf + offset, size - offset, "Backends:\n"); + offset += scnprintf(buf + offset, size - offset, "Backends:\n"); if (list_empty(&fe->dpcm[stream].be_clients)) { - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " No active DSP links\n"); goto out; } @@ -96,16 +96,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "- %s\n", be->dai_link->name); - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " State: %s\n", dpcm_state_string(be->dpcm[stream].state)); if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " Hardware Params: " "Format = %s, Channels = %d, Rate = %d\n", snd_pcm_format_name(params_format(params)), From 34aa7994ad15af31f2a7c1146d03b1f8d08af3cc Mon Sep 17 00:00:00 2001 From: Jules Irenge Date: Wed, 11 Mar 2020 01:09:07 +0000 Subject: [PATCH 256/415] ALSA: firewire-tascam: Add missing annotation for tscm_hwdep_read_queue() Sparse reports a warning at tscm_hwdep_read_queue() warning: context imbalance in tscm_hwdep_read_queue() - unexpected unlock The root cause is the missing annotation at tscm_hwdep_read_queue() Add the missing __releases(&tscm->lock) annotation Signed-off-by: Jules Irenge Acked-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20200311010908.42366-8-jbi.octave@gmail.com Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam-hwdep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index c29a97f6f63888..9801e33e7f2a30 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -36,6 +36,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf, long remained, loff_t *offset) + __releases(&tscm->lock) { char __user *pos = buf; unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL; From 3db1b00f2122bfe259bdb19f7c0bfcaec54568e3 Mon Sep 17 00:00:00 2001 From: Jules Irenge Date: Wed, 11 Mar 2020 01:09:08 +0000 Subject: [PATCH 257/415] ALSA: firewire-tascam: Add missing annotation for tscm_hwdep_read_locked() Sparse reports a warning at tscm_hwdep_read_locked() warning: context imbalance in tscm_hwdep_read_locked() - unexpected unlock The root cause is the missing annotation at tscm_hwdep_read_locked() Add the missing __releases(&tscm->lock) annotation Signed-off-by: Jules Irenge Acked-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20200311010908.42366-9-jbi.octave@gmail.com Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam-hwdep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index 9801e33e7f2a30..6f38335fe10bfd 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -17,6 +17,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, long count, loff_t *offset) + __releases(&tscm->lock) { struct snd_firewire_event_lock_status event = { .type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS, From 97249a89c17e8f1288fed1ebc617ea2e9e88d501 Mon Sep 17 00:00:00 2001 From: Benjamin Gaignard Date: Fri, 28 Feb 2020 16:27:06 +0100 Subject: [PATCH 258/415] ASoC: Convert cirrus,cs42l51 to json-schema Convert cirrus,cs42l51 to yaml format. Signed-off-by: Benjamin Gaignard Reviewed-by: Rob Herring Link: https://lore.kernel.org/r/20200228152706.29749-1-benjamin.gaignard@st.com Signed-off-by: Mark Brown --- .../bindings/sound/cirrus,cs42l51.yaml | 69 +++++++++++++++++++ .../devicetree/bindings/sound/cs42l51.txt | 33 --------- 2 files changed, 69 insertions(+), 33 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml new file mode 100644 index 00000000000000..efce847a340837 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42l51.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: CS42L51 audio codec DT bindings + +maintainers: + - Olivier Moysan + +properties: + compatible: + const: cirrus,cs42l51 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: MCLK + + reset-gpios: + maxItems: 1 + + VL-supply: + description: phandle to voltage regulator of digital interface section + + VD-supply: + description: phandle to voltage regulator of digital internal section + + VA-supply: + description: phandle to voltage regulator of analog internal section + + VAHP-supply: + description: phandle to voltage regulator of headphone + +required: + - compatible + - reg + - "#sound-dai-cells" + +examples: + - | + #include + i2c@0 { + #address-cells = <1>; + #size-cells = <0>; + + cs42l51@4a { + compatible = "cirrus,cs42l51"; + reg = <0x4a>; + #sound-dai-cells = <0>; + clocks = <&mclk_prov>; + clock-names = "MCLK"; + VL-supply = <®_audio>; + VD-supply = <®_audio>; + VA-supply = <®_audio>; + VAHP-supply = <®_audio>; + reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/cs42l51.txt b/Documentation/devicetree/bindings/sound/cs42l51.txt deleted file mode 100644 index acbd68ddd2cbc7..00000000000000 --- a/Documentation/devicetree/bindings/sound/cs42l51.txt +++ /dev/null @@ -1,33 +0,0 @@ -CS42L51 audio CODEC - -Required properties: - - - compatible : "cirrus,cs42l51" - - - reg : the I2C address of the device for I2C. - -Optional properties: - - VL-supply, VD-supply, VA-supply, VAHP-supply: power supplies for the device, - as covered in Documentation/devicetree/bindings/regulator/regulator.txt. - - - reset-gpios : GPIO specification for the reset pin. If specified, it will be - deasserted before starting the communication with the codec. - - - clocks : a list of phandles + clock-specifiers, one for each entry in - clock-names - - - clock-names : must contain "MCLK" - -Example: - -cs42l51: cs42l51@4a { - compatible = "cirrus,cs42l51"; - reg = <0x4a>; - clocks = <&mclk_prov>; - clock-names = "MCLK"; - VL-supply = <®_audio>; - VD-supply = <®_audio>; - VA-supply = <®_audio>; - VAHP-supply = <®_audio>; - reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; -}; From 103ae95513803102d2a2c91458cfac5dfbaad124 Mon Sep 17 00:00:00 2001 From: tangbin Date: Wed, 11 Mar 2020 22:46:46 +0800 Subject: [PATCH 259/415] ASoC: zte: zx-tdm: remove redundant variables dev In this function, the variable 'dev' is assigned to '&pdev->dev', but in the following code, all the assignments to 'struce device' are used '&pdev->dev' instead of 'dev',except 'zx_tdm->dev'. So,the variable 'dev' in this function is redundant and can be replaced by '&pdev->dev' as elsewhere. Signed-off-by: tangbin Link: https://lore.kernel.org/r/20200311144646.11292-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/zte/zx-tdm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index 0e5a05b25a7713..4f787185d6300e 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -371,7 +371,6 @@ static struct snd_soc_dai_driver zx_tdm_dai = { static int zx_tdm_probe(struct platform_device *pdev) { - struct device *dev = &pdev->dev; struct of_phandle_args out_args; unsigned int dma_reg_offset; struct zx_tdm_info *zx_tdm; @@ -384,7 +383,7 @@ static int zx_tdm_probe(struct platform_device *pdev) if (!zx_tdm) return -ENOMEM; - zx_tdm->dev = dev; + zx_tdm->dev = &pdev->dev; zx_tdm->dai_wclk = devm_clk_get(&pdev->dev, "wclk"); if (IS_ERR(zx_tdm->dai_wclk)) { From 9032cdd96a2d4b0ef2f43499328f8a68050be2ec Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Fri, 17 Jan 2020 18:03:52 +0100 Subject: [PATCH 260/415] ASoC: dt-bindings: stm32: convert spdfirx to json-schema Convert the STM32 SPDIFRX bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan Reviewed-by: Rob Herring Link: https://lore.kernel.org/r/20200117170352.16040-1-olivier.moysan@st.com Signed-off-by: Mark Brown --- .../bindings/sound/st,stm32-spdifrx.txt | 56 ------------- .../bindings/sound/st,stm32-spdifrx.yaml | 80 +++++++++++++++++++ 2 files changed, 80 insertions(+), 56 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt deleted file mode 100644 index 33826f2459fa80..00000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt +++ /dev/null @@ -1,56 +0,0 @@ -STMicroelectronics STM32 S/PDIF receiver (SPDIFRX). - -The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with -IEC-60958 and IEC-61937. - -Required properties: - - compatible: should be "st,stm32h7-spdifrx" - - reg: cpu DAI IP base address and size - - clocks: must contain an entry for kclk (used as S/PDIF signal reference) - - clock-names: must contain "kclk" - - interrupts: cpu DAI interrupt line - - dmas: DMA specifiers for audio data DMA and iec control flow DMA - See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: two dmas have to be defined, "rx" and "rx-ctrl" - -Optional properties: - - resets: Reference to a reset controller asserting the SPDIFRX - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -spdifrx: spdifrx@40004000 { - compatible = "st,stm32h7-spdifrx"; - reg = <0x40004000 0x400>; - clocks = <&rcc SPDIFRX_CK>; - clock-names = "kclk"; - interrupts = <97>; - dmas = <&dmamux1 2 93 0x400 0x0>, - <&dmamux1 3 94 0x400 0x0>; - dma-names = "rx", "rx-ctrl"; - pinctrl-0 = <&spdifrx_pins>; - pinctrl-names = "default"; - - spdifrx_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - }; - }; -}; - -spdif_in: spdif-in { - compatible = "linux,spdif-dir"; - - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; - -soundcard { - compatible = "audio-graph-card"; - dais = <&spdifrx_port>; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml new file mode 100644 index 00000000000000..b7f7dc45223192 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) + +maintainers: + - Olivier Moysan + +description: | + The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with + IEC-60958 and IEC-61937. + +properties: + compatible: + enum: + - st,stm32h7-spdifrx + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: kclk + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio data capture DMA + - description: IEC status bits capture DMA + + dma-names: + items: + - const: rx + - const: rx-ctrl + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include + #include + spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + #sound-dai-cells = <0>; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIF_K>; + clock-names = "kclk"; + interrupts = ; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + }; + +... From 9b60441692d94effcd37a141035c6106a91ddf8c Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 11 Mar 2020 18:04:21 +0000 Subject: [PATCH 261/415] ASoC: qdsp6: q6asm-dai: only enable dais from device tree Existing code enables all the playback and capture dais even if there is no device tree entry. This can lead to un-necessary dais in the system which will never be used. So honour whats specfied in device tree. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200311180422.28363-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 30 +++++++++++++++++++++++------- 1 file changed, 23 insertions(+), 7 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c0d422d0ab94fd..8b48815ff9185e 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -69,6 +69,8 @@ struct q6asm_dai_rtd { }; struct q6asm_dai_data { + struct snd_soc_dai_driver *dais; + int num_dais; long long int sid; }; @@ -889,7 +891,7 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = { .compr_ops = &q6asm_dai_compr_ops, }; -static struct snd_soc_dai_driver q6asm_fe_dais[] = { +static struct snd_soc_dai_driver q6asm_fe_dais_template[] = { Q6ASM_FEDAI_DRIVER(1), Q6ASM_FEDAI_DRIVER(2), Q6ASM_FEDAI_DRIVER(3), @@ -903,10 +905,22 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = { static int of_q6asm_parse_dai_data(struct device *dev, struct q6asm_dai_data *pdata) { - static struct snd_soc_dai_driver *dai_drv; + struct snd_soc_dai_driver *dai_drv; struct snd_soc_pcm_stream empty_stream; struct device_node *node; - int ret, id, dir; + int ret, id, dir, idx = 0; + + + pdata->num_dais = of_get_child_count(dev->of_node); + if (!pdata->num_dais) { + dev_err(dev, "No dais found in DT\n"); + return -EINVAL; + } + + pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv), + GFP_KERNEL); + if (!pdata->dais) + return -ENOMEM; memset(&empty_stream, 0, sizeof(empty_stream)); @@ -917,7 +931,8 @@ static int of_q6asm_parse_dai_data(struct device *dev, continue; } - dai_drv = &q6asm_fe_dais[id]; + dai_drv = &pdata->dais[idx++]; + *dai_drv = q6asm_fe_dais_template[id]; ret = of_property_read_u32(node, "direction", &dir); if (ret) @@ -955,11 +970,12 @@ static int q6asm_dai_probe(struct platform_device *pdev) dev_set_drvdata(dev, pdata); - of_q6asm_parse_dai_data(dev, pdata); + rc = of_q6asm_parse_dai_data(dev, pdata); + if (rc) + return rc; return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component, - q6asm_fe_dais, - ARRAY_SIZE(q6asm_fe_dais)); + pdata->dais, pdata->num_dais); } static const struct of_device_id q6asm_dai_device_id[] = { From f864edff110d3e6f8995636a9a604f6b98eaa308 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 11 Mar 2020 18:04:22 +0000 Subject: [PATCH 262/415] ASoC: qdsp6: q6routing: remove default routing Frontend dais can be configured to rx or tx or both, however having default routes without considering this configuration can lead to failures during card probe as below for compress rx only case. These routing have to come from sound card routing table in device tree. "routing: ASoC: Failed to add route MM_UL1 -> direct -> MultiMedia1 Capture msm-snd-sdm845 sound: ASoC: failed to instantiate card -19 " Reported-by: Vinod Koul Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200311180422.28363-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 20724102e85a0e..4d5915b9a06d52 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -918,25 +918,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL6", NULL, "MultiMedia6 Mixer"}, {"MM_UL7", NULL, "MultiMedia7 Mixer"}, {"MM_UL8", NULL, "MultiMedia8 Mixer"}, - - {"MM_DL1", NULL, "MultiMedia1 Playback" }, - {"MM_DL2", NULL, "MultiMedia2 Playback" }, - {"MM_DL3", NULL, "MultiMedia3 Playback" }, - {"MM_DL4", NULL, "MultiMedia4 Playback" }, - {"MM_DL5", NULL, "MultiMedia5 Playback" }, - {"MM_DL6", NULL, "MultiMedia6 Playback" }, - {"MM_DL7", NULL, "MultiMedia7 Playback" }, - {"MM_DL8", NULL, "MultiMedia8 Playback" }, - - {"MultiMedia1 Capture", NULL, "MM_UL1"}, - {"MultiMedia2 Capture", NULL, "MM_UL2"}, - {"MultiMedia3 Capture", NULL, "MM_UL3"}, - {"MultiMedia4 Capture", NULL, "MM_UL4"}, - {"MultiMedia5 Capture", NULL, "MM_UL5"}, - {"MultiMedia6 Capture", NULL, "MM_UL6"}, - {"MultiMedia7 Capture", NULL, "MM_UL7"}, - {"MultiMedia8 Capture", NULL, "MM_UL8"}, - }; static int routing_hw_params(struct snd_soc_component *component, From d95cf9324b1c334809029896651b067204c63d0b Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Wed, 11 Mar 2020 19:45:37 +0200 Subject: [PATCH 263/415] MAINTAINERS: add entry for Sound Open Firmware drivers Sound Open Firmware (SOF) is an open source audio DSP firwmare instrastructure and SDK. The kernel drivers for SOF are part of the ALSA subsystem. Signed-off-by: Kai Vehmanen Acked-by: Ranjani Sridharan Acked-by: Liam Girdwood Acked-by: Daniel Baluta Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200311174537.24497-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- MAINTAINERS | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 5e5382e2fe212c..3bb6895ba5b95b 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -15439,6 +15439,17 @@ F: sound/soc/ F: include/dt-bindings/sound/ F: include/sound/soc* +SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS +M: Pierre-Louis Bossart +M: Liam Girdwood +M: Ranjani Sridharan +M: Kai Vehmanen +M: Daniel Baluta +L: sound-open-firmware@alsa-project.org (moderated for non-subscribers) +W: https://github.com/thesofproject/linux/ +S: Supported +F: sound/soc/sof/ + SOUNDWIRE SUBSYSTEM M: Vinod Koul M: Sanyog Kale From a5107b1a0993ff035dc8a1cf331da7bf961e4cbb Mon Sep 17 00:00:00 2001 From: James Schulman Date: Tue, 10 Mar 2020 14:27:51 -0500 Subject: [PATCH 264/415] MAINTAINERS: Update Cirrus Logic codec driver maintainers Brian & Paul are no longer active audio codec driver maintainers. Update list to reflect myself and David Rhodes as the active maintainers. Signed-off-by: James Schulman Link: https://lore.kernel.org/r/20200310192751.24487-1-james.schulman@cirrus.com Signed-off-by: Mark Brown --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 3bb6895ba5b95b..c6fc68b66434f6 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3964,8 +3964,8 @@ F: Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt F: sound/soc/codecs/cros_ec_codec.* CIRRUS LOGIC AUDIO CODEC DRIVERS -M: Brian Austin -M: Paul Handrigan +M: James Schulman +M: David Rhodes L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/codecs/cs* From 16dcefc23eefda57ee705dcffba67f1bd42c01bc Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 12 Mar 2020 13:00:58 +0100 Subject: [PATCH 265/415] ASoC: SOF: Intel: Fix stream cleanup on hw free Field "substream" gets assigned during stream setup in hda_dsp_pcm_hw_params() but it is never cleared afterwards during cleanup procedure. Now, any non-pcm operation e.g.: compress can mistakenly make use of that pointer as it's bypassing all "if (s->substream)" checks. Nulling the pointer during hw_free operation ensures no wild pointers are left behind. Fixes: cdae3b9a47aa ("ASoC: SOF: Intel: Add Intel specific HDA PCM operations") Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312120058.15057-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index c0ab9bb2a7978c..d2234f802788aa 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -547,6 +547,8 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, SOF_HDA_REG_PP_PPCTL, mask, 0); spin_unlock_irq(&bus->reg_lock); + stream->substream = NULL; + return 0; } From 8cce6569e417f557781fe7f3a84667e611c3a160 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 12 Mar 2020 10:52:13 +0100 Subject: [PATCH 266/415] ASoC: (cosmetic) simplify dpcm_prune_paths() Currently dpcm_prune_paths() has up to 4 nested condition and loop levels, which forces the code to use flags for flow control. Extracting widget status verification code from dpcm_prune_paths() into a separate function simplifies the code. Signed-off-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20200312095214.15126-2-guennadi.liakhovetski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 63 +++++++++++++++++++++++---------------------- 1 file changed, 32 insertions(+), 31 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 733d7e8a0e5512..bf4c5dc903ce59 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1626,45 +1626,46 @@ void dpcm_path_put(struct snd_soc_dapm_widget_list **list) snd_soc_dapm_dai_free_widgets(list); } -static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, - struct snd_soc_dapm_widget_list **list_) +static bool dpcm_be_is_active(struct snd_soc_dpcm *dpcm, int stream, + struct snd_soc_dapm_widget_list *list) { - struct snd_soc_dpcm *dpcm; - struct snd_soc_dapm_widget_list *list = *list_; struct snd_soc_dapm_widget *widget; struct snd_soc_dai *dai; - int prune = 0; - int do_prune; + unsigned int i; - /* Destroy any old FE <--> BE connections */ - for_each_dpcm_be(fe, stream, dpcm) { - unsigned int i; + /* is there a valid CPU DAI widget for this BE */ + for_each_rtd_cpu_dais(dpcm->be, i, dai) { + widget = snd_soc_dai_get_widget(dai, stream); - /* is there a valid CPU DAI widget for this BE */ - do_prune = 1; - for_each_rtd_cpu_dais(dpcm->be, i, dai) { - widget = snd_soc_dai_get_widget(dai, stream); + /* + * The BE is pruned only if none of the cpu_dai + * widgets are in the active list. + */ + if (widget && widget_in_list(list, widget)) + return true; + } - /* - * The BE is pruned only if none of the cpu_dai - * widgets are in the active list. - */ - if (widget && widget_in_list(list, widget)) - do_prune = 0; - } - if (!do_prune) - continue; + /* is there a valid CODEC DAI widget for this BE */ + for_each_rtd_codec_dais(dpcm->be, i, dai) { + widget = snd_soc_dai_get_widget(dai, stream); - /* is there a valid CODEC DAI widget for this BE */ - do_prune = 1; - for_each_rtd_codec_dais(dpcm->be, i, dai) { - widget = snd_soc_dai_get_widget(dai, stream); + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + return true; + } - /* prune the BE if it's no longer in our active list */ - if (widget && widget_in_list(list, widget)) - do_prune = 0; - } - if (!do_prune) + return false; +} + +static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_dpcm *dpcm; + int prune = 0; + + /* Destroy any old FE <--> BE connections */ + for_each_dpcm_be(fe, stream, dpcm) { + if (dpcm_be_is_active(dpcm, stream, *list_)) continue; dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n", From f17a14789e55f45514d1d72a4e51dcc6bdd8d463 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 12 Mar 2020 10:52:14 +0100 Subject: [PATCH 267/415] ASoC: export DPCM runtime update functions This makes DPCM runtime update functions available for external calling. As an example, virtualised ASoC component drivers may need to call these when managing shared DAPM routes that are used by more than one driver (i.e. when host driver and guest drivers have a DAPM path from guest PCM to host DAI where some parts are owned by host driver and others by guest driver). Signed-off-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20200312095214.15126-3-guennadi.liakhovetski@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 4 ++-- sound/soc/soc-dapm.c | 8 ++++---- sound/soc/soc-pcm.c | 5 ++++- 3 files changed, 10 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 40223577ec4a70..0f6c50b17bba8d 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -132,8 +132,8 @@ int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, struct snd_pcm_substream * snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream); -/* internal use only */ -int soc_dpcm_runtime_update(struct snd_soc_card *); +/* update audio routing between PCMs and any DAI links */ +int snd_soc_dpcm_runtime_update(struct snd_soc_card *card); #ifdef CONFIG_DEBUG_FS void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e00a465a7c32ce..d5eb52fe115b4f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2291,7 +2291,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(card); + snd_soc_dpcm_runtime_update(card); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); @@ -2356,7 +2356,7 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(card); + snd_soc_dpcm_runtime_update(card); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); @@ -3396,7 +3396,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(card); + snd_soc_dpcm_runtime_update(card); return change; } @@ -3501,7 +3501,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(card); + snd_soc_dpcm_runtime_update(card); return change; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index bf4c5dc903ce59..2b915f41e955b2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -295,6 +295,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { snd_soc_runtime_action(rtd, stream, 1); } +EXPORT_SYMBOL_GPL(snd_soc_runtime_activate); /** * snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components @@ -310,6 +311,7 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { snd_soc_runtime_action(rtd, stream, -1); } +EXPORT_SYMBOL_GPL(snd_soc_runtime_deactivate); /** * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay @@ -2969,7 +2971,7 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) /* Called by DAPM mixer/mux changes to update audio routing between PCMs and * any DAI links. */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +int snd_soc_dpcm_runtime_update(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *fe; int ret = 0; @@ -2993,6 +2995,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) mutex_unlock(&card->mutex); return ret; } +EXPORT_SYMBOL_GPL(snd_soc_dpcm_runtime_update); static void dpcm_fe_dai_cleanup(struct snd_pcm_substream *fe_substream) { From b239d0c238126f478d2fcd26ad8ffc346547ce67 Mon Sep 17 00:00:00 2001 From: Rob Herring Date: Wed, 11 Mar 2020 15:58:41 -0500 Subject: [PATCH 268/415] ASoC: dt-bindings: google, cros-ec-codec: Fix dtc warnings in example Extra dtc warnings (roughly what W=1 enables) are now enabled by default when building the binding examples. These were fixed treewide in 5.6-rc5, but the newly added google,cros-ec-codec schema adds some new warnings: Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:17.28-21.11: Warning (unit_address_vs_reg): /example-0/reserved_mem: node has a reg or ranges property, but no unit name Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:22.19-32.11: Warning (unit_address_vs_reg): /example-0/cros-ec@0: node has a unit name, but no reg property Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:26.37-31.15: Warning (unit_address_vs_reg): /example-0/cros-ec@0/ec-codec: node has a reg or ranges property, but no unit name Fixing the above, then results in: Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:26.13-23: Warning (reg_format): /example-0/cros-ec@0:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1) Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:27.37-32.15: Warning (unit_address_vs_reg): /example-0/cros-ec@0/ec-codec: node has a reg or ranges property, but no unit name Fixes: eadd54c75f1e ("dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format.") Signed-off-by: Rob Herring Reviewed-by: Enric Balletbo i Serra Cc: alsa-devel@alsa-project.org Cc: Benson Leung Cc: Mark Brown Cc: Liam Girdwood Cc: Guenter Roeck Cc: Enric Balletbo i Serra Cc: Cheng-Yi Chiang Link: https://lore.kernel.org/r/20200311205841.2710-1-robh@kernel.org Signed-off-by: Mark Brown --- .../bindings/sound/google,cros-ec-codec.yaml | 27 +++++++++++-------- 1 file changed, 16 insertions(+), 11 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index 94a85d0cbf433a..c84e656afb0a6c 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -44,19 +44,24 @@ additionalProperties: false examples: - | - reserved_mem: reserved_mem { + reserved_mem: reserved-mem@52800000 { compatible = "shared-dma-pool"; - reg = <0 0x52800000 0 0x100000>; + reg = <0x52800000 0x100000>; no-map; }; - cros-ec@0 { - compatible = "google,cros-ec-spi"; - #address-cells = <2>; - #size-cells = <1>; - cros_ec_codec: ec-codec { - compatible = "google,cros-ec-codec"; - #sound-dai-cells = <1>; - reg = <0x0 0x10500000 0x80000>; - memory-region = <&reserved_mem>; + spi { + #address-cells = <1>; + #size-cells = <0>; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + #address-cells = <2>; + #size-cells = <1>; + reg = <0>; + cros_ec_codec: ec-codec@10500000 { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; }; }; From c42464a4e67383d8daeeaa668ff875e399a38105 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 12 Mar 2020 08:22:39 -0400 Subject: [PATCH 269/415] ASoC: topology: Perform component check upfront MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Function soc_tplg_dbytes_create(), calls soc_tplg_init_kcontrol() to perform additional driver specific initialization. While soc_tplg_init_kcontrol() ensures that component is valid before invoking ops->control_load, there is no such check at the end of soc_tplg_dbytes_create() where list_add() is used. Also in quite a few places, there is reference of tplg->comp->dapm or tplg->comp->card, without any checks for tplg->comp. In consequence of the above this may lead to referencing NULL pointer. This allows for removal of now unnecessary checks. Signed-off-by: Amadeusz Sławiński Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20200312122239.14489-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 22c38df40d5ab9..c98766957c1081 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -251,7 +251,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, { int ret = 0; - if (tplg->comp && tplg->ops && tplg->ops->vendor_load) + if (tplg->ops && tplg->ops->vendor_load) ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", @@ -283,7 +283,7 @@ static int soc_tplg_vendor_load(struct soc_tplg *tplg, static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { - if (tplg->comp && tplg->ops && tplg->ops->widget_load) + if (tplg->ops && tplg->ops->widget_load) return tplg->ops->widget_load(tplg->comp, tplg->index, w, tplg_w); @@ -295,7 +295,7 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { - if (tplg->comp && tplg->ops && tplg->ops->widget_ready) + if (tplg->ops && tplg->ops->widget_ready) return tplg->ops->widget_ready(tplg->comp, tplg->index, w, tplg_w); @@ -307,7 +307,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg, struct snd_soc_dai_driver *dai_drv, struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - if (tplg->comp && tplg->ops && tplg->ops->dai_load) + if (tplg->ops && tplg->ops->dai_load) return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, pcm, dai); @@ -318,7 +318,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg, static int soc_tplg_dai_link_load(struct soc_tplg *tplg, struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { - if (tplg->comp && tplg->ops && tplg->ops->link_load) + if (tplg->ops && tplg->ops->link_load) return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; @@ -327,7 +327,7 @@ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, /* tell the component driver that all firmware has been loaded in this request */ static void soc_tplg_complete(struct soc_tplg *tplg) { - if (tplg->comp && tplg->ops && tplg->ops->complete) + if (tplg->ops && tplg->ops->complete) tplg->ops->complete(tplg->comp); } @@ -684,7 +684,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_bind_event); static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { - if (tplg->comp && tplg->ops && tplg->ops->control_load) + if (tplg->ops && tplg->ops->control_load) return tplg->ops->control_load(tplg->comp, tplg->index, k, hdr); @@ -1174,7 +1174,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, static int soc_tplg_add_route(struct soc_tplg *tplg, struct snd_soc_dapm_route *route) { - if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load) + if (tplg->ops && tplg->ops->dapm_route_load) return tplg->ops->dapm_route_load(tplg->comp, tplg->index, route); @@ -2563,7 +2563,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, } /* pass control to component driver for optional further init */ - if (tplg->comp && tplg->ops && tplg->ops->manifest) + if (tplg->ops && tplg->ops->manifest) ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ @@ -2735,6 +2735,10 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, struct soc_tplg tplg; int ret; + /* component needs to exist to keep and reference data while parsing */ + if (!comp) + return -EINVAL; + /* setup parsing context */ memset(&tplg, 0, sizeof(tplg)); tplg.fw = fw; From 578194290d0bf85b087e73e35ea74574012cfd96 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Mar 2020 14:03:34 +0100 Subject: [PATCH 270/415] ASoC: wm_adsp: Use scnprintf() for the limited buffer output snprintf() is a hard-to-use function, it's especially difficult to use it for concatenating substrings in a buffer with a limited size. Since snprintf() returns the would-be-output size, not the actual size, the subsequent use of snprintf() may point to the incorrect position. Use scnprintf() instead for fixing such potential errors. Signed-off-by: Takashi Iwai Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20200313130334.9028-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 2a9b610f6d435e..9e5b6c4ac4750c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1432,12 +1432,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, subname = NULL; /* don't append subname */ break; case 2: - ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s%c %.12s %x", dsp->name, *region_name, wm_adsp_fw_text[dsp->fw], alg_region->alg); break; default: - ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %.12s %x", dsp->name, wm_adsp_fw_text[dsp->fw], alg_region->alg); break; From 5b7ddb86e61311f711eaa091dc3e9d8ccfc08e3a Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Fri, 13 Mar 2020 10:38:50 +0800 Subject: [PATCH 271/415] ASoC: rt5682: Revise the DAC1 volume setting The max volume of the DAC1 Playback Volume is 0dB. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200313023850.28875-2-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index ae6f6121bc1b0f..063dd338539d58 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1146,7 +1146,7 @@ static void rt5682_jack_detect_handler(struct work_struct *work) static const struct snd_kcontrol_new rt5682_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, - RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), + RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 87, 0, dac_vol_tlv), /* IN Boost Volume */ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, From 296a37fd029dd8a911396e3c28e0bbd5a2720a5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Mar 2020 14:02:23 +0100 Subject: [PATCH 272/415] ALSA: pcm: Fix superfluous snprintf() usage show_pcm_class() returns obviously a short string that can't overflow PAGE_SIZE. And even if it were to overflow, using snprintf() there is just wrong, as it doesn't return the correct size. So simplify with sprintf() instead. Link: https://lore.kernel.org/r/20200313130223.8908-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index a141a301369f5d..b6d2331a82f70a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1019,7 +1019,7 @@ static ssize_t show_pcm_class(struct device *dev, str = "none"; else str = strs[pcm->dev_class]; - return snprintf(buf, PAGE_SIZE, "%s\n", str); + return sprintf(buf, "%s\n", str); } static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); From 0a7efa14e61ae2132fd718887deb28479b33386b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Mar 2020 14:02:41 +0100 Subject: [PATCH 273/415] ALSA: hda: Use scnprintf() for string truncation snd_hdac_codec_modalias() truncates the string to the given size and returns its size, but it returned a wrong size from snprintf(). snprintf() returns the would-be-output size, not the actual size. Use scnprintf() instead to return the correct size. Link: https://lore.kernel.org/r/20200313130241.8970-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdac_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 9a526aeef8da53..e3119f5cb0d511 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -204,7 +204,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); */ int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size) { - return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", + return scnprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", codec->vendor_id, codec->revision_id, codec->type); } EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias); From e81d47e94c569f537e008ede59e70e4f27904c86 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 12 Mar 2020 15:06:17 -0500 Subject: [PATCH 274/415] ASoC: SOF: Intel: hda-dai: add stream capability snd_soc_dai_stream_valid() will check if the stream is valid by testing stream->channels_min. So we do need the information in dai driver. The stream name is not added since we want to sure playback_widget/capture_widget will be created by topology. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre Bossart Reviewed-by: Guennadi Liakhovetski Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200312200622.24477-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 96 +++++++++++++++++++++++++++++++++++ 1 file changed, 96 insertions(+) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index ed5e7d2c0d432a..b9e3ce65e778e3 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -422,56 +422,152 @@ static struct snd_soc_cdai_ops sof_probe_compr_ops = { struct snd_soc_dai_driver skl_dai[] = { { .name = "SSP0 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP1 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP2 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP3 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP4 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP5 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "DMIC01 Pin", + .capture = { + .channels_min = 1, + .channels_max = 4, + }, }, { .name = "DMIC16k Pin", + .capture = { + .channels_min = 1, + .channels_max = 4, + }, }, #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) { .name = "iDisp1 Pin", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "iDisp2 Pin", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "iDisp3 Pin", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "iDisp4 Pin", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "Analog CPU DAI", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 16, + }, + .capture = { + .channels_min = 1, + .channels_max = 16, + }, }, { .name = "Digital CPU DAI", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 16, + }, + .capture = { + .channels_min = 1, + .channels_max = 16, + }, }, { .name = "Alt Analog CPU DAI", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 16, + }, + .capture = { + .channels_min = 1, + .channels_max = 16, + }, }, #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) { From 4ea25785259a9e7a542a5e8ceb8b208ae6929739 Mon Sep 17 00:00:00 2001 From: Karol Trzcinski Date: Thu, 12 Mar 2020 15:06:18 -0500 Subject: [PATCH 275/415] ASoC: SOF: Make sof_ipc_ext_data enum more rigid It's a part of ABI interface, so enum value shouldn't change for example after removing some old enum code. Signed-off-by: Karol Trzcinski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312200622.24477-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/info.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h index 1c560144996ca4..cc3b50b6ae52aa 100644 --- a/include/sound/sof/info.h +++ b/include/sound/sof/info.h @@ -28,9 +28,9 @@ /* extended data types that can be appended onto end of sof_ipc_fw_ready */ enum sof_ipc_ext_data { - SOF_IPC_EXT_DMA_BUFFER = 0, - SOF_IPC_EXT_WINDOW, - SOF_IPC_EXT_CC_INFO, + SOF_IPC_EXT_DMA_BUFFER = 0, + SOF_IPC_EXT_WINDOW = 1, + SOF_IPC_EXT_CC_INFO = 2, }; /* FW version - SOF_IPC_GLB_VERSION */ From a6096f88a0b344d792606ebfaf1ef1ec2d7e0655 Mon Sep 17 00:00:00 2001 From: Karol Trzcinski Date: Thu, 12 Mar 2020 15:06:19 -0500 Subject: [PATCH 276/415] ASoC: SOF: Remove SOF_IPC_EXT_DMA_BUFFER This enum code, and what's more important, related structures is unused in whole source code, so it shouldn't be kept. Signed-off-by: Karol Trzcinski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312200622.24477-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/info.h | 18 +----------------- sound/soc/sof/loader.c | 3 --- sound/soc/sof/sof-priv.h | 1 - 3 files changed, 1 insertion(+), 21 deletions(-) diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h index cc3b50b6ae52aa..438a11fcf272b2 100644 --- a/include/sound/sof/info.h +++ b/include/sound/sof/info.h @@ -28,7 +28,7 @@ /* extended data types that can be appended onto end of sof_ipc_fw_ready */ enum sof_ipc_ext_data { - SOF_IPC_EXT_DMA_BUFFER = 0, + SOF_IPC_EXT_UNUSED = 0, SOF_IPC_EXT_WINDOW = 1, SOF_IPC_EXT_CC_INFO = 2, }; @@ -83,22 +83,6 @@ struct sof_ipc_ext_data_hdr { uint32_t type; /**< SOF_IPC_EXT_ */ } __packed; -struct sof_ipc_dma_buffer_elem { - struct sof_ipc_hdr hdr; - uint32_t type; /**< SOF_IPC_REGION_ */ - uint32_t id; /**< platform specific - used to map to host memory */ - struct sof_ipc_host_buffer buffer; -} __packed; - -/* extended data DMA buffers for IPC, trace and debug */ -struct sof_ipc_dma_buffer_data { - struct sof_ipc_ext_data_hdr ext_hdr; - uint32_t num_buffers; - - /* host files in buffer[n].buffer */ - struct sof_ipc_dma_buffer_elem buffer[]; -} __packed; - struct sof_ipc_window_elem { struct sof_ipc_hdr hdr; uint32_t type; /**< SOF_IPC_REGION_ */ diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index fc4ab51bacf477..67fc95ace42b50 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -95,9 +95,6 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) /* process structure data */ switch (ext_hdr->type) { - case SOF_IPC_EXT_DMA_BUFFER: - ret = 0; - break; case SOF_IPC_EXT_WINDOW: ret = get_ext_windows(sdev, ext_hdr); break; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 5d16f668d16aa7..38dce54755a6c9 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -415,7 +415,6 @@ struct snd_sof_dev { u32 enabled_cores_mask; /* keep track of enabled cores */ /* FW configuration */ - struct sof_ipc_dma_buffer_data *info_buffer; struct sof_ipc_window *info_window; /* IPC timeouts in ms */ From 9b65b2a80e700c0dbc5c554d198e50a9e798a6d0 Mon Sep 17 00:00:00 2001 From: Amery Song Date: Thu, 12 Mar 2020 15:06:20 -0500 Subject: [PATCH 277/415] ASoC: SOF: Intel: hda: remove unnecessary ROM IPC filter function The HDA_DSP_IPC_PURGE_FW IPC from ROM is already handled in cl_dsp_init(), and as IPC IRQ is disabled at this stage, this IPC will be never received in the IRQ thread. The function hda_dsp_ipc_is_sof for filtering the ROM IPC can be removed safely. Signed-off-by: Amery Song Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Keyon Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200312200622.24477-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ipc.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 922052883b0ad5..a6052849555196 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -125,12 +125,6 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) } -static bool hda_dsp_ipc_is_sof(uint32_t msg) -{ - return (msg & (HDA_DSP_IPC_PURGE_FW | 0xf << 9)) != msg || - (msg & HDA_DSP_IPC_PURGE_FW) != HDA_DSP_IPC_PURGE_FW; -} - /* IPC handler thread */ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) { @@ -176,11 +170,9 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) */ spin_lock_irq(&sdev->ipc_lock); - /* handle immediate reply from DSP core - ignore ROM messages */ - if (hda_dsp_ipc_is_sof(msg)) { - hda_dsp_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, msg); - } + /* handle immediate reply from DSP core */ + hda_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, msg); /* wake up sleeper if we are loading code */ if (sdev->code_loading) { From 828c2f7871d8f8051c7f412c74115ef2c583b1ce Mon Sep 17 00:00:00 2001 From: Amery Song Date: Thu, 12 Mar 2020 15:06:21 -0500 Subject: [PATCH 278/415] ASoC: SOF: Intel: remove unnecessary waitq before loading firmware The HDA_DSP_IPC_PURGE_FW IPC from ROM is already handled in cl_dsp_init(), and it will never be received in the IRQ thread, so the wait condition on this IPC will never be satisfied. The wait before loading firmware is redundant and can be removed safely. Signed-off-by: Amery Song Signed-off-by: Pierre-Louis Bossart Reviewed-by: Keyon Jie Reviewed-by: Kai Vehmanen Reviewed-by: Pierre Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20200312200622.24477-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 5 ----- sound/soc/sof/intel/hda-ipc.c | 6 ------ sound/soc/sof/intel/hda-loader.c | 3 --- sound/soc/sof/intel/hda.c | 3 --- sound/soc/sof/intel/hda.h | 1 - sound/soc/sof/loader.c | 3 --- sound/soc/sof/sof-priv.h | 4 ---- 7 files changed, 25 deletions(-) diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 05125cb0be6e28..e427d00eca71fd 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -65,11 +65,6 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) hda_dsp_ipc_get_reply(sdev); snd_sof_ipc_reply(sdev, msg); - if (sdev->code_loading) { - sdev->code_loading = 0; - wake_up(&sdev->waitq); - } - cnl_ipc_dsp_done(sdev); spin_unlock_irq(&sdev->ipc_lock); diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index a6052849555196..6062bb6011fb7b 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -174,12 +174,6 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) hda_dsp_ipc_get_reply(sdev); snd_sof_ipc_reply(sdev, msg); - /* wake up sleeper if we are loading code */ - if (sdev->code_loading) { - sdev->code_loading = 0; - wake_up(&sdev->waitq); - } - /* set the done bit */ hda_dsp_ipc_dsp_done(sdev); diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 03b05d7f66da3d..0633b76dab4910 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -179,9 +179,6 @@ static int cl_trigger(struct snd_sof_dev *sdev, /* code loader is special case that reuses stream ops */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: - wait_event_timeout(sdev->waitq, !sdev->code_loading, - HDA_DSP_CL_TRIGGER_TIMEOUT); - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, 1 << hstream->index, 1 << hstream->index); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7ca887041a3441..b2681245daafb7 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -585,9 +585,6 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) hda_dsp_ctrl_ppcap_enable(sdev, true); hda_dsp_ctrl_ppcap_int_enable(sdev, true); - /* initialize waitq for code loading */ - init_waitqueue_head(&sdev->waitq); - /* set default mailbox offset for FW ready message */ sdev->dsp_box.offset = HDA_DSP_MBOX_UPLINK_OFFSET; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 537c0a930a15c8..2a57fc9f3e5895 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -175,7 +175,6 @@ * value cannot be read back within the specified time. */ #define HDA_DSP_STREAM_RUN_TIMEOUT 300 -#define HDA_DSP_CL_TRIGGER_TIMEOUT 300 #define HDA_DSP_SPIB_ENABLE 1 #define HDA_DSP_SPIB_DISABLE 0 diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 67fc95ace42b50..1f2e0be812bd9d 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -466,9 +466,6 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) const char *fw_filename; int ret; - /* set code loading condition to true */ - sdev->code_loading = 1; - /* Don't request firmware again if firmware is already requested */ if (plat_data->fw) return 0; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 38dce54755a6c9..a4b297c842dfce 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -421,10 +421,6 @@ struct snd_sof_dev { int ipc_timeout; int boot_timeout; - /* Wait queue for code loading */ - wait_queue_head_t waitq; - int code_loading; - #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) unsigned int extractor_stream_tag; #endif From c59aca98c912e570e907b448d4f1b1b49ef9572e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 12 Mar 2020 15:06:22 -0500 Subject: [PATCH 279/415] ASoC: SOF: topology: connect dai widget to all cpu-dais Extend code from single cpu-dai to multi-dai Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312200622.24477-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9f4f8868b3860b..058de94fb8cf63 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1240,6 +1240,8 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, { struct snd_soc_card *card = scomp->card; struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *cpu_dai; + int i; list_for_each_entry(rtd, &card->rtd_list, list) { dev_vdbg(scomp->dev, "tplg: check widget: %s stream: %s dai stream: %s\n", @@ -1254,13 +1256,15 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, switch (w->id) { case snd_soc_dapm_dai_out: - rtd->cpu_dai->capture_widget = w; + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + cpu_dai->capture_widget = w; dai->name = rtd->dai_link->name; dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n", w->name, rtd->dai_link->name); break; case snd_soc_dapm_dai_in: - rtd->cpu_dai->playback_widget = w; + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + cpu_dai->playback_widget = w; dai->name = rtd->dai_link->name; dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n", w->name, rtd->dai_link->name); From 5c82813ce43e3a2017c43b32a57dc7a8802d9ad4 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 12 Mar 2020 14:48:50 -0500 Subject: [PATCH 280/415] ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI Having a reverse dependency to a config that has its own additional dependencies, is generally not recommended. And this applies to select statements for SND_HDA_CODEC_HDMI, e.g. the case where SND_HDA and SND_SOC_SOF_HDA are built as modules, but the machine driver is built-in, leading to compile errors (reported as i386-randconfig-e003-20200206). Give up on trying to define different dependencies based on SOF/SST selection, and simply add a "depends on" for SND_HDA_CODEC_HDMI. This fixes the issue with randconfigs. Only downside is that SND_HDA_CODEC_HDMI may be built unnecessarily in some cases, but this seems like the lesser evil. Fixes: aa2b4a5 ('ASoC: Intel: boards: fix incorrect HDMI Kconfig dependency') Reported-by: kbuild test robot Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 755e1de19df9e3..67d85a7be5594c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -289,7 +289,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON @@ -302,6 +301,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Broxton-P platforms @@ -402,6 +402,7 @@ config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH tristate "GLK with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Geminilake platforms @@ -413,10 +414,10 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT5682 select SND_SOC_MAX98357A select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for Geminilake platforms @@ -430,7 +431,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC + depends on SND_HDA_CODEC_HDMI select SND_SOC_HDAC_HDMI select SND_SOC_DMIC # SND_SOC_HDAC_HDA is already selected @@ -448,9 +449,9 @@ config SND_SOC_INTEL_SOF_RT5682_MACH depends on I2C && ACPI depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT5682 select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for SOF platforms @@ -490,11 +491,11 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH tristate "CML with RT1011 and RT5682 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT1011 select SND_SOC_RT5682 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platform with RT1011 + RT5682 I2S codec. @@ -509,10 +510,10 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH tristate "SOF with DA7219 and MAX98373 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_DA7219 select SND_SOC_MAX98373 select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platforms with DA7219 + MAX98373 I2S audio codec. From 4399afd21a0169a125a16c2dbba68f7d9a2cffdb Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 12 Mar 2020 14:48:51 -0500 Subject: [PATCH 281/415] ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI Having a reverse dependency to a config that has its own additional dependencies, is generally not recommended. And this applies to select statements for SND_HDA_CODEC_HDMI, e.g. the case where SND_HDA and SND_SOC_SOF_HDA are built as modules, but the machine driver is built-in, leading to compile errors (reported as i386-randconfig-e003-20200206). Give up on trying to define different dependencies based on SOF/SST selection, and simply add a "depends on" for SND_HDA_CODEC_HDMI. This fixes the issue with randconfigs. Only downside is that SND_HDA_CODEC_HDMI may be built unnecessarily in some cases, but this seems like the lesser evil. Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 67d85a7be5594c..6833ef548710da 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -464,8 +464,8 @@ config SND_SOC_INTEL_SOF_PCM512x_MACH depends on I2C && ACPI depends on (SND_SOC_SOF_HDA_AUDIO_CODEC && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + depends on SND_HDA_CODEC_HDMI select SND_SOC_PCM512x_I2C - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platforms with TI PCM512x I2S audio codec. From 15a5a89597e57e67d4dde1d57fa85105c5930bb3 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 12 Mar 2020 14:48:52 -0500 Subject: [PATCH 282/415] ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms Make HDMI optional for APL and later platforms. If no HDMI codec is found on the HDA bus, the graphics side driver is missing or correct codec driver is not part of kernel build, codec_mask reflects this and HDMI is disabled. The DSP topology will still have the links for HDMI, so connect these to dummy codec to avoid failures in topology loading. This change also fixes a kernel oops that was triggered if sof_pcm512x was used with SOF configured to use hdac-hdmi (can be done via "use_common_hdmi=0" or by selecting CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC=n). This is not a supported configuration. Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_pcm512x.c | 38 +++++++++++++++++++++------- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 626153bd71e779..4ce707b6eb7995 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -27,6 +27,8 @@ #define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) #define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) +#define IDISP_CODEC_MASK 0x4 + /* Default: SSP5 */ static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); @@ -40,6 +42,7 @@ struct sof_hdmi_pcm { struct sof_card_private { struct list_head hdmi_pcm_list; + bool idisp_codec; }; static int sof_pcm512x_quirk_cb(const struct dmi_system_id *id) @@ -136,6 +139,9 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (list_empty(&ctx->hdmi_pcm_list)) return -EINVAL; + if (!ctx->idisp_codec) + return 0; + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); @@ -214,7 +220,8 @@ SND_SOC_DAILINK_DEF(dmic_component, static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int dmic_be_num, - int hdmi_num) + int hdmi_num, + bool idisp_codec) { struct snd_soc_dai_link_component *idisp_components; struct snd_soc_dai_link_component *cpus; @@ -316,11 +323,19 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (!links[id].cpus->dai_name) goto devm_err; - idisp_components[i - 1].name = "ehdaudio0D2"; - idisp_components[i - 1].dai_name = devm_kasprintf(dev, - GFP_KERNEL, - "intel-hdmi-hifi%d", - i); + /* + * topology cannot be loaded if codec is missing, so + * use the dummy codec if needed + */ + if (idisp_codec) { + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = + devm_kasprintf(dev, GFP_KERNEL, + "intel-hdmi-hifi%d", i); + } else { + idisp_components[i - 1].name = "snd-soc-dummy"; + idisp_components[i - 1].dai_name = "snd-soc-dummy-dai"; + } if (!idisp_components[i - 1].dai_name) goto devm_err; @@ -341,8 +356,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, static int sof_audio_probe(struct platform_device *pdev) { + struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; - struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; int ret, ssp_codec; @@ -360,6 +375,11 @@ static int sof_audio_probe(struct platform_device *pdev) } else { dmic_be_num = 2; #if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) + if (mach->mach_params.common_hdmi_codec_drv && + (mach->mach_params.codec_mask & IDISP_CODEC_MASK)) + ctx->idisp_codec = true; + + /* links are always present in topology */ hdmi_num = 3; #endif } @@ -374,7 +394,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, - dmic_be_num, hdmi_num); + dmic_be_num, hdmi_num, + ctx->idisp_codec); if (!dai_links) return -ENOMEM; @@ -383,7 +404,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_pcm512x.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_pcm512x, From 42c67753cae12f68b936901792ecd367e8dfca9b Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 12 Mar 2020 14:48:53 -0500 Subject: [PATCH 283/415] ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC To help user-space with HDMI codec driver transition, both a kernel module parameter and a kernel option were initially provided to configure default behaviour of SOF on Intel hardware with commit 139c7febad1a ("ASoC: SOF: Intel: add support for snd-hda-codec-hdmi"). As hdac-hdmi is already now lagging in features compared to snd-hda-codec-hdmi, move ahead with the transition and remove the build option to select between the two, and instead default to snd-hda-codec-hdmi if it is enabled in kernel build. The old behaviour of using hdac-hdmi driver can still be forced via the kernel module parameter. Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 11 ----------- sound/soc/sof/intel/hda.c | 3 +-- 2 files changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 3bc64dee7c39c3..c9a2bee4b55cd0 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -324,17 +324,6 @@ config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1 Say Y if you want to enable DMI Link L1 If unsure, select "N". -config SND_SOC_SOF_HDA_COMMON_HDMI_CODEC - bool "SOF common HDA HDMI codec driver" - depends on SND_SOC_SOF_HDA_LINK - depends on SND_HDA_CODEC_HDMI - default SND_HDA_CODEC_HDMI - help - This adds support for HDMI audio by using the common HDA - HDMI/DisplayPort codec driver. - Say Y if you want to use the common codec driver with SOF. - If unsure select "Y". - endif ## SND_SOC_SOF_HDA_COMMON config SND_SOC_SOF_HDA_LINK_BASELINE diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7ca887041a3441..1de750a1dd19df 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -54,8 +54,7 @@ static int hda_dmic_num = -1; module_param_named(dmic_num, hda_dmic_num, int, 0444); MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number"); -static bool hda_codec_use_common_hdmi = - IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC); +static bool hda_codec_use_common_hdmi = IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI); module_param_named(use_common_hdmi, hda_codec_use_common_hdmi, bool, 0444); MODULE_PARM_DESC(use_common_hdmi, "SOF HDA use common HDMI codec driver"); #endif From c4aafb337d311556d4448f8de857ca2683f5ca5b Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 12 Mar 2020 14:48:54 -0500 Subject: [PATCH 284/415] ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations Remove several redundant variable initialisations. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index e6558475e006d0..fba9b749839ded 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1998,11 +1998,11 @@ static struct hdac_hdmi_drv_data intel_drv_data = { static int hdac_hdmi_dev_probe(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi_priv = NULL; + struct hdac_hdmi_priv *hdmi_priv; struct snd_soc_dai_driver *hdmi_dais = NULL; - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int num_dais = 0; - int ret = 0; + int ret; struct hdac_driver *hdrv = drv_to_hdac_driver(hdev->dev.driver); const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv); From ca841843a3a8038494e48968c2fd1c7ec5473ce3 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 12 Mar 2020 14:48:55 -0500 Subject: [PATCH 285/415] ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations Variables, used as loop iterators, don't need to be initialised. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f755ca2484cff7..1aa8114a4f77cb 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -359,7 +359,7 @@ static int skl_resume(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct hdac_bus *bus = pci_get_drvdata(pci); struct skl_dev *skl = bus_to_skl(bus); - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int ret; /* @@ -794,7 +794,7 @@ static void skl_probe_work(struct work_struct *work) { struct skl_dev *skl = container_of(work, struct skl_dev, probe_work); struct hdac_bus *bus = skl_to_bus(skl); - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int err; if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { From 42432196cfb01500ec058e8acc8dcfcf27eb76c9 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 12 Mar 2020 14:48:56 -0500 Subject: [PATCH 286/415] ASoC: Intel: (cosmetic) simplify structure member access Fix a clumsy structure member dereference in all machine drivers. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 19 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index 1a302436d45037..058abf3eec501f 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -298,7 +298,7 @@ static int bdw_rt5650_probe(struct platform_device *pdev) return -ENOMEM; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card, mach->mach_params.platform); diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index bb643c99069dfd..a94f498388e145 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -412,7 +412,7 @@ static int bdw_rt5677_probe(struct platform_device *pdev) } /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5677_card, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index b9c12e24c70bc0..25178000c6a55c 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -283,7 +283,7 @@ static int broadwell_audio_probe(struct platform_device *pdev) broadwell_rt286.dev = &pdev->dev; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 9177401c37a53d..061462248bce02 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -721,7 +721,7 @@ static int broxton_audio_probe(struct platform_device *pdev) } /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(&broxton_audio_card, diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 4b67f261377c25..4b5e7f6dbdf1cf 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -627,7 +627,7 @@ static int broxton_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(card, diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index eda7a500cad6c3..d6b912c013fc56 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -231,7 +231,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) int ret_val = 0; int i; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; card = &bytcht_da7213_card; card->dev = &pdev->dev; diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 70bb86f3342ff9..ea119d52392659 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -553,7 +553,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* override plaform name, if required */ snd_soc_card_cht.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht, diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 501bad3976fbfc..34d4e17e329594 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -259,7 +259,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* override plaform name, if required */ snd_soc_card_cht.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index b5b016d493f187..452691db12ccb7 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -539,7 +539,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (!drv) return -ENOMEM; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) { if (acpi_dev_found(snd_soc_cards[i].codec_id) && diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index 2a6e5b12409923..30de502b4fbb40 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -451,7 +451,7 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) return -ENOMEM; INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 8e947bad143c86..ea1de8b3f3cdd4 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -604,7 +604,7 @@ static int geminilake_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(card, platform_name); diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 3dadf9bff796a3..6589fa56873f32 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -193,7 +193,7 @@ static int haswell_audio_probe(struct platform_device *pdev) haswell_rt5640.dev = &pdev->dev; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index f65feee1c16603..20d566e9dd9d80 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -962,7 +962,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 341bb47311a682..6493ede8930014 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -772,7 +772,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&kabylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index fe2d3a23a4efe2..3be764299ab003 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -182,7 +182,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (!mach) return -EINVAL; diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index e6de3b28d8406e..8216c15fc8da17 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -660,7 +660,7 @@ static int skylake_audio_probe(struct platform_device *pdev) skylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index c99c8b23e509b0..6f68712ffce95d 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -703,7 +703,7 @@ static int skylake_audio_probe(struct platform_device *pdev) skylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 8f44f13d284812..8c657da5fcf0fc 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -335,7 +335,7 @@ static int audio_probe(struct platform_device *pdev) card = (struct snd_soc_card *)pdev->id_entry->driver_data; card->dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(card, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5d878873a8e085..99b5a5e01e3814 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -604,7 +604,7 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; /* A speaker amp might not be present when the quirk claims one is. * Detect this via whether the machine driver match includes quirk_data. From 3f32e596b03ef50fd2c49f2e13d42e1931525f35 Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Thu, 12 Mar 2020 14:48:57 -0500 Subject: [PATCH 287/415] ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp Add Maxim MAX98360A plug-and-play Class-D amplifier support on SSP1, new card ID is sofda7219max98360a, name sof-da7219max98360a. Signed-off-by: Yong Zhi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 4 +- sound/soc/intel/boards/sof_da7219_max98373.c | 74 +++++++++++++++++-- .../intel/common/soc-acpi-intel-jsl-match.c | 19 ++++- 3 files changed, 87 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6833ef548710da..ab4ce652cc1a95 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -507,7 +507,7 @@ endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK if SND_SOC_SOF_JASPERLAKE config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH - tristate "SOF with DA7219 and MAX98373 in I2S Mode" + tristate "SOF with DA7219 and MAX98373/MAX98360A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI @@ -516,7 +516,7 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH select SND_SOC_DMIC help This adds support for ASoC machine driver for SOF platforms - with DA7219 + MAX98373 I2S audio codec. + with DA7219 + MAX98373/MAX98360A I2S audio codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 8c657da5fcf0fc..7847dd44f41b60 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -2,7 +2,7 @@ // Copyright(c) 2019 Intel Corporation. /* - * Intel SOF Machine driver for DA7219 + MAX98373 codec + * Intel SOF Machine driver for DA7219 + MAX98373/MAX98360A codec */ #include @@ -69,11 +69,14 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Right Spk"), }; +static const struct snd_kcontrol_new m98360a_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), @@ -83,15 +86,23 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "HPL" }, { "Headphone Jack", NULL, "HPR" }, - { "Left Spk", NULL, "Left BE_OUT" }, - { "Right Spk", NULL, "Right BE_OUT" }, - { "MIC", NULL, "Headset Mic" }, { "Headphone Jack", NULL, "Platform Clock" }, { "Headset Mic", NULL, "Platform Clock" }, }; +/* For MAX98373 amp */ +static const struct snd_soc_dapm_widget max98373_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_route max98373_map[] = { + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + static struct snd_soc_jack headset; static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) @@ -133,6 +144,21 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static int speaker_amp_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + + /* Add widgets */ + ret = snd_soc_dapm_new_controls(&rtd->card->dapm, max98373_widgets, + ARRAY_SIZE(max98373_widgets)); + if (ret) + return ret; + + /* Add routes */ + return snd_soc_dapm_add_routes(&rtd->card->dapm, max98373_map, + ARRAY_SIZE(max98373_map)); +} + static int ssp1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -223,6 +249,8 @@ SND_SOC_DAILINK_DEF(ssp1_amps, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI), /* Right */ COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI))); +/* For the driver-less spk amp */ +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -254,6 +282,7 @@ static struct snd_soc_dai_link dais[] = { .id = 0, .ignore_pmdown_time = 1, .no_pcm = 1, + .init = speaker_amp_init, .dpcm_playback = 1, .dpcm_capture = 1, /* IV feedback */ .ops = &ssp1_ops, @@ -320,6 +349,21 @@ static struct snd_soc_card card_da7219_m98373 = { .late_probe = card_late_probe, }; +static struct snd_soc_card card_da7219_m98360a = { + .name = "da7219max98360a", + .owner = THIS_MODULE, + .dai_link = dais, + .num_links = ARRAY_SIZE(dais), + .controls = m98360a_controls, + .num_controls = ARRAY_SIZE(m98360a_controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), + .fully_routed = true, + .late_probe = card_late_probe, +}; + static int audio_probe(struct platform_device *pdev) { static struct snd_soc_card *card; @@ -331,6 +375,17 @@ static int audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; + /* By default dais[0] is configured for max98373 */ + if (!strcmp(pdev->name, "sof_da7219_max98360a")) { + dais[0] = (struct snd_soc_dai_link) { + .name = "SSP1-Codec", + .id = 0, + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_pmdown_time = 1, + SND_SOC_DAILINK_REG(ssp1_pin, dummy, platform) }; + } + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); card = (struct snd_soc_card *)pdev->id_entry->driver_data; card->dev = &pdev->dev; @@ -351,13 +406,17 @@ static const struct platform_device_id board_ids[] = { .name = "sof_da7219_max98373", .driver_data = (kernel_ulong_t)&card_da7219_m98373, }, + { + .name = "sof_da7219_max98360a", + .driver_data = (kernel_ulong_t)&card_da7219_m98360a, + }, { } }; static struct platform_driver audio = { .probe = audio_probe, .driver = { - .name = "sof_da7219_max98373", + .name = "sof_da7219_max98_360a_373", .pm = &snd_soc_pm_ops, }, .id_table = board_ids, @@ -368,4 +427,5 @@ module_platform_driver(audio) MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver"); MODULE_AUTHOR("Yong Zhi "); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_da7219_max98360a"); MODULE_ALIAS("platform:sof_da7219_max98373"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index ed2b125f6a11a1..70f01495a1662e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -9,13 +9,30 @@ #include #include +static struct snd_soc_acpi_codecs jsl_7219_98373_codecs = { + .num_codecs = 1, + .codecs = {"MX98373"} +}; + +/* + * When adding new entry to the snd_soc_acpi_intel_jsl_machines array, + * use .quirk_data member to distinguish different machine driver, + * and keep ACPI .id field unchanged for the common codec. + */ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { { .id = "DLGS7219", .drv_name = "sof_da7219_max98373", - .machine_quirk = snd_soc_acpi_codec_list, .sof_fw_filename = "sof-jsl.ri", .sof_tplg_filename = "sof-jsl-da7219.tplg", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &jsl_7219_98373_codecs, + }, + { + .id = "DLGS7219", + .drv_name = "sof_da7219_max98360a", + .sof_fw_filename = "sof-jsl.ri", + .sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg", }, {}, }; From a79ae0f6c95652752774e66fd4fa5191d868df9f Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Thu, 12 Mar 2020 14:48:58 -0500 Subject: [PATCH 288/415] ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support This patch adds jsl_rt5682_rt1015 which supports the RT5682 headset codec and RT1015 speaker amplifier combination on JasperLake platform. Signed-off-by: Yong Zhi Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200312194859.4051-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_rt5682.c | 108 +++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 15 ++- 3 files changed, 118 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index ab4ce652cc1a95..fb8d83518c4733 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -450,6 +450,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) depends on SND_HDA_CODEC_HDMI + select SND_SOC_RT1015 select SND_SOC_RT5682 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 99b5a5e01e3814..6defe7c85c32da 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1,9 +1,9 @@ // SPDX-License-Identifier: GPL-2.0 -// Copyright(c) 2019 Intel Corporation. +// Copyright(c) 2019-2020 Intel Corporation. /* * Intel SOF Machine Driver with Realtek rt5682 Codec - * and speaker codec MAX98357A + * and speaker codec MAX98357A or RT1015. */ #include #include @@ -18,6 +18,7 @@ #include #include #include +#include "../../codecs/rt1015.h" #include "../../codecs/rt5682.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" @@ -39,6 +40,7 @@ #define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) #define SOF_RT5682_NUM_HDMIDEV(quirk) \ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) +#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -260,6 +262,42 @@ static struct snd_soc_ops sof_rt5682_ops = { .hw_params = sof_rt5682_hw_params, }; +static int sof_rt1015_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai; + int i, ret; + + if (!snd_soc_card_get_codec_dai(card, "rt1015-aif")) + return 0; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + params_rate(params) * 50, + params_rate(params) * 256); + if (ret < 0) { + dev_err(card->dev, "failed to set pll\n"); + return ret; + } + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "failed to set sysclk\n"); + return ret; + } + } + + return 0; +} + +static struct snd_soc_ops sof_rt1015_ops = { + .hw_params = sof_rt1015_hw_params, +}; + static struct snd_soc_dai_link_component platform_component[] = { { /* name might be overridden during probe */ @@ -316,12 +354,17 @@ static const struct snd_kcontrol_new sof_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Spk"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), + }; static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), }; static const struct snd_soc_dapm_widget dmic_widgets[] = { @@ -342,11 +385,22 @@ static const struct snd_soc_dapm_route speaker_map[] = { { "Spk", NULL, "Speaker" }, }; +static const struct snd_soc_dapm_route speaker_map_lr[] = { + { "Left Spk", NULL, "Left SPO" }, + { "Right Spk", NULL, "Right SPO" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) +{ + return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); +} + static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -382,6 +436,17 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static struct snd_soc_codec_conf rt1015_amp_conf[] = { + { + .dlc = COMP_CODEC_CONF("i2c-10EC1015:00"), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF("i2c-10EC1015:01"), + .name_prefix = "Right", + }, +}; + /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { .name = "rt5682", /* the sof- prefix is added by the core */ @@ -417,6 +482,17 @@ static struct snd_soc_dai_link_component max98357a_component[] = { } }; +static struct snd_soc_dai_link_component rt1015_components[] = { + { + .name = "i2c-10EC1015:00", + .dai_name = "rt1015-aif", + }, + { + .name = "i2c-10EC1015:01", + .dai_name = "rt1015-aif", + }, +}; + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, @@ -556,11 +632,18 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; links[id].id = id; - links[id].codecs = max98357a_component; - links[id].num_codecs = ARRAY_SIZE(max98357a_component); + if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { + links[id].codecs = rt1015_components; + links[id].num_codecs = ARRAY_SIZE(rt1015_components); + links[id].init = speaker_codec_init_lr; + links[id].ops = &sof_rt1015_ops; + } else { + links[id].codecs = max98357a_component; + links[id].num_codecs = ARRAY_SIZE(max98357a_component); + links[id].init = speaker_codec_init; + } links[id].platforms = platform_component; links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].init = speaker_codec_init, links[id].nonatomic = true; links[id].dpcm_playback = 1; links[id].no_pcm = 1; @@ -669,6 +752,11 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_rt5682.dai_link = dai_links; + if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { + sof_audio_card_rt5682.codec_conf = rt1015_amp_conf; + sof_audio_card_rt5682.num_configs = ARRAY_SIZE(rt1015_amp_conf); + } + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_rt5682.dev = &pdev->dev; @@ -714,6 +802,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1015", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; @@ -735,3 +832,4 @@ MODULE_AUTHOR("Sathya Prakash M R "); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); MODULE_ALIAS("platform:tgl_max98357a_rt5682"); +MODULE_ALIAS("platform:jsl_rt5682_rt1015"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 70f01495a1662e..4388a32718d8d3 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -2,7 +2,7 @@ /* * soc-apci-intel-jsl-match.c - tables and support for JSL ACPI enumeration. * - * Copyright (c) 2019, Intel Corporation. + * Copyright (c) 2019-2020, Intel Corporation. * */ @@ -14,6 +14,11 @@ static struct snd_soc_acpi_codecs jsl_7219_98373_codecs = { .codecs = {"MX98373"} }; +static struct snd_soc_acpi_codecs rt1015_spk = { + .num_codecs = 1, + .codecs = {"10EC1015"} +}; + /* * When adding new entry to the snd_soc_acpi_intel_jsl_machines array, * use .quirk_data member to distinguish different machine driver, @@ -34,6 +39,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .sof_fw_filename = "sof-jsl.ri", .sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg", }, + { + .id = "10EC5682", + .drv_name = "jsl_rt5682_rt1015", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1015_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); From 2e6529a51a8bda287ac242b2ddc8a5046a3bb7c9 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 12 Mar 2020 14:48:59 -0500 Subject: [PATCH 289/415] ASoC: Intel: don't use GFP_ATOMIC for machine driver contexts We've removed GFP_ATOMIC in all machine drivers and somehow this keeps coming back due to copy-paste. Move to GFP_KERNEL. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Guennadi Liakhovetski Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200312194859.4051-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index 30de502b4fbb40..ed6c26a256e7a3 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -446,7 +446,7 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) const char *platform_name; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 7847dd44f41b60..6d210ba06d195e 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -371,7 +371,7 @@ static int audio_probe(struct platform_device *pdev) struct card_private *ctx; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; From 88eb404ccc3efd300fb6fb6a9f8c99c1d5db3747 Mon Sep 17 00:00:00 2001 From: Kevin Li Date: Thu, 12 Mar 2020 15:32:41 -0700 Subject: [PATCH 290/415] ASoC: brcm: Add DSL/PON SoC audio driver This patch adds Broadcom DSL/PON SoC audio driver with Whistler I2S block. The SoC supported by this patch are BCM63158B0,BCM63178 and BCM47622/6755 Signed-off-by: Kevin Li Link: https://lore.kernel.org/r/20200312223242.2843-2-kevin-ke.li@broadcom.com Signed-off-by: Mark Brown --- sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 4 + sound/soc/bcm/bcm63xx-i2s-whistler.c | 317 +++++++++++++++++ sound/soc/bcm/bcm63xx-i2s.h | 90 +++++ sound/soc/bcm/bcm63xx-pcm-whistler.c | 485 +++++++++++++++++++++++++++ 5 files changed, 905 insertions(+) create mode 100644 sound/soc/bcm/bcm63xx-i2s-whistler.c create mode 100644 sound/soc/bcm/bcm63xx-i2s.h create mode 100644 sound/soc/bcm/bcm63xx-pcm-whistler.c diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 0037e96aa228fe..4218057b087426 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -17,3 +17,12 @@ config SND_SOC_CYGNUS Cygnus chips (bcm958300, bcm958305, bcm911360) If you don't know what to do here, say N. + +config SND_BCM63XX_I2S_WHISTLER + tristate "SoC Audio support for the Broadcom BCM63XX I2S module" + select REGMAP_MMIO + help + Say Y if you want to add support for ASoC audio on Broadcom + DSL/PON chips (bcm63158, bcm63178) + + If you don't know what to do here, say N diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile index b81fa421ec2722..7c2d7899603b7c 100644 --- a/sound/soc/bcm/Makefile +++ b/sound/soc/bcm/Makefile @@ -9,3 +9,7 @@ snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o +# BCM63XX Platform Support +snd-soc-63xx-objs := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o + +obj-$(CONFIG_SND_BCM63XX_I2S_WHISTLER) += snd-soc-63xx.o \ No newline at end of file diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c new file mode 100644 index 00000000000000..246a57ac667998 --- /dev/null +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -0,0 +1,317 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// linux/sound/bcm/bcm63xx-i2s-whistler.c +// BCM63xx whistler i2s driver +// Copyright (c) 2020 Broadcom Corporation +// Author: Kevin-Ke Li + +#include +#include +#include +#include +#include +#include +#include +#include "bcm63xx-i2s.h" + +#define DRV_NAME "brcm-i2s" + +static bool brcm_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG ... I2S_TX_DESC_IFF_LEN: + case I2S_TX_CFG_2 ... I2S_RX_DESC_IFF_LEN: + case I2S_RX_CFG_2 ... I2S_REG_MAX: + return true; + default: + return false; + } +} + +static bool brcm_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG ... I2S_REG_MAX: + return true; + default: + return false; + } +} + +static bool brcm_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG: + case I2S_TX_IRQ_CTL: + case I2S_TX_DESC_IFF_ADDR: + case I2S_TX_DESC_IFF_LEN: + case I2S_TX_DESC_OFF_ADDR: + case I2S_TX_DESC_OFF_LEN: + case I2S_TX_CFG_2: + case I2S_RX_CFG: + case I2S_RX_IRQ_CTL: + case I2S_RX_DESC_OFF_ADDR: + case I2S_RX_DESC_OFF_LEN: + case I2S_RX_DESC_IFF_LEN: + case I2S_RX_DESC_IFF_ADDR: + case I2S_RX_CFG_2: + return true; + default: + return false; + } +} + +static const struct regmap_config brcm_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = I2S_REG_MAX, + .writeable_reg = brcm_i2s_wr_reg, + .readable_reg = brcm_i2s_rd_reg, + .volatile_reg = brcm_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int bcm63xx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + + ret = clk_set_rate(i2s_priv->i2s_clk, params_rate(params)); + if (ret < 0) + dev_err(i2s_priv->dev, + "Can't set sample rate, err: %d\n", ret); + + return ret; +} + +static int bcm63xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + unsigned int slavemode; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + struct regmap *regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(regmap_i2s, I2S_TX_CFG, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE); + regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 1); + + /* TX and RX block each have an independent bit to indicate + * if it is generating the clock for the I2S bus. The bus + * clocks need to be generated from either the TX or RX block, + * but not both + */ + regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode); + if (slavemode & I2S_RX_SLAVE_MODE_MASK) + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_MASTER_MODE); + else + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_SLAVE_MODE); + } else { + regmap_update_bits(regmap_i2s, I2S_RX_CFG, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE); + regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 1); + + regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode); + if (slavemode & I2S_TX_SLAVE_MODE_MASK) + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, 0); + else + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, + I2S_RX_SLAVE_MODE); + } + return 0; +} + +static void bcm63xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + unsigned int enabled, slavemode; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + struct regmap *regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(regmap_i2s, I2S_TX_CFG, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 1); + regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 4); + regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 4); + + regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode); + slavemode = slavemode & I2S_TX_SLAVE_MODE_MASK; + if (!slavemode) { + regmap_read(regmap_i2s, I2S_RX_CFG, &enabled); + enabled = enabled & I2S_RX_ENABLE_MASK; + if (enabled) + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, + I2S_RX_MASTER_MODE); + } + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_SLAVE_MODE); + } else { + regmap_update_bits(regmap_i2s, I2S_RX_CFG, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 1); + regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 4); + regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 4); + + regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode); + slavemode = slavemode & I2S_RX_SLAVE_MODE_MASK; + if (!slavemode) { + regmap_read(regmap_i2s, I2S_TX_CFG, &enabled); + enabled = enabled & I2S_TX_ENABLE_MASK; + if (enabled) + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_MASTER_MODE); + } + + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, I2S_RX_SLAVE_MODE); + } +} + +static const struct snd_soc_dai_ops bcm63xx_i2s_dai_ops = { + .startup = bcm63xx_i2s_startup, + .shutdown = bcm63xx_i2s_shutdown, + .hw_params = bcm63xx_i2s_hw_params, +}; + +static struct snd_soc_dai_driver bcm63xx_i2s_dai = { + .name = DRV_NAME, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &bcm63xx_i2s_dai_ops, + .symmetric_rates = 1, + .symmetric_channels = 1, +}; + +static const struct snd_soc_component_driver bcm63xx_i2s_component = { + .name = "bcm63xx", +}; + +static int bcm63xx_i2s_dev_probe(struct platform_device *pdev) +{ + int ret = 0; + void __iomem *regs; + struct resource *r_mem, *region; + struct bcm_i2s_priv *i2s_priv; + struct regmap *regmap_i2s; + struct clk *i2s_clk; + + i2s_priv = devm_kzalloc(&pdev->dev, sizeof(*i2s_priv), GFP_KERNEL); + if (!i2s_priv) + return -ENOMEM; + + i2s_clk = devm_clk_get(&pdev->dev, "i2sclk"); + if (IS_ERR(i2s_clk)) { + dev_err(&pdev->dev, "%s: cannot get a brcm clock: %ld\n", + __func__, PTR_ERR(i2s_clk)); + return PTR_ERR(i2s_clk); + } + + r_mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r_mem) { + dev_err(&pdev->dev, "Unable to get register resource.\n"); + return -ENODEV; + } + + region = devm_request_mem_region(&pdev->dev, r_mem->start, + resource_size(r_mem), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + return -EBUSY; + } + + regs = devm_ioremap_resource(&pdev->dev, r_mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + return ret; + } + + regmap_i2s = devm_regmap_init_mmio(&pdev->dev, + regs, &brcm_i2s_regmap_config); + if (IS_ERR(regmap_i2s)) + return PTR_ERR(regmap_i2s); + + regmap_update_bits(regmap_i2s, I2S_MISC_CFG, + I2S_PAD_LVL_LOOP_DIS_MASK, + I2S_PAD_LVL_LOOP_DIS_ENABLE); + + ret = devm_snd_soc_register_component(&pdev->dev, + &bcm63xx_i2s_component, + &bcm63xx_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register the dai\n"); + return ret; + } + + i2s_priv->dev = &pdev->dev; + i2s_priv->i2s_clk = i2s_clk; + i2s_priv->regmap_i2s = regmap_i2s; + dev_set_drvdata(&pdev->dev, i2s_priv); + + ret = bcm63xx_soc_platform_probe(pdev, i2s_priv); + if (ret) + dev_err(&pdev->dev, "failed to register the pcm\n"); + + return ret; +} + +static int bcm63xx_i2s_dev_remove(struct platform_device *pdev) +{ + bcm63xx_soc_platform_remove(pdev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id snd_soc_bcm_audio_match[] = { + {.compatible = "brcm,bcm63xx-i2s"}, + { } +}; +#endif + +static struct platform_driver bcm63xx_i2s_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(snd_soc_bcm_audio_match), + }, + .probe = bcm63xx_i2s_dev_probe, + .remove = bcm63xx_i2s_dev_remove, +}; + +module_platform_driver(bcm63xx_i2s_driver); + +MODULE_AUTHOR("Kevin,Li "); +MODULE_DESCRIPTION("Broadcom DSL XPON ASOC I2S Interface"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/bcm/bcm63xx-i2s.h b/sound/soc/bcm/bcm63xx-i2s.h new file mode 100644 index 00000000000000..edc328ba53d3b0 --- /dev/null +++ b/sound/soc/bcm/bcm63xx-i2s.h @@ -0,0 +1,90 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// linux/sound/soc/bcm/bcm63xx-i2s.h +// Copyright (c) 2020 Broadcom Corporation +// Author: Kevin-Ke Li + +#ifndef __BCM63XX_I2S_H +#define __BCM63XX_I2S_H + +#define I2S_DESC_FIFO_DEPTH 8 +#define I2S_MISC_CFG (0x003C) +#define I2S_PAD_LVL_LOOP_DIS_MASK (1 << 2) +#define I2S_PAD_LVL_LOOP_DIS_ENABLE I2S_PAD_LVL_LOOP_DIS_MASK + +#define I2S_TX_ENABLE_MASK (1 << 31) +#define I2S_TX_ENABLE I2S_TX_ENABLE_MASK +#define I2S_TX_OUT_R (1 << 19) +#define I2S_TX_DATA_ALIGNMENT (1 << 2) +#define I2S_TX_DATA_ENABLE (1 << 1) +#define I2S_TX_CLOCK_ENABLE (1 << 0) + +#define I2S_TX_DESC_OFF_LEVEL_SHIFT 12 +#define I2S_TX_DESC_OFF_LEVEL_MASK (0x0F << I2S_TX_DESC_OFF_LEVEL_SHIFT) +#define I2S_TX_DESC_IFF_LEVEL_SHIFT 8 +#define I2S_TX_DESC_IFF_LEVEL_MASK (0x0F << I2S_TX_DESC_IFF_LEVEL_SHIFT) +#define I2S_TX_DESC_OFF_INTR_EN_MSK (1 << 1) +#define I2S_TX_DESC_OFF_INTR_EN I2S_TX_DESC_OFF_INTR_EN_MSK + +#define I2S_TX_CFG (0x0000) +#define I2S_TX_IRQ_CTL (0x0004) +#define I2S_TX_IRQ_EN (0x0008) +#define I2S_TX_IRQ_IFF_THLD (0x000c) +#define I2S_TX_IRQ_OFF_THLD (0x0010) +#define I2S_TX_DESC_IFF_ADDR (0x0014) +#define I2S_TX_DESC_IFF_LEN (0x0018) +#define I2S_TX_DESC_OFF_ADDR (0x001C) +#define I2S_TX_DESC_OFF_LEN (0x0020) +#define I2S_TX_CFG_2 (0x0024) +#define I2S_TX_SLAVE_MODE_SHIFT 13 +#define I2S_TX_SLAVE_MODE_MASK (1 << I2S_TX_SLAVE_MODE_SHIFT) +#define I2S_TX_SLAVE_MODE I2S_TX_SLAVE_MODE_MASK +#define I2S_TX_MASTER_MODE 0 +#define I2S_TX_INTR_MASK 0x0F + +#define I2S_RX_ENABLE_MASK (1 << 31) +#define I2S_RX_ENABLE I2S_RX_ENABLE_MASK +#define I2S_RX_IN_R (1 << 19) +#define I2S_RX_DATA_ALIGNMENT (1 << 2) +#define I2S_RX_CLOCK_ENABLE (1 << 0) + +#define I2S_RX_DESC_OFF_LEVEL_SHIFT 12 +#define I2S_RX_DESC_OFF_LEVEL_MASK (0x0F << I2S_RX_DESC_OFF_LEVEL_SHIFT) +#define I2S_RX_DESC_IFF_LEVEL_SHIFT 8 +#define I2S_RX_DESC_IFF_LEVEL_MASK (0x0F << I2S_RX_DESC_IFF_LEVEL_SHIFT) +#define I2S_RX_DESC_OFF_INTR_EN_MSK (1 << 1) +#define I2S_RX_DESC_OFF_INTR_EN I2S_RX_DESC_OFF_INTR_EN_MSK + +#define I2S_RX_CFG (0x0040) /* 20c0 */ +#define I2S_RX_IRQ_CTL (0x0044) +#define I2S_RX_IRQ_EN (0x0048) +#define I2S_RX_IRQ_IFF_THLD (0x004C) +#define I2S_RX_IRQ_OFF_THLD (0x0050) +#define I2S_RX_DESC_IFF_ADDR (0x0054) +#define I2S_RX_DESC_IFF_LEN (0x0058) +#define I2S_RX_DESC_OFF_ADDR (0x005C) +#define I2S_RX_DESC_OFF_LEN (0x0060) +#define I2S_RX_CFG_2 (0x0064) +#define I2S_RX_SLAVE_MODE_SHIFT 13 +#define I2S_RX_SLAVE_MODE_MASK (1 << I2S_RX_SLAVE_MODE_SHIFT) +#define I2S_RX_SLAVE_MODE I2S_RX_SLAVE_MODE_MASK +#define I2S_RX_MASTER_MODE 0 +#define I2S_RX_INTR_MASK 0x0F + +#define I2S_REG_MAX 0x007C + +struct bcm_i2s_priv { + struct device *dev; + struct resource *r_irq; + struct regmap *regmap_i2s; + struct clk *i2s_clk; + struct snd_pcm_substream *play_substream; + struct snd_pcm_substream *capture_substream; + struct i2s_dma_desc *play_dma_desc; + struct i2s_dma_desc *capture_dma_desc; +}; + +extern int bcm63xx_soc_platform_probe(struct platform_device *pdev, + struct bcm_i2s_priv *i2s_priv); +extern int bcm63xx_soc_platform_remove(struct platform_device *pdev); + +#endif diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c new file mode 100644 index 00000000000000..55c760f1cf4da2 --- /dev/null +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -0,0 +1,485 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// linux/sound/bcm/bcm63xx-pcm-whistler.c +// BCM63xx whistler pcm interface +// Copyright (c) 2020 Broadcom Corporation +// Author: Kevin-Ke Li + +#include +#include +#include +#include +#include +#include +#include +#include "bcm63xx-i2s.h" + + +struct i2s_dma_desc { + unsigned char *dma_area; + dma_addr_t dma_addr; + unsigned int dma_len; +}; + +struct bcm63xx_runtime_data { + int dma_len; + dma_addr_t dma_addr; + dma_addr_t dma_addr_next; +}; + +static const struct snd_pcm_hardware bcm63xx_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S32_LE, /* support S32 only */ + .period_bytes_max = 8192 - 32, + .periods_min = 1, + .periods_max = PAGE_SIZE/sizeof(struct i2s_dma_desc), + .buffer_bytes_max = 128 * 1024, + .fifo_size = 32, +}; + +static int bcm63xx_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + dma_desc = kzalloc(sizeof(*dma_desc), GFP_NOWAIT); + if (!dma_desc) + return -ENOMEM; + + snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_desc); + + return 0; +} + +static int bcm63xx_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + kfree(dma_desc); + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int bcm63xx_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd; + struct bcm_i2s_priv *i2s_priv; + struct regmap *regmap_i2s; + + rtd = substream->private_data; + i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + regmap_update_bits(regmap_i2s, + I2S_TX_IRQ_EN, + I2S_TX_DESC_OFF_INTR_EN, + I2S_TX_DESC_OFF_INTR_EN); + regmap_update_bits(regmap_i2s, + I2S_TX_CFG, + I2S_TX_ENABLE_MASK, + I2S_TX_ENABLE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_write(regmap_i2s, + I2S_TX_IRQ_EN, + 0); + regmap_update_bits(regmap_i2s, + I2S_TX_CFG, + I2S_TX_ENABLE_MASK, + 0); + break; + default: + ret = -EINVAL; + } + } else { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + regmap_update_bits(regmap_i2s, + I2S_RX_IRQ_EN, + I2S_RX_DESC_OFF_INTR_EN_MSK, + I2S_RX_DESC_OFF_INTR_EN); + regmap_update_bits(regmap_i2s, + I2S_RX_CFG, + I2S_RX_ENABLE_MASK, + I2S_RX_ENABLE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(regmap_i2s, + I2S_RX_IRQ_EN, + I2S_RX_DESC_OFF_INTR_EN_MSK, + 0); + regmap_update_bits(regmap_i2s, + I2S_RX_CFG, + I2S_RX_ENABLE_MASK, + 0); + break; + default: + ret = -EINVAL; + } + } + return ret; +} + +static int bcm63xx_pcm_prepare(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct i2s_dma_desc *dma_desc; + struct regmap *regmap_i2s; + struct bcm_i2s_priv *i2s_priv; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + uint32_t regaddr_desclen, regaddr_descaddr; + + dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_desc->dma_len = snd_pcm_lib_period_bytes(substream); + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regaddr_desclen = I2S_TX_DESC_IFF_LEN; + regaddr_descaddr = I2S_TX_DESC_IFF_ADDR; + } else { + regaddr_desclen = I2S_RX_DESC_IFF_LEN; + regaddr_descaddr = I2S_RX_DESC_IFF_ADDR; + } + + i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + regmap_i2s = i2s_priv->regmap_i2s; + + regmap_write(regmap_i2s, regaddr_desclen, dma_desc->dma_len); + regmap_write(regmap_i2s, regaddr_descaddr, dma_desc->dma_addr); + + return 0; +} + +static snd_pcm_uframes_t +bcm63xx_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + snd_pcm_uframes_t x; + struct bcm63xx_runtime_data *prtd = substream->runtime->private_data; + + if ((void *)prtd->dma_addr_next == NULL) + prtd->dma_addr_next = substream->runtime->dma_addr; + + x = bytes_to_frames(substream->runtime, + prtd->dma_addr_next - substream->runtime->dma_addr); + + return x == substream->runtime->buffer_size ? 0 : x; +} + +static int bcm63xx_pcm_mmap(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_wc(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + +} + +static int bcm63xx_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct bcm63xx_runtime_data *prtd; + + runtime->hw = bcm63xx_pcm_hardware; + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); + if (ret) + goto out; + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); + if (ret) + goto out; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + ret = -ENOMEM; + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + goto out; + + runtime->private_data = prtd; + return 0; +out: + return ret; +} + +static int bcm63xx_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct bcm63xx_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + return 0; +} + +static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv) +{ + unsigned int availdepth, ifflevel, offlevel, int_status, val_1, val_2; + struct bcm63xx_runtime_data *prtd; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct regmap *regmap_i2s; + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd; + struct bcm_i2s_priv *i2s_priv; + + i2s_priv = (struct bcm_i2s_priv *)bcm_i2s_priv; + regmap_i2s = i2s_priv->regmap_i2s; + + /* rx */ + regmap_read(regmap_i2s, I2S_RX_IRQ_CTL, &int_status); + + if (int_status & I2S_RX_DESC_OFF_INTR_EN_MSK) { + substream = i2s_priv->capture_substream; + runtime = substream->runtime; + rtd = substream->private_data; + prtd = runtime->private_data; + dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + offlevel = (int_status & I2S_RX_DESC_OFF_LEVEL_MASK) >> + I2S_RX_DESC_OFF_LEVEL_SHIFT; + while (offlevel) { + regmap_read(regmap_i2s, I2S_RX_DESC_OFF_ADDR, &val_1); + regmap_read(regmap_i2s, I2S_RX_DESC_OFF_LEN, &val_2); + offlevel--; + } + prtd->dma_addr_next = val_1 + val_2; + ifflevel = (int_status & I2S_RX_DESC_IFF_LEVEL_MASK) >> + I2S_RX_DESC_IFF_LEVEL_SHIFT; + + availdepth = I2S_DESC_FIFO_DEPTH - ifflevel; + while (availdepth) { + dma_desc->dma_addr += + snd_pcm_lib_period_bytes(substream); + dma_desc->dma_area += + snd_pcm_lib_period_bytes(substream); + if (dma_desc->dma_addr - runtime->dma_addr >= + runtime->dma_bytes) { + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + } + + prtd->dma_addr = dma_desc->dma_addr; + regmap_write(regmap_i2s, I2S_RX_DESC_IFF_LEN, + snd_pcm_lib_period_bytes(substream)); + regmap_write(regmap_i2s, I2S_RX_DESC_IFF_ADDR, + dma_desc->dma_addr); + availdepth--; + } + + snd_pcm_period_elapsed(substream); + + /* Clear interrupt by writing 0 */ + regmap_update_bits(regmap_i2s, I2S_RX_IRQ_CTL, + I2S_RX_INTR_MASK, 0); + } + + /* tx */ + regmap_read(regmap_i2s, I2S_TX_IRQ_CTL, &int_status); + + if (int_status & I2S_TX_DESC_OFF_INTR_EN_MSK) { + substream = i2s_priv->play_substream; + runtime = substream->runtime; + rtd = substream->private_data; + prtd = runtime->private_data; + dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + offlevel = (int_status & I2S_TX_DESC_OFF_LEVEL_MASK) >> + I2S_TX_DESC_OFF_LEVEL_SHIFT; + while (offlevel) { + regmap_read(regmap_i2s, I2S_TX_DESC_OFF_ADDR, &val_1); + regmap_read(regmap_i2s, I2S_TX_DESC_OFF_LEN, &val_2); + prtd->dma_addr_next = val_1 + val_2; + offlevel--; + } + + ifflevel = (int_status & I2S_TX_DESC_IFF_LEVEL_MASK) >> + I2S_TX_DESC_IFF_LEVEL_SHIFT; + availdepth = I2S_DESC_FIFO_DEPTH - ifflevel; + + while (availdepth) { + dma_desc->dma_addr += + snd_pcm_lib_period_bytes(substream); + dma_desc->dma_area += + snd_pcm_lib_period_bytes(substream); + + if (dma_desc->dma_addr - runtime->dma_addr >= + runtime->dma_bytes) { + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + } + + prtd->dma_addr = dma_desc->dma_addr; + regmap_write(regmap_i2s, I2S_TX_DESC_IFF_LEN, + snd_pcm_lib_period_bytes(substream)); + regmap_write(regmap_i2s, I2S_TX_DESC_IFF_ADDR, + dma_desc->dma_addr); + availdepth--; + } + + snd_pcm_period_elapsed(substream); + + /* Clear interrupt by writing 0 */ + regmap_update_bits(regmap_i2s, I2S_TX_IRQ_CTL, + I2S_TX_INTR_MASK, 0); + } + + return IRQ_HANDLED; +} + +static int bcm63xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bcm63xx_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_wc(pcm->card->dev, + size, &buf->addr, + GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static int bcm63xx_soc_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + struct bcm_i2s_priv *i2s_priv; + int ret; + + i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + + of_dma_configure(pcm->card->dev, pcm->card->dev->of_node, 1); + + ret = dma_coerce_mask_and_coherent(pcm->card->dev, DMA_BIT_MASK(32)); + if (ret) + goto out; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = bcm63xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + + i2s_priv->play_substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = bcm63xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + i2s_priv->capture_substream = + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + } + +out: + return ret; +} + +static void bcm63xx_pcm_free_dma_buffers(struct snd_soc_component *component, + struct snd_pcm *pcm) +{ + int stream; + struct snd_dma_buffer *buf; + struct snd_pcm_substream *substream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_wc(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static const struct snd_soc_component_driver bcm63xx_soc_platform = { + .open = bcm63xx_pcm_open, + .close = bcm63xx_pcm_close, + .hw_params = bcm63xx_pcm_hw_params, + .hw_free = bcm63xx_pcm_hw_free, + .prepare = bcm63xx_pcm_prepare, + .trigger = bcm63xx_pcm_trigger, + .pointer = bcm63xx_pcm_pointer, + .mmap = bcm63xx_pcm_mmap, + .pcm_construct = bcm63xx_soc_pcm_new, + .pcm_destruct = bcm63xx_pcm_free_dma_buffers, +}; + +int bcm63xx_soc_platform_probe(struct platform_device *pdev, + struct bcm_i2s_priv *i2s_priv) +{ + int ret; + + i2s_priv->r_irq = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!i2s_priv->r_irq) { + dev_err(&pdev->dev, "Unable to get register irq resource.\n"); + return -ENODEV; + } + + ret = devm_request_irq(&pdev->dev, i2s_priv->r_irq->start, i2s_dma_isr, + i2s_priv->r_irq->flags, "i2s_dma", (void *)i2s_priv); + if (ret) { + dev_err(&pdev->dev, + "i2s_init: failed to request interrupt.ret=%d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(&pdev->dev, + &bcm63xx_soc_platform, NULL, 0); +} + +int bcm63xx_soc_platform_remove(struct platform_device *pdev) +{ + return 0; +} + +MODULE_AUTHOR("Kevin,Li "); +MODULE_DESCRIPTION("Broadcom DSL XPON ASOC PCM Interface"); +MODULE_LICENSE("GPL v2"); From 2834a736371eab06182fcdfb0c32d23d34068764 Mon Sep 17 00:00:00 2001 From: Kevin Li Date: Thu, 12 Mar 2020 15:32:40 -0700 Subject: [PATCH 291/415] ASoC: brcm: DSL/PON SoC device tree bindings of audio driver Signed-off-by: Kevin Li Link: https://lore.kernel.org/r/20200312223242.2843-1-kevin-ke.li@broadcom.com Signed-off-by: Mark Brown --- .../bindings/sound/brcm,bcm63xx-audio.txt | 29 +++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt new file mode 100644 index 00000000000000..007f524b4d150d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt @@ -0,0 +1,29 @@ +Broadcom DSL/PON BCM63xx Audio I2S controller + +Required properties: +- compatible: Should be "brcm,bcm63xx-i2s". +- #address-cells: 32bit valued, 1 cell. +- #size-cells: 32bit valued, 0 cell. +- reg: Should contain audio registers location and length +- interrupts: Should contain the interrupt for the controller. +- clocks: Must contain an entry for each entry in clock-names. + Please refer to clock-bindings.txt. +- clock-names: One of each entry matching the clocks phandles list: + - "i2sclk" (generated clock) Required. + - "i2sosc" (fixed 200MHz clock) Required. + +(1) : The generated clock is required only when any of TX and RX + works on Master Mode. +(2) : The fixed 200MHz clock is from internal chip and always on + +Example: + + i2s: bcm63xx-i2s { + #address-cells = <1>; + #size-cells = <0>; + compatible = "brcm,bcm63xx-i2s"; + reg = <0xFF802080 0xFF>; + interrupts = ; + clocks = <&i2sclk>, <&osc>; + clock-names = "i2sclk","i2sosc"; + }; From a252d78cf772f86c2dcc40df8117d9461eed88d6 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Fri, 13 Mar 2020 10:38:49 +0800 Subject: [PATCH 292/415] ASoC: rt5682: Fine tune the HP performance in soundwire mode The setting is sync with I2C/I2S mode. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200313023850.28875-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e1df2d0765333a..f4b8af1288284a 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -3462,6 +3462,8 @@ int rt5682_io_init(struct device *dev, struct sdw_slave *slave) RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); + regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, + RT5682_PM_HP_MASK, RT5682_PM_HP_HV); /* Soundwire */ regmap_write(rt5682->regmap, RT5682_PLL2_INTERNAL, 0xa266); From 1a0f2433d7380c957d5d29e60a2eeb03ca3afe21 Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Fri, 13 Mar 2020 10:55:26 -0500 Subject: [PATCH 293/415] ASoC: max98357a: Add ACPI HID MAX98360A Maxim MAX98360A audio amplifier is functionally identical to MAX98357A, add ACPI ID "MAX98360A" for driver reuse. Signed-off-by: Yong Zhi Link: https://lore.kernel.org/r/1584114926-29287-1-git-send-email-yong.zhi@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 16313b973eaa41..eb3d8255ea6c3e 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -135,6 +135,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); #ifdef CONFIG_ACPI static const struct acpi_device_id max98357a_acpi_match[] = { { "MX98357A", 0 }, + { "MX98360A", 0 }, {}, }; MODULE_DEVICE_TABLE(acpi, max98357a_acpi_match); From 0aef31b75272a9894f7205c2c888c7dbc248ce94 Mon Sep 17 00:00:00 2001 From: Chris Wulff Date: Sat, 14 Mar 2020 12:54:48 -0400 Subject: [PATCH 294/415] ALSA: usb-audio: Fix mixer controls' USB interface for Kingston HyperX Amp (0951:16d8) Use the USB interface of the mixer that the control was created on instead of the default control interface. This fixes the Kingston HyperX AMP (0951:16d8) which has controls on two interfaces. Signed-off-by: Chris Wulff Link: https://lore.kernel.org/r/20200314165449.4086-2-crwulff@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d5c7283f6e9ff5..721d12130d0cb4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -292,6 +292,11 @@ static int uac2_ctl_value_size(int val_type) * retrieve a mixer value */ +static inline int mixer_ctrl_intf(struct usb_mixer_interface *mixer) +{ + return get_iface_desc(mixer->hostif)->bInterfaceNumber; +} + static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { @@ -306,7 +311,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, return -EIO; while (timeout-- > 0) { - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, val_len); @@ -354,7 +359,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, if (ret) goto error; - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, size); @@ -479,7 +484,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, return -EIO; while (timeout-- > 0) { - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, @@ -1209,7 +1214,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { usb_audio_err(cval->head.mixer->chip, "%d:%d: cannot get min/max values for control %d (id %d)\n", - cval->head.id, snd_usb_ctrl_intf(cval->head.mixer->chip), + cval->head.id, mixer_ctrl_intf(cval->head.mixer), cval->control, cval->head.id); return -EINVAL; } @@ -1428,7 +1433,7 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, if (ret) goto error; - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); if (cval->head.mixer->protocol == UAC_VERSION_2) { struct uac2_connectors_ctl_blk uac2_conn; @@ -3219,7 +3224,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, list_for_each_entry(mixer, &chip->mixer_list, list) { snd_iprintf(buffer, "USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n", - chip->usb_id, snd_usb_ctrl_intf(chip), + chip->usb_id, mixer_ctrl_intf(mixer), mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { From 55f7326170d9e83e2d828591938e1101982a679c Mon Sep 17 00:00:00 2001 From: Chris Wulff Date: Sat, 14 Mar 2020 12:54:49 -0400 Subject: [PATCH 295/415] ALSA: usb-audio: Create a registration quirk for Kingston HyperX Amp (0951:16d8) Create a quirk that allows special processing and/or skipping the call to snd_card_register. For HyperX AMP, which uses two interfaces, but only has a capture stream in the second, this allows the capture stream to merge with the first PCM. Signed-off-by: Chris Wulff Link: https://lore.kernel.org/r/20200314165449.4086-3-crwulff@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/card.c | 12 ++++++++---- sound/usb/quirks.c | 14 ++++++++++++++ sound/usb/quirks.h | 3 +++ 3 files changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 827fb0bc8b5619..16bbe2a50fb7b5 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -662,10 +662,14 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } - /* we are allowed to call snd_card_register() many times */ - err = snd_card_register(chip->card); - if (err < 0) - goto __error; + /* we are allowed to call snd_card_register() many times, but first + * check to see if a device needs to skip it or do anything special + */ + if (snd_usb_registration_quirk(chip, ifnum) == 0) { + err = snd_card_register(chip->card); + if (err < 0) + goto __error; + } if (quirk && quirk->shares_media_device) { /* don't want to fail when snd_media_device_create() fails */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3cc745fe24d8dd..d605aff801b848 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1808,3 +1808,17 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, break; } } + +int snd_usb_registration_quirk(struct snd_usb_audio *chip, + int iface) +{ + switch (chip->usb_id) { + case USB_ID(0x0951, 0x16d8): /* Kingston HyperX AMP */ + /* Register only when we reach interface 2 so that streams can + * merge correctly into PCMs from interface 0 + */ + return (iface != 2); + } + /* Register as normal */ + return 0; +} diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index df0355843a4c15..3afc01eabc7e2a 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -51,4 +51,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, struct audioformat *fp, int stream); +int snd_usb_registration_quirk(struct snd_usb_audio *chip, + int iface); + #endif /* __USBAUDIO_QUIRKS_H */ From 28ddd846077a0c7d8382f41cef6840167d236c83 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 16 Mar 2020 12:03:03 +0000 Subject: [PATCH 296/415] ASoC: codecs: wsa881x: request gpio direction before setting Make sure that power down gpio direction is set to ouput before even setting it. Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200316120303.3780-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index b59f1d0e7f843d..25776aa64d7451 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1150,7 +1150,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, wsa881x->sconfig.type = SDW_STREAM_PDM; pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; - gpiod_set_value(wsa881x->sd_n, 1); + gpiod_direction_output(wsa881x->sd_n, 1); wsa881x->regmap = devm_regmap_init_sdw(pdev, &wsa881x_regmap_config); if (IS_ERR(wsa881x->regmap)) { From 3bd7ac41d85527126c31f336a75cb2275a6a8898 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:13 +0530 Subject: [PATCH 297/415] ALSA: compress: add wma codec profiles Some codec profiles were missing for WMA, like WMA9/10 lossless and wma10 pro, so add these profiles Signed-off-by: Vinod Koul Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-2-vkoul@kernel.org Signed-off-by: Mark Brown --- include/uapi/sound/compress_params.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index 9c96fb0e4d903d..a47d9df0fd7b59 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -142,6 +142,9 @@ #define SND_AUDIOPROFILE_WMA8 ((__u32) 0x00000002) #define SND_AUDIOPROFILE_WMA9 ((__u32) 0x00000004) #define SND_AUDIOPROFILE_WMA10 ((__u32) 0x00000008) +#define SND_AUDIOPROFILE_WMA9_PRO ((__u32) 0x00000010) +#define SND_AUDIOPROFILE_WMA9_LOSSLESS ((__u32) 0x00000020) +#define SND_AUDIOPROFILE_WMA10_LOSSLESS ((__u32) 0x00000040) #define SND_AUDIOMODE_WMA_LEVEL1 ((__u32) 0x00000001) #define SND_AUDIOMODE_WMA_LEVEL2 ((__u32) 0x00000002) From 20ff1456d26807aa5441be02f0f5459014664aad Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:14 +0530 Subject: [PATCH 298/415] ALSA: compress: Add wma decoder params Some WMA decoders like WMAv10 etc need some additional encoder option parameters, so add these as WMA decoder params. Signed-off-by: Vinod Koul Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-3-vkoul@kernel.org Signed-off-by: Mark Brown --- include/uapi/sound/compress_params.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index a47d9df0fd7b59..bf6f7155e775e7 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -329,6 +329,13 @@ struct snd_dec_flac { __u16 reserved; } __attribute__((packed, aligned(4))); +struct snd_dec_wma { + __u32 encoder_option; + __u32 adv_encoder_option; + __u32 adv_encoder_option2; + __u32 reserved; +} __attribute__((packed, aligned(4))); + union snd_codec_options { struct snd_enc_wma wma; struct snd_enc_vorbis vorbis; @@ -336,6 +343,7 @@ union snd_codec_options { struct snd_enc_flac flac; struct snd_enc_generic generic; struct snd_dec_flac flac_d; + struct snd_dec_wma wma_d; } __attribute__((packed, aligned(4))); /** struct snd_codec_desc - description of codec capabilities From 8504a72f7ce231989cf2a82f0cdbba5b72f65015 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:15 +0530 Subject: [PATCH 299/415] ASoC: qcom: q6asm: pass codec profile to q6asm_open_write Codec profile is required to be passed for WMA codecs so that we know the codec profile present and tell DSP accordingly, so update this API to pass the codec profile as argument Signed-off-by: Vinod Koul Reviewed-by: Srinivas Kandagatla Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-4-vkoul@kernel.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/qcom/qdsp6/q6asm.h | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 8b48815ff9185e..bc0e3f7cfd8ea3 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -252,7 +252,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM, - prtd->bits_per_sample); + 0, prtd->bits_per_sample); } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM, prtd->bits_per_sample); @@ -654,7 +654,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, prtd->bits_per_sample = 16; if (dir == SND_COMPRESS_PLAYBACK) { ret = q6asm_open_write(prtd->audio_client, params->codec.id, - prtd->bits_per_sample); + params->codec.profile, prtd->bits_per_sample); if (ret < 0) { dev_err(dev, "q6asm_open_write failed\n"); diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 36e0eab13a9831..64eb7b6ba3052d 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -858,7 +858,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt) * Return: Will be an negative value on error or zero on success */ int q6asm_open_write(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample) + u32 codec_profile, uint16_t bits_per_sample) { struct asm_stream_cmd_open_write_v3 *open; struct apr_pkt *pkt; diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 6764f55f7078a8..1cff7f68b95def 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -55,7 +55,7 @@ void q6asm_audio_client_free(struct audio_client *ac); int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, uint32_t lsw_ts, uint32_t flags); int q6asm_open_write(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample); + u32 codec_profile, uint16_t bits_per_sample); int q6asm_open_read(struct audio_client *ac, uint32_t format, uint16_t bits_per_sample); From 97163eadf18bff306b1cf8a8fa81938c5509899c Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:16 +0530 Subject: [PATCH 300/415] ASoC: qcom: q6asm: add support to wma config Qualcomm DSPs expect wma v9 and wma v10 configs to be set for wma decoders, so add the API to program the respective wma config to the DSP Signed-off-by: Vinod Koul Reviewed-by: Srinivas Kandagatla Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-5-vkoul@kernel.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm.c | 123 +++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 17 +++++ 2 files changed, 140 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 64eb7b6ba3052d..4cec95c657ba9d 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -39,6 +39,8 @@ #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5 #define ASM_MEDIA_FMT_MP3 0x00010BE9 #define ASM_MEDIA_FMT_FLAC 0x00010C16 +#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8 +#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7 #define ASM_DATA_CMD_WRITE_V2 0x00010DAB #define ASM_DATA_CMD_READ_V2 0x00010DAC #define ASM_SESSION_CMD_SUSPEND 0x00010DEC @@ -104,6 +106,33 @@ struct asm_flac_fmt_blk_v2 { u16 reserved; } __packed; +struct asm_wmastdv9_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 fmtag; + u16 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u16 blk_align; + u16 bits_per_sample; + u32 channel_mask; + u16 enc_options; + u16 reserved; +} __packed; + +struct asm_wmaprov10_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 fmtag; + u16 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u16 blk_align; + u16 bits_per_sample; + u32 channel_mask; + u16 enc_options; + u16 advanced_enc_options1; + u32 advanced_enc_options2; +} __packed; + struct asm_stream_cmd_set_encdec_param { u32 param_id; u32 param_size; @@ -894,6 +923,24 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, case SND_AUDIOCODEC_FLAC: open->dec_fmt_id = ASM_MEDIA_FMT_FLAC; break; + case SND_AUDIOCODEC_WMA: + switch (codec_profile) { + case SND_AUDIOPROFILE_WMA9: + open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9; + break; + case SND_AUDIOPROFILE_WMA10: + case SND_AUDIOPROFILE_WMA9_PRO: + case SND_AUDIOPROFILE_WMA9_LOSSLESS: + case SND_AUDIOPROFILE_WMA10_LOSSLESS: + open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10; + break; + default: + dev_err(ac->dev, "Invalid codec profile 0x%x\n", + codec_profile); + rc = -EINVAL; + goto err; + } + break; default: dev_err(ac->dev, "Invalid format 0x%x\n", format); rc = -EINVAL; @@ -1075,6 +1122,82 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac, return rc; } EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac); + +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + struct q6asm_wma_cfg *cfg) +{ + struct asm_wmastdv9_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + fmt->fmtag = cfg->fmtag; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->bytes_per_sec = cfg->bytes_per_sec; + fmt->blk_align = cfg->block_align; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->channel_mask = cfg->channel_mask; + fmt->enc_options = cfg->enc_options; + fmt->reserved = 0; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9); + +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + struct q6asm_wma_cfg *cfg) +{ + struct asm_wmaprov10_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + fmt->fmtag = cfg->fmtag; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->bytes_per_sec = cfg->bytes_per_sec; + fmt->blk_align = cfg->block_align; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->channel_mask = cfg->channel_mask; + fmt->enc_options = cfg->enc_options; + fmt->advanced_enc_options1 = cfg->adv_enc_options; + fmt->advanced_enc_options2 = cfg->adv_enc_options2; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10); + /** * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 1cff7f68b95def..5d9fbc75688c86 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -45,6 +45,19 @@ struct q6asm_flac_cfg { u16 md5_sum; }; +struct q6asm_wma_cfg { + u32 fmtag; + u32 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u32 block_align; + u32 bits_per_sample; + u32 channel_mask; + u32 enc_options; + u32 adv_enc_options; + u32 adv_enc_options2; +}; + typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token, void *payload, void *priv); struct audio_client; @@ -69,6 +82,10 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, uint16_t bits_per_sample); int q6asm_stream_media_format_block_flac(struct audio_client *ac, struct q6asm_flac_cfg *cfg); +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + struct q6asm_wma_cfg *cfg); +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + struct q6asm_wma_cfg *cfg); int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, From 40519a1c02303ebdf09e07fe57a52c2d1d188b01 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:17 +0530 Subject: [PATCH 301/415] ASoC: qcom: q6asm-dai: add support to wma decoder Qualcomm DSPs also supports the wma decoder, so add support for wma decoder and convert the snd_codec_params to qdsp format. Signed-off-by: Vinod Koul Reviewed-by: Srinivas Kandagatla Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-6-vkoul@kernel.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++- 1 file changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index bc0e3f7cfd8ea3..fa685fe4a027b0 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -629,10 +629,13 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, int dir = stream->direction; struct q6asm_dai_data *pdata; struct q6asm_flac_cfg flac_cfg; + struct q6asm_wma_cfg wma_cfg; + unsigned int wma_v9 = 0; struct device *dev = c->dev; int ret; union snd_codec_options *codec_options; struct snd_dec_flac *flac; + struct snd_dec_wma *wma; codec_options = &(prtd->codec_param.codec.options); @@ -694,6 +697,67 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, return -EIO; } break; + + case SND_AUDIOCODEC_WMA: + wma = &codec_options->wma_d; + + memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg)); + + wma_cfg.sample_rate = params->codec.sample_rate; + wma_cfg.num_channels = params->codec.ch_in; + wma_cfg.bytes_per_sec = params->codec.bit_rate / 8; + wma_cfg.block_align = params->codec.align; + wma_cfg.bits_per_sample = prtd->bits_per_sample; + wma_cfg.enc_options = wma->encoder_option; + wma_cfg.adv_enc_options = wma->adv_encoder_option; + wma_cfg.adv_enc_options2 = wma->adv_encoder_option2; + + if (wma_cfg.num_channels == 1) + wma_cfg.channel_mask = 4; /* Mono Center */ + else if (wma_cfg.num_channels == 2) + wma_cfg.channel_mask = 3; /* Stereo FL/FR */ + else + return -EINVAL; + + /* check the codec profile */ + switch (params->codec.profile) { + case SND_AUDIOPROFILE_WMA9: + wma_cfg.fmtag = 0x161; + wma_v9 = 1; + break; + + case SND_AUDIOPROFILE_WMA10: + wma_cfg.fmtag = 0x166; + break; + + case SND_AUDIOPROFILE_WMA9_PRO: + wma_cfg.fmtag = 0x162; + break; + + case SND_AUDIOPROFILE_WMA9_LOSSLESS: + wma_cfg.fmtag = 0x163; + break; + + case SND_AUDIOPROFILE_WMA10_LOSSLESS: + wma_cfg.fmtag = 0x167; + break; + + default: + dev_err(dev, "Unknown WMA profile:%x\n", + params->codec.profile); + return -EIO; + } + + if (wma_v9) + ret = q6asm_stream_media_format_block_wma_v9( + prtd->audio_client, &wma_cfg); + else + ret = q6asm_stream_media_format_block_wma_v10( + prtd->audio_client, &wma_cfg); + if (ret < 0) { + dev_err(dev, "WMA9 CMD failed:%d\n", ret); + return -EIO; + } default: break; } @@ -793,9 +857,10 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream, caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; - caps->num_codecs = 2; + caps->num_codecs = 3; caps->codecs[0] = SND_AUDIOCODEC_MP3; caps->codecs[1] = SND_AUDIOCODEC_FLAC; + caps->codecs[2] = SND_AUDIOCODEC_WMA; return 0; } From 0f546d6f0292fb624b4cf0cc70096ddb9ea070e6 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:18 +0530 Subject: [PATCH 302/415] ALSA: compress: add alac & ape decoder params Add ALAC (Apple Lossless Audio Codec) and APE (Monkey's Lossless Audio Codec) defines and parameters required to configure these. Signed-off-by: Vinod Koul Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-7-vkoul@kernel.org Signed-off-by: Mark Brown --- include/uapi/sound/compress_params.h | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index bf6f7155e775e7..79b14389ae4112 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -75,7 +75,9 @@ #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) #define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE +#define SND_AUDIOCODEC_ALAC ((__u32) 0x0000000F) +#define SND_AUDIOCODEC_APE ((__u32) 0x00000010) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_APE /* * Profile and modes are listed with bit masks. This allows for a @@ -336,6 +338,26 @@ struct snd_dec_wma { __u32 reserved; } __attribute__((packed, aligned(4))); +struct snd_dec_alac { + __u32 frame_length; + __u8 compatible_version; + __u8 pb; + __u8 mb; + __u8 kb; + __u32 max_run; + __u32 max_frame_bytes; +} __attribute__((packed, aligned(4))); + +struct snd_dec_ape { + __u16 compatible_version; + __u16 compression_level; + __u32 format_flags; + __u32 blocks_per_frame; + __u32 final_frame_blocks; + __u32 total_frames; + __u32 seek_table_present; +} __attribute__((packed, aligned(4))); + union snd_codec_options { struct snd_enc_wma wma; struct snd_enc_vorbis vorbis; @@ -344,6 +366,8 @@ union snd_codec_options { struct snd_enc_generic generic; struct snd_dec_flac flac_d; struct snd_dec_wma wma_d; + struct snd_dec_alac alac_d; + struct snd_dec_ape ape_d; } __attribute__((packed, aligned(4))); /** struct snd_codec_desc - description of codec capabilities From 7076bf4da0b1b53dd5d96dc930c9899a8ad6f217 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:19 +0530 Subject: [PATCH 303/415] ASoC: qcom: q6asm: add support for alac and ape configs Qualcomm DSPs expect ALAC and APE configs to be send for decoders, so add the API to program the respective config to the DSP. Signed-off-by: Vinod Koul Reviewed-by: Srinivas Kandagatla Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-8-vkoul@kernel.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm.c | 118 +++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 32 ++++++++++ 2 files changed, 150 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 4cec95c657ba9d..0e0e8f7a460ab5 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -48,6 +48,8 @@ #define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4 #define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A #define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D +#define ASM_MEDIA_FMT_ALAC 0x00012f31 +#define ASM_MEDIA_FMT_APE 0x00012f32 #define ASM_LEGACY_STREAM_SESSION 0 @@ -133,6 +135,36 @@ struct asm_wmaprov10_fmt_blk_v2 { u32 advanced_enc_options2; } __packed; +struct asm_alac_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u32 frame_length; + u8 compatible_version; + u8 bit_depth; + u8 pb; + u8 mb; + u8 kb; + u8 num_channels; + u16 max_run; + u32 max_frame_bytes; + u32 avg_bit_rate; + u32 sample_rate; + u32 channel_layout_tag; +} __packed; + +struct asm_ape_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 compatible_version; + u16 compression_level; + u32 format_flags; + u32 blocks_per_frame; + u32 final_frame_blocks; + u32 total_frames; + u16 bits_per_sample; + u16 num_channels; + u32 sample_rate; + u32 seek_table_present; +} __packed; + struct asm_stream_cmd_set_encdec_param { u32 param_id; u32 param_size; @@ -941,6 +973,12 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, goto err; } break; + case SND_AUDIOCODEC_ALAC: + open->dec_fmt_id = ASM_MEDIA_FMT_ALAC; + break; + case SND_AUDIOCODEC_APE: + open->dec_fmt_id = ASM_MEDIA_FMT_APE; + break; default: dev_err(ac->dev, "Invalid format 0x%x\n", format); rc = -EINVAL; @@ -1198,6 +1236,86 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, } EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10); +int q6asm_stream_media_format_block_alac(struct audio_client *ac, + struct q6asm_alac_cfg *cfg) +{ + struct asm_alac_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + + fmt->frame_length = cfg->frame_length; + fmt->compatible_version = cfg->compatible_version; + fmt->bit_depth = cfg->bit_depth; + fmt->num_channels = cfg->num_channels; + fmt->max_run = cfg->max_run; + fmt->max_frame_bytes = cfg->max_frame_bytes; + fmt->avg_bit_rate = cfg->avg_bit_rate; + fmt->sample_rate = cfg->sample_rate; + fmt->channel_layout_tag = cfg->channel_layout_tag; + fmt->pb = cfg->pb; + fmt->mb = cfg->mb; + fmt->kb = cfg->kb; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac); + +int q6asm_stream_media_format_block_ape(struct audio_client *ac, + struct q6asm_ape_cfg *cfg) +{ + struct asm_ape_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + + fmt->compatible_version = cfg->compatible_version; + fmt->compression_level = cfg->compression_level; + fmt->format_flags = cfg->format_flags; + fmt->blocks_per_frame = cfg->blocks_per_frame; + fmt->final_frame_blocks = cfg->final_frame_blocks; + fmt->total_frames = cfg->total_frames; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->seek_table_present = cfg->seek_table_present; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape); + /** * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 5d9fbc75688c86..38a207d6cd9587 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -58,6 +58,34 @@ struct q6asm_wma_cfg { u32 adv_enc_options2; }; +struct q6asm_alac_cfg { + u32 frame_length; + u8 compatible_version; + u8 bit_depth; + u8 pb; + u8 mb; + u8 kb; + u8 num_channels; + u16 max_run; + u32 max_frame_bytes; + u32 avg_bit_rate; + u32 sample_rate; + u32 channel_layout_tag; +}; + +struct q6asm_ape_cfg { + u16 compatible_version; + u16 compression_level; + u32 format_flags; + u32 blocks_per_frame; + u32 final_frame_blocks; + u32 total_frames; + u16 bits_per_sample; + u16 num_channels; + u32 sample_rate; + u32 seek_table_present; +}; + typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token, void *payload, void *priv); struct audio_client; @@ -86,6 +114,10 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, struct q6asm_wma_cfg *cfg); int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, struct q6asm_wma_cfg *cfg); +int q6asm_stream_media_format_block_alac(struct audio_client *ac, + struct q6asm_alac_cfg *cfg); +int q6asm_stream_media_format_block_ape(struct audio_client *ac, + struct q6asm_ape_cfg *cfg); int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, From 4c3189380c6748a3e9fc6ab8aeb4bde3dd2032ed Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:20 +0530 Subject: [PATCH 304/415] ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders Qualcomm DSPs also supports the ALAC and APE decoders, so add support for these and convert the snd_codec_params to qdsp format. Signed-off-by: Vinod Koul Reviewed-by: Srinivas Kandagatla Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-9-vkoul@kernel.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 70 +++++++++++++++++++++++++++++++- 1 file changed, 69 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index fa685fe4a027b0..8b5d86be9acec7 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -41,6 +41,9 @@ #define Q6ASM_DAI_TX 1 #define Q6ASM_DAI_RX 2 +#define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) +#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) + enum stream_state { Q6ASM_STREAM_IDLE = 0, Q6ASM_STREAM_STOPPED, @@ -630,12 +633,16 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, struct q6asm_dai_data *pdata; struct q6asm_flac_cfg flac_cfg; struct q6asm_wma_cfg wma_cfg; + struct q6asm_alac_cfg alac_cfg; + struct q6asm_ape_cfg ape_cfg; unsigned int wma_v9 = 0; struct device *dev = c->dev; int ret; union snd_codec_options *codec_options; struct snd_dec_flac *flac; struct snd_dec_wma *wma; + struct snd_dec_alac *alac; + struct snd_dec_ape *ape; codec_options = &(prtd->codec_param.codec.options); @@ -758,6 +765,65 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, dev_err(dev, "WMA9 CMD failed:%d\n", ret); return -EIO; } + break; + + case SND_AUDIOCODEC_ALAC: + memset(&alac_cfg, 0x0, sizeof(alac_cfg)); + alac = &codec_options->alac_d; + + alac_cfg.sample_rate = params->codec.sample_rate; + alac_cfg.avg_bit_rate = params->codec.bit_rate; + alac_cfg.bit_depth = prtd->bits_per_sample; + alac_cfg.num_channels = params->codec.ch_in; + + alac_cfg.frame_length = alac->frame_length; + alac_cfg.pb = alac->pb; + alac_cfg.mb = alac->mb; + alac_cfg.kb = alac->kb; + alac_cfg.max_run = alac->max_run; + alac_cfg.compatible_version = alac->compatible_version; + alac_cfg.max_frame_bytes = alac->max_frame_bytes; + + switch (params->codec.ch_in) { + case 1: + alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO; + break; + case 2: + alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO; + break; + } + ret = q6asm_stream_media_format_block_alac(prtd->audio_client, + &alac_cfg); + if (ret < 0) { + dev_err(dev, "ALAC CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + + case SND_AUDIOCODEC_APE: + memset(&ape_cfg, 0x0, sizeof(ape_cfg)); + ape = &codec_options->ape_d; + + ape_cfg.sample_rate = params->codec.sample_rate; + ape_cfg.num_channels = params->codec.ch_in; + ape_cfg.bits_per_sample = prtd->bits_per_sample; + + ape_cfg.compatible_version = ape->compatible_version; + ape_cfg.compression_level = ape->compression_level; + ape_cfg.format_flags = ape->format_flags; + ape_cfg.blocks_per_frame = ape->blocks_per_frame; + ape_cfg.final_frame_blocks = ape->final_frame_blocks; + ape_cfg.total_frames = ape->total_frames; + ape_cfg.seek_table_present = ape->seek_table_present; + + ret = q6asm_stream_media_format_block_ape(prtd->audio_client, + &ape_cfg); + if (ret < 0) { + dev_err(dev, "APE CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + default: break; } @@ -857,10 +923,12 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream, caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; - caps->num_codecs = 3; + caps->num_codecs = 5; caps->codecs[0] = SND_AUDIOCODEC_MP3; caps->codecs[1] = SND_AUDIOCODEC_FLAC; caps->codecs[2] = SND_AUDIOCODEC_WMA; + caps->codecs[3] = SND_AUDIOCODEC_ALAC; + caps->codecs[4] = SND_AUDIOCODEC_APE; return 0; } From 54ce83a3080c3d06b14faa7533a9adb4e158dcea Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 16 Mar 2020 11:22:21 +0530 Subject: [PATCH 305/415] ALSA: compress: bump the version We have added support for bunch of new decoders and parameters for decoders. To help users find the support bump the version up to 0,2,0. Signed-off-by: Vinod Koul Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20200316055221.1944464-10-vkoul@kernel.org Signed-off-by: Mark Brown --- include/uapi/sound/compress_offload.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 56d95673ce0f02..7184265c0b0d4e 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -31,7 +31,7 @@ #include -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 2, 0) /** * struct snd_compressed_buffer - compressed buffer * @fragment_size: size of buffer fragment in bytes From 308811a327c38364fff7752d3009160632f5dd5e Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 16 Mar 2020 15:11:10 +0000 Subject: [PATCH 306/415] ASoC: soc-dai: return proper error for get_sdw_stream() snd_soc_dai_get_sdw_stream() returns null if dai does not support this callback, this is no very useful for the caller to differentiate if this is an error or unsupported call for the dai. return -ENOTSUPP in cases where this callback is not supported. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200316151110.2580-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 7f70db149b812a..78bac995db151f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -433,7 +433,7 @@ static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai, * This routine only retrieves that was previously configured * with snd_soc_dai_get_sdw_stream() * - * Returns pointer to stream or NULL; + * Returns pointer to stream or -ENOTSUPP if callback is not supported; */ static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai, int direction) @@ -441,7 +441,7 @@ static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai, if (dai->driver->ops->get_sdw_stream) return dai->driver->ops->get_sdw_stream(dai, direction); else - return NULL; + return ERR_PTR(-ENOTSUPP); } #endif From 557270e8dc79f66a1db8907eb3079ade60241fe7 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 17 Mar 2020 15:33:08 +0800 Subject: [PATCH 307/415] ASoC: rt5682: fix the random recording noise of headset The cycle time of FIFO clock should increase 2 times to avoid the random recording noise issue. This setting could apply to all known situations in i2s mode. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20200317073308.11572-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 2 ++ sound/soc/codecs/rt5682.h | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 063dd338539d58..3e0b5c43ece8ce 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2651,6 +2651,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_FIFO_CLK_DIV_MASK, RT5682_FIFO_CLK_DIV_2); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 18faaa2a49a0ce..fc99e748428341 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -651,6 +651,8 @@ #define RT5682_DMIC_1_EN_SFT 15 #define RT5682_DMIC_1_DIS (0x0 << 15) #define RT5682_DMIC_1_EN (0x1 << 15) +#define RT5682_FIFO_CLK_DIV_MASK (0x7 << 12) +#define RT5682_FIFO_CLK_DIV_2 (0x1 << 12) #define RT5682_DMIC_1_DP_MASK (0x3 << 4) #define RT5682_DMIC_1_DP_SFT 4 #define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4) From a168dae5ea14283e8992d5282237bb0d6a3e1c06 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 18 Mar 2020 15:41:23 +0100 Subject: [PATCH 308/415] ASoC: stm32: spdifrx: fix regmap status check Release resources when exiting on error. Fixes: 1a5c0b28fc56 ("ASoC: stm32: spdifrx: manage identification registers") Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200318144125.9163-2-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 3769d9ce5dbef9..e6e75897cce838 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -1009,6 +1009,8 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) if (idr == SPDIFRX_IPIDR_NUMBER) { ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver); + if (ret) + goto error; dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n", FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver), From 794df9448edb55978e50372f083aeedade1b2844 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 18 Mar 2020 15:41:24 +0100 Subject: [PATCH 309/415] ASoC: stm32: spdifrx: manage rebind issue The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. Corrections: - Call snd_dmaengine_pcm_register() before snd_soc_register_component(). - Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SPDFIRX driver. Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200318144125.9163-3-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 62 ++++++++++++++++++----------------- 1 file changed, 32 insertions(+), 30 deletions(-) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 49766afdae6163..ae7a0f46a6fb13 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -944,6 +944,22 @@ static int stm32_spdifrx_parse_of(struct platform_device *pdev, return 0; } +static int stm32_spdifrx_remove(struct platform_device *pdev) +{ + struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev); + + if (spdifrx->ctrl_chan) + dma_release_channel(spdifrx->ctrl_chan); + + if (spdifrx->dmab) + snd_dma_free_pages(spdifrx->dmab); + + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + static int stm32_spdifrx_probe(struct platform_device *pdev) { struct stm32_spdifrx_data *spdifrx; @@ -995,25 +1011,27 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) udelay(2); reset_control_deassert(rst); - ret = devm_snd_soc_register_component(&pdev->dev, - &stm32_spdifrx_component, - stm32_spdifrx_dai, - ARRAY_SIZE(stm32_spdifrx_dai)); - if (ret) - return ret; - - ret = stm32_spdifrx_dma_ctrl_register(&pdev->dev, spdifrx); - if (ret) - goto error; - pcm_config = &stm32_spdifrx_pcm_config; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0); + ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0); if (ret) { if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); - goto error; + return ret; } + ret = snd_soc_register_component(&pdev->dev, + &stm32_spdifrx_component, + stm32_spdifrx_dai, + ARRAY_SIZE(stm32_spdifrx_dai)); + if (ret) { + snd_dmaengine_pcm_unregister(&pdev->dev); + return ret; + } + + ret = stm32_spdifrx_dma_ctrl_register(&pdev->dev, spdifrx); + if (ret) + goto error; + ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_IDR, &idr); if (ret) goto error; @@ -1029,27 +1047,11 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) return ret; error: - if (!IS_ERR(spdifrx->ctrl_chan)) - dma_release_channel(spdifrx->ctrl_chan); - if (spdifrx->dmab) - snd_dma_free_pages(spdifrx->dmab); + stm32_spdifrx_remove(pdev); return ret; } -static int stm32_spdifrx_remove(struct platform_device *pdev) -{ - struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev); - - if (spdifrx->ctrl_chan) - dma_release_channel(spdifrx->ctrl_chan); - - if (spdifrx->dmab) - snd_dma_free_pages(spdifrx->dmab); - - return 0; -} - MODULE_DEVICE_TABLE(of, stm32_spdifrx_ids); #ifdef CONFIG_PM_SLEEP From caff4ce8cc582a97b17d10b7c7f5fe8500323135 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 18 Mar 2020 15:41:25 +0100 Subject: [PATCH 310/415] ASoC: stm32: i2s: manage rebind issue The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. Corrections: - Call snd_dmaengine_pcm_register() before snd_soc_register_component(). - Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from I2S driver. Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20200318144125.9163-4-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 40 ++++++++++++++++++++++++++++----------- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 2478405727c378..7c4d63c33f15b3 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -888,6 +888,14 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, return 0; } +static int stm32_i2s_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + static int stm32_i2s_probe(struct platform_device *pdev) { struct stm32_i2s_data *i2s; @@ -921,47 +929,56 @@ static int stm32_i2s_probe(struct platform_device *pdev) return PTR_ERR(i2s->regmap); } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_i2s_component, - i2s->dai_drv, 1); - if (ret) - return ret; - - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, - &stm32_i2s_pcm_config, 0); + ret = snd_dmaengine_pcm_register(&pdev->dev, &stm32_i2s_pcm_config, 0); if (ret) { if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); return ret; } + ret = snd_soc_register_component(&pdev->dev, &stm32_i2s_component, + i2s->dai_drv, 1); + if (ret) { + snd_dmaengine_pcm_unregister(&pdev->dev); + return ret; + } + /* Set SPI/I2S in i2s mode */ ret = regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, I2S_CGFR_I2SMOD, I2S_CGFR_I2SMOD); if (ret) - return ret; + goto error; ret = regmap_read(i2s->regmap, STM32_I2S_IPIDR_REG, &val); if (ret) - return ret; + goto error; if (val == I2S_IPIDR_NUMBER) { ret = regmap_read(i2s->regmap, STM32_I2S_HWCFGR_REG, &val); if (ret) - return ret; + goto error; if (!FIELD_GET(I2S_HWCFGR_I2S_SUPPORT_MASK, val)) { dev_err(&pdev->dev, "Device does not support i2s mode\n"); - return -EPERM; + ret = -EPERM; + goto error; } ret = regmap_read(i2s->regmap, STM32_I2S_VERR_REG, &val); + if (ret) + goto error; dev_dbg(&pdev->dev, "I2S version: %lu.%lu registered\n", FIELD_GET(I2S_VERR_MAJ_MASK, val), FIELD_GET(I2S_VERR_MIN_MASK, val)); } + return ret; + +error: + stm32_i2s_remove(pdev); + return ret; } @@ -998,6 +1015,7 @@ static struct platform_driver stm32_i2s_driver = { .pm = &stm32_i2s_pm_ops, }, .probe = stm32_i2s_probe, + .remove = stm32_i2s_remove, }; module_platform_driver(stm32_i2s_driver); From 16252a8f3af77f69c2193fdc7b2f595b30845a44 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 17 Mar 2020 15:12:33 +0000 Subject: [PATCH 311/415] ASoC: codecs: wsa881x: remove soundwire stream handling There could be multiple instances of this codec on any platform, so handling stream directly in this codec driver can lead to multiple calls to prepare/enable/disable on the same SoundWire stream. Move this stream handling to machine driver to fix this issue. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200317151233.8763-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 44 +------------------------------------- 1 file changed, 1 insertion(+), 43 deletions(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index b59f1d0e7f843d..3d525297eac933 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -676,7 +676,6 @@ struct wsa881x_priv { int active_ports; bool port_prepared[WSA881X_MAX_SWR_PORTS]; bool port_enable[WSA881X_MAX_SWR_PORTS]; - bool stream_prepared; }; static void wsa881x_init(struct wsa881x_priv *wsa881x) @@ -954,41 +953,6 @@ static const struct snd_soc_dapm_widget wsa881x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPKR"), }; -static int wsa881x_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev); - int ret; - - if (wsa881x->stream_prepared) { - sdw_disable_stream(wsa881x->sruntime); - sdw_deprepare_stream(wsa881x->sruntime); - wsa881x->stream_prepared = false; - } - - - ret = sdw_prepare_stream(wsa881x->sruntime); - if (ret) - return ret; - - /** - * NOTE: there is a strict hw requirement about the ordering of port - * enables and actual PA enable. PA enable should only happen after - * soundwire ports are enabled if not DC on the line is accumulated - * resulting in Click/Pop Noise - * PA enable/mute are handled as part of DAPM and digital mute. - */ - - ret = sdw_enable_stream(wsa881x->sruntime); - if (ret) { - sdw_deprepare_stream(wsa881x->sruntime); - return ret; - } - wsa881x->stream_prepared = true; - - return ret; -} - static int wsa881x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1016,12 +980,7 @@ static int wsa881x_hw_free(struct snd_pcm_substream *substream, { struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev); - if (wsa881x->stream_prepared) { - sdw_disable_stream(wsa881x->sruntime); - sdw_deprepare_stream(wsa881x->sruntime); - sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime); - wsa881x->stream_prepared = false; - } + sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime); return 0; } @@ -1052,7 +1011,6 @@ static int wsa881x_digital_mute(struct snd_soc_dai *dai, int mute, int stream) static struct snd_soc_dai_ops wsa881x_dai_ops = { .hw_params = wsa881x_hw_params, - .prepare = wsa881x_prepare, .hw_free = wsa881x_hw_free, .mute_stream = wsa881x_digital_mute, .set_sdw_stream = wsa881x_set_sdw_stream, From 1b93a88431470ea0b943157999084d9c7e6e3bd3 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 17 Mar 2020 15:12:32 +0000 Subject: [PATCH 312/415] ASoC: qcom: sdm845: handle soundwire stream In existing setup WSA881x codec handles soundwire stream, however DB845c and other machines based on SDM845c have 2 instances for WSA881x codec. This will force soundwire stream to be prepared/enabled twice or multiple times. Handling SoundWire Stream in machine driver would fix this issue. Signed-off-by: Srinivas Kandagatla Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200317151233.8763-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/sdm845.c | 67 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 6530d2462a9e0b..f51b28d1b94d87 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,7 +99,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && CROS_EC && I2C + depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 3ac02204a7069d..67a55edf755fe2 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -11,6 +11,7 @@ #include #include #include +#include #include #include "common.h" #include "qdsp6/q6afe.h" @@ -31,10 +32,12 @@ struct sdm845_snd_data { struct snd_soc_jack jack; bool jack_setup; + bool stream_prepared[SLIM_MAX_RX_PORTS]; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; uint32_t sec_mi2s_clk_count; uint32_t quat_tdm_clk_count; + struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS]; }; static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28}; @@ -45,11 +48,18 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; + struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card); u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; + struct sdw_stream_runtime *sruntime; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0, i; for_each_rtd_codec_dais(rtd, i, codec_dai) { + sruntime = snd_soc_dai_get_sdw_stream(codec_dai, + substream->stream); + if (sruntime != ERR_PTR(-ENOTSUPP)) + pdata->sruntime[cpu_dai->id] = sruntime; + ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); @@ -425,8 +435,65 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) } } +static int sdm845_snd_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; + int ret; + + if (!sruntime) + return 0; + + if (data->stream_prepared[cpu_dai->id]) { + sdw_disable_stream(sruntime); + sdw_deprepare_stream(sruntime); + data->stream_prepared[cpu_dai->id] = false; + } + + ret = sdw_prepare_stream(sruntime); + if (ret) + return ret; + + /** + * NOTE: there is a strict hw requirement about the ordering of port + * enables and actual WSA881x PA enable. PA enable should only happen + * after soundwire ports are enabled if not DC on the line is + * accumulated resulting in Click/Pop Noise + * PA enable/mute are handled as part of codec DAPM and digital mute. + */ + + ret = sdw_enable_stream(sruntime); + if (ret) { + sdw_deprepare_stream(sruntime); + return ret; + } + data->stream_prepared[cpu_dai->id] = true; + + return ret; +} + +static int sdm845_snd_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; + + if (sruntime && data->stream_prepared[cpu_dai->id]) { + sdw_disable_stream(sruntime); + sdw_deprepare_stream(sruntime); + data->stream_prepared[cpu_dai->id] = false; + } + + return 0; +} + static const struct snd_soc_ops sdm845_be_ops = { .hw_params = sdm845_snd_hw_params, + .hw_free = sdm845_snd_hw_free, + .prepare = sdm845_snd_prepare, .startup = sdm845_snd_startup, .shutdown = sdm845_snd_shutdown, }; From 243de01deb545e9c288459458e28e5ff0656ca1f Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 17 Mar 2020 15:33:21 +0800 Subject: [PATCH 313/415] ASoC: rt5682: remove noisy debug messages Some debug messages are too noisy. This patch removes it. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20200317073321.12660-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7ca02a5e52e9cc..513429478d27e4 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1197,11 +1197,11 @@ static int rt5682_div_sel(struct rt5682_priv *rt5682, } for (i = 0; i < size - 1; i++) { - pr_info("div[%d]=%d\n", i, div[i]); + dev_dbg(rt5682->component->dev, "div[%d]=%d\n", i, div[i]); if (target * div[i] == rt5682->sysclk) return i; if (target * div[i + 1] > rt5682->sysclk) { - pr_err("can't find div for sysclk %d\n", + dev_dbg(rt5682->component->dev, "can't find div for sysclk %d\n", rt5682->sysclk); return i; } From e9097e47e349b747dee50f935216de0ffb662962 Mon Sep 17 00:00:00 2001 From: Geoffrey Allott Date: Thu, 19 Mar 2020 14:00:48 +0000 Subject: [PATCH 314/415] ALSA: hda/ca0132 - Add Recon3Di quirk to handle integrated sound on EVGA X99 Classified motherboard I have a system which has an EVGA X99 Classified motherboard. The pin assignments for the HD Audio controller are not correct under Linux. Windows 10 works fine and informs me that it's using the Recon3Di driver, and on Linux, `cat /sys/class/sound/card0/device/subsystem_{vendor,device}` yields 0x3842 0x1038 This patch adds a corresponding entry to the quirk list. Signed-off-by: Geoffrey Allott Cc: Link: https://lore.kernel.org/r/a6cd56b678c00ce2db3685e4278919f2584f8244.camel@allott.email Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index ded8bc07d755ac..10223e080d59b6 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1180,6 +1180,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), + SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI), SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5), {} From a9107de4b03604ce0d279315c91b31b8065ee4ea Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 11 Mar 2020 11:35:44 +0000 Subject: [PATCH 315/415] soundwire: stream: Add read_only_wordlength flag to port properties According to SoundWire Specification Version 1.2. "A Data Port number X (in the range 0-14) which supports only one value of WordLength may implement the WordLength field in the DPX_BlockCtrl1 Register as Read-Only, returning the fixed value of WordLength in response to reads." As WSA881x interfaces in PDM mode making the only field "WordLength" in DPX_BlockCtrl1" fixed and read-only. Behaviour of writing to this register on WSA881x soundwire slave with Qualcomm Soundwire Controller is throwing up an error. Not sure how other controllers deal with writing to readonly registers, but this patch provides a way to avoid writes to DPN_BlockCtrl1 register by providing a read_only_wordlength flag in struct sdw_dpn_prop Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20200311113545.23773-2-srinivas.kandagatla@linaro.org Signed-off-by: Vinod Koul --- drivers/soundwire/stream.c | 16 +++++++++------- include/linux/soundwire/sdw.h | 2 ++ 2 files changed, 11 insertions(+), 7 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 178ae92b8cc1cc..7fb89a94d9c0dc 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -167,13 +167,15 @@ static int sdw_program_slave_port_params(struct sdw_bus *bus, return ret; } - /* Program DPN_BlockCtrl1 register */ - ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); - if (ret < 0) { - dev_err(&s_rt->slave->dev, - "DPN_BlockCtrl1 register write failed for port %d\n", - t_params->port_num); - return ret; + if (!dpn_prop->read_only_wordlength) { + /* Program DPN_BlockCtrl1 register */ + ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); + if (ret < 0) { + dev_err(&s_rt->slave->dev, + "DPN_BlockCtrl1 register write failed for port %d\n", + t_params->port_num); + return ret; + } } /* Program DPN_SampleCtrl1 register */ diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index b451bb62233584..2dfe14ed3bb08b 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -284,6 +284,7 @@ struct sdw_dpn_audio_mode { * @max_async_buffer: Number of samples that this port can buffer in * asynchronous modes * @block_pack_mode: Type of block port mode supported + * @read_only_wordlength: Read Only wordlength field in DPN_BlockCtrl1 register * @port_encoding: Payload Channel Sample encoding schemes supported * @audio_modes: Audio modes supported */ @@ -307,6 +308,7 @@ struct sdw_dpn_prop { u32 modes; u32 max_async_buffer; bool block_pack_mode; + bool read_only_wordlength; u32 port_encoding; struct sdw_dpn_audio_mode *audio_modes; }; From 22a2fc81658b3eebcfcc110de97bcbd32f5ee301 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Mar 2020 15:36:58 +0900 Subject: [PATCH 316/415] ASoC: soc-core: Merge CPU/Codec DAIs ALSA SoC is currently categorizing CPU/Codec DAIs, and it works well. But modern devices require more complex connections, for example Codec to Codec, etc, and future devices will enable to more complex connections. Because of these background, CPU/Codec DAIs categorizing is no longer good much to modern device. Currently, rtd has both CPU/Codec DAIs pointer. rtd->cpu_dais = [][][][][][][][][] rtd->codec_dais = [][][][][][][][][] This patch merges these into DAIs pointer. rtd->dais = [][][][][][][][][][][][][][][][][][] ^cpu_dais ^codec_dais |--- num_cpus ---|--- num_codecs --| Then, we can merge for_each_rtd_cpu/codec_dais() from this patch. - for_each_rtd_cpu_dais() { - ... - } - for_each_rtd_codec_dais() { - ... - } + for_each_rtd_dais() { + ... + } Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87wo7kolfa.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 7 ++++++- sound/soc/soc-core.c | 18 +++++++++--------- 2 files changed, 15 insertions(+), 10 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d97c4aa779a269..539211bd0f9493 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1145,6 +1145,7 @@ struct snd_soc_pcm_runtime { struct snd_compr *compr; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; + struct snd_soc_dai **dais; struct snd_soc_dai **codec_dais; unsigned int num_codecs; @@ -1184,7 +1185,11 @@ struct snd_soc_pcm_runtime { (i)++) #define for_each_rtd_codec_dais_rollback(rtd, i, dai) \ for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) - +#define for_each_rtd_dais(rtd, i, dai) \ + for ((i) = 0; \ + ((i) < (rtd)->num_cpus + (rtd)->num_codecs) && \ + ((dai) = (rtd)->dais[i]); \ + (i)++) void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e7e70b47590a5e..0fd582c19c0388 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -475,22 +475,22 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); /* - * for rtd->codec_dais + * for rtd->dais */ - rtd->codec_dais = devm_kcalloc(dev, dai_link->num_codecs, + rtd->dais = devm_kcalloc(dev, dai_link->num_cpus + dai_link->num_codecs, sizeof(struct snd_soc_dai *), GFP_KERNEL); - if (!rtd->codec_dais) + if (!rtd->dais) goto free_rtd; /* - * for rtd->cpu_dais + * dais = [][][][][][][][][][][][][][][][][][] + * ^cpu_dais ^codec_dais + * |--- num_cpus ---|--- num_codecs --| */ - rtd->cpu_dais = devm_kcalloc(dev, dai_link->num_cpus, - sizeof(struct snd_soc_dai *), - GFP_KERNEL); - if (!rtd->cpu_dais) - goto free_rtd; + rtd->cpu_dais = &rtd->dais[0]; + rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; + /* * rtd remaining settings */ From 3af6ff5035ad95ee02728aff271a9b543b912f15 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Mar 2020 15:37:03 +0900 Subject: [PATCH 317/415] ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais() Now we can use for_each_rtd_dais(). Let's use it instead of for_each_rtd_cpu/codec_dais(). Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87v9n4olf4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 25 +++++++------------------ 1 file changed, 7 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0fd582c19c0388..246d59966795b4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1323,26 +1323,22 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order) static void soc_remove_link_dais(struct snd_soc_card *card) { int i; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; + struct snd_soc_dai *dai; struct snd_soc_pcm_runtime *rtd; int order; for_each_comp_order(order) { for_each_card_rtds(card, rtd) { - /* remove the CODEC DAI */ - for_each_rtd_codec_dais(rtd, i, codec_dai) - soc_remove_dai(codec_dai, order); - - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - soc_remove_dai(cpu_dai, order); + /* remove DAIs */ + for_each_rtd_dais(rtd, i, dai) + soc_remove_dai(dai, order); } } } static int soc_probe_link_dais(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai, *cpu_dai; + struct snd_soc_dai *dai; struct snd_soc_pcm_runtime *rtd; int i, order, ret; @@ -1354,15 +1350,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card) card->name, rtd->num, order); /* probe the CPU DAI */ - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - ret = soc_probe_dai(cpu_dai, order); - if (ret) - return ret; - } - - /* probe the CODEC DAI */ - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = soc_probe_dai(codec_dai, order); + for_each_rtd_dais(rtd, i, dai) { + ret = soc_probe_dai(dai, order); if (ret) return ret; } From e3c3cf71013fd959abf455abc20386051d37c529 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Mar 2020 15:37:09 +0900 Subject: [PATCH 318/415] ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais() Now we can use for_each_rtd_dais(). Let's use it instead of for_each_rtd_cpu/codec_dais(). Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87tv2ooley.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d5eb52fe115b4f..04da7928c873d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4433,14 +4433,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; + struct snd_soc_dai *dai; int i; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - soc_dapm_dai_stream_event(cpu_dai, stream, event); - for_each_rtd_codec_dais(rtd, i, codec_dai) - soc_dapm_dai_stream_event(codec_dai, stream, event); + for_each_rtd_dais(rtd, i, dai) + soc_dapm_dai_stream_event(dai, stream, event); dapm_power_widgets(rtd->card, event); } From c840f7698d26b078695dbc863ccb6a14ca765f98 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Mar 2020 15:37:14 +0900 Subject: [PATCH 319/415] ASoC: soc-pcm: Merge for_each_rtd_cpu/codec_dais() Now we can use for_each_rtd_dais(). Let's use it instead of for_each_rtd_cpu/codec_dais(). Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87sgi8olet.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 311 +++++++++++--------------------------------- 1 file changed, 75 insertions(+), 236 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2b915f41e955b2..e256d438ee6896 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -259,25 +259,15 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd, static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i; lockdep_assert_held(&rtd->card->pcm_mutex); - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - cpu_dai->stream_active[stream] += action; - - for_each_rtd_codec_dais(rtd, i, codec_dai) - codec_dai->stream_active[stream] += action; - - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - cpu_dai->active += action; - cpu_dai->component->active += action; - } - for_each_rtd_codec_dais(rtd, i, codec_dai) { - codec_dai->active += action; - codec_dai->component->active += action; + for_each_rtd_dais(rtd, i, dai) { + dai->stream_active[stream] += action; + dai->active += action; + dai->component->active += action; } } @@ -446,8 +436,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai; struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; unsigned int rate, channels, sample_bits, symmetry, i; rate = params_rate(params); @@ -457,11 +447,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, /* reject unmatched parameters when applying symmetry */ symmetry = rtd->dai_link->symmetric_rates; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - symmetry |= cpu_dai->driver->symmetric_rates; - - for_each_rtd_codec_dais(rtd, i, codec_dai) - symmetry |= codec_dai->driver->symmetric_rates; + for_each_rtd_cpu_dais(rtd, i, dai) + symmetry |= dai->driver->symmetric_rates; if (symmetry) { for_each_rtd_cpu_dais(rtd, i, cpu_dai) { @@ -475,11 +462,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = rtd->dai_link->symmetric_channels; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - symmetry |= cpu_dai->driver->symmetric_channels; - - for_each_rtd_codec_dais(rtd, i, codec_dai) - symmetry |= codec_dai->driver->symmetric_channels; + for_each_rtd_dais(rtd, i, dai) + symmetry |= dai->driver->symmetric_channels; if (symmetry) { for_each_rtd_cpu_dais(rtd, i, cpu_dai) { @@ -494,11 +478,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = rtd->dai_link->symmetric_samplebits; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - symmetry |= cpu_dai->driver->symmetric_samplebits; - - for_each_rtd_codec_dais(rtd, i, codec_dai) - symmetry |= codec_dai->driver->symmetric_samplebits; + for_each_rtd_dais(rtd, i, dai) + symmetry |= dai->driver->symmetric_samplebits; if (symmetry) { for_each_rtd_cpu_dais(rtd, i, cpu_dai) { @@ -518,25 +499,18 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *link = rtd->dai_link; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; + struct snd_soc_dai *dai; unsigned int symmetry, i; symmetry = link->symmetric_rates || link->symmetric_channels || link->symmetric_samplebits; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - symmetry = symmetry || - cpu_dai->driver->symmetric_rates || - cpu_dai->driver->symmetric_channels || - cpu_dai->driver->symmetric_samplebits; - - for_each_rtd_codec_dais(rtd, i, codec_dai) + for_each_rtd_dais(rtd, i, dai) symmetry = symmetry || - codec_dai->driver->symmetric_rates || - codec_dai->driver->symmetric_channels || - codec_dai->driver->symmetric_samplebits; + dai->driver->symmetric_rates || + dai->driver->symmetric_channels || + dai->driver->symmetric_samplebits; return symmetry; } @@ -774,19 +748,15 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); snd_soc_runtime_deactivate(rtd, substream->stream); - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - snd_soc_dai_shutdown(cpu_dai, substream); - - for_each_rtd_codec_dais(rtd, i, codec_dai) - snd_soc_dai_shutdown(codec_dai, substream); + for_each_rtd_dais(rtd, i, dai) + snd_soc_dai_shutdown(dai, substream); soc_rtd_shutdown(rtd, substream); @@ -818,8 +788,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; const char *codec_dai_name = "multicodec"; const char *cpu_dai_name = "multicpu"; int i, ret = 0; @@ -844,28 +813,19 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } /* startup the audio subsystem */ - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - ret = snd_soc_dai_startup(cpu_dai, substream); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_startup(dai, substream); if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", - cpu_dai->name, ret); - goto cpu_dai_err; - } - } - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_startup(codec_dai, substream); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: can't open codec %s: %d\n", - codec_dai->name, ret); + dev_err(dai->dev, + "ASoC: can't open DAI %s: %d\n", + dai->name, ret); goto config_err; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_dai->tx_mask = 0; + dai->tx_mask = 0; else - codec_dai->rx_mask = 0; + dai->rx_mask = 0; } /* Dynamic PCM DAI links compat checks use dynamic capabilities */ @@ -905,17 +865,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) soc_pcm_apply_msb(substream); /* Symmetry only applies if we've already got an active stream. */ - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - if (cpu_dai->active) { - ret = soc_pcm_apply_symmetry(substream, cpu_dai); - if (ret != 0) - goto config_err; - } - } - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - if (codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream, codec_dai); + for_each_rtd_dais(rtd, i, dai) { + if (dai->active) { + ret = soc_pcm_apply_symmetry(substream, dai); if (ret != 0) goto config_err; } @@ -937,11 +889,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - for_each_rtd_codec_dais(rtd, i, codec_dai) - snd_soc_dai_shutdown(codec_dai, substream); -cpu_dai_err: - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - snd_soc_dai_shutdown(cpu_dai, substream); + for_each_rtd_dais(rtd, i, dai) + snd_soc_dai_shutdown(dai, substream); soc_rtd_shutdown(rtd, substream); rtd_startup_err: @@ -980,8 +929,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -1002,21 +950,11 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_prepare(codec_dai, substream); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: codec DAI prepare error: %d\n", - ret); - goto out; - } - } - - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - ret = snd_soc_dai_prepare(cpu_dai, substream); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_prepare(dai, substream); if (ret < 0) { - dev_err(cpu_dai->dev, - "ASoC: cpu DAI prepare error: %d\n", ret); + dev_err(dai->dev, + "ASoC: DAI prepare error: %d\n", ret); goto out; } } @@ -1031,11 +969,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - for_each_rtd_codec_dais(rtd, i, codec_dai) - snd_soc_dai_digital_mute(codec_dai, 0, - substream->stream); - for_each_rtd_cpu_dais(rtd, i, cpu_dai) - snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); + for_each_rtd_dais(rtd, i, dai) + snd_soc_dai_digital_mute(dai, 0, substream->stream); out: mutex_unlock(&rtd->card->pcm_mutex); @@ -1219,44 +1154,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* clear the corresponding DAIs parameters when going to be inactive */ - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - if (cpu_dai->active == 1) { - cpu_dai->rate = 0; - cpu_dai->channels = 0; - cpu_dai->sample_bits = 0; - } - } + for_each_rtd_dais(rtd, i, dai) { + int active = dai->stream_active[substream->stream]; - for_each_rtd_codec_dais(rtd, i, codec_dai) { - if (codec_dai->active == 1) { - codec_dai->rate = 0; - codec_dai->channels = 0; - codec_dai->sample_bits = 0; + if (dai->active == 1) { + dai->rate = 0; + dai->channels = 0; + dai->sample_bits = 0; } - } - - /* apply codec digital mute */ - for_each_rtd_codec_dais(rtd, i, codec_dai) { - int active = codec_dai->stream_active[substream->stream]; if (active == 1) - snd_soc_dai_digital_mute(codec_dai, 1, - substream->stream); - } - - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - int active = cpu_dai->stream_active[substream->stream]; - - if (active == 1) - snd_soc_dai_digital_mute(cpu_dai, 1, - substream->stream); + snd_soc_dai_digital_mute(dai, 1, substream->stream); } /* free any machine hw params */ @@ -1266,18 +1180,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ - for_each_rtd_codec_dais(rtd, i, codec_dai) { - if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) - continue; - - snd_soc_dai_hw_free(codec_dai, substream); - } - - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) + for_each_rtd_dais(rtd, i, dai) { + if (!snd_soc_dai_stream_valid(dai, substream->stream)) continue; - snd_soc_dai_hw_free(cpu_dai, substream); + snd_soc_dai_hw_free(dai, substream); } mutex_unlock(&rtd->card->pcm_mutex); @@ -1288,8 +1195,7 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i, ret; ret = soc_rtd_trigger(rtd, substream, cmd); @@ -1302,14 +1208,8 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) return ret; } - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); - if (ret < 0) - return ret; - } - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_trigger(dai, substream, cmd); if (ret < 0) return ret; } @@ -1321,18 +1221,11 @@ static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i, ret; - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); - if (ret < 0) - return ret; - } - - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_trigger(dai, substream, cmd); if (ret < 0) return ret; } @@ -1376,18 +1269,11 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i, ret; - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd); - if (ret < 0) - return ret; - } - - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_bespoke_trigger(dai, substream, cmd); if (ret < 0) return ret; } @@ -1546,7 +1432,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; - for_each_rtd_cpu_dais(be, i, dai) { + for_each_rtd_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); dev_dbg(card->dev, "ASoC: try BE : %s\n", @@ -1555,13 +1441,6 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (w == widget) return be; } - - for_each_rtd_codec_dais(be, i, dai) { - w = snd_soc_dai_get_widget(dai, stream); - - if (w == widget) - return be; - } } /* Widget provided is not a BE */ @@ -1635,27 +1514,18 @@ static bool dpcm_be_is_active(struct snd_soc_dpcm *dpcm, int stream, struct snd_soc_dai *dai; unsigned int i; - /* is there a valid CPU DAI widget for this BE */ - for_each_rtd_cpu_dais(dpcm->be, i, dai) { + /* is there a valid DAI widget for this BE */ + for_each_rtd_dais(dpcm->be, i, dai) { widget = snd_soc_dai_get_widget(dai, stream); /* - * The BE is pruned only if none of the cpu_dai + * The BE is pruned only if none of the dai * widgets are in the active list. */ if (widget && widget_in_list(list, widget)) return true; } - /* is there a valid CODEC DAI widget for this BE */ - for_each_rtd_codec_dais(dpcm->be, i, dai) { - widget = snd_soc_dai_get_widget(dai, stream); - - /* prune the BE if it's no longer in our active list */ - if (widget && widget_in_list(list, widget)) - return true; - } - return false; } @@ -2001,43 +1871,23 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_pcm_stream *codec_stream; - struct snd_soc_pcm_stream *cpu_stream; + struct snd_soc_pcm_stream *pcm; struct snd_soc_dai *dai; int i; - for_each_rtd_cpu_dais(be, i, dai) { + for_each_rtd_dais(be, i, dai) { /* - * Skip CPUs which don't support the current stream + * Skip DAIs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details */ if (!snd_soc_dai_stream_valid(dai, stream)) continue; - cpu_stream = snd_soc_dai_get_pcm_stream(dai, stream); + pcm = snd_soc_dai_get_pcm_stream(dai, stream); - *rate_min = max(*rate_min, cpu_stream->rate_min); - *rate_max = min_not_zero(*rate_max, - cpu_stream->rate_max); - *rates = snd_pcm_rate_mask_intersect(*rates, - cpu_stream->rates); - } - - for_each_rtd_codec_dais(be, i, dai) { - /* - * Skip CODECs which don't support the current stream - * type. See soc_pcm_init_runtime_hw() for more details - */ - if (!snd_soc_dai_stream_valid(dai, stream)) - continue; - - codec_stream = snd_soc_dai_get_pcm_stream(dai, stream); - - *rate_min = max(*rate_min, codec_stream->rate_min); - *rate_max = min_not_zero(*rate_max, - codec_stream->rate_max); - *rates = snd_pcm_rate_mask_intersect(*rates, - codec_stream->rates); + *rate_min = max(*rate_min, pcm->rate_min); + *rate_max = min_not_zero(*rate_max, pcm->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, pcm->rates); } } } @@ -2120,8 +1970,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; + struct snd_soc_dai *dai; int i; /* A backend may not have the requested substream */ @@ -2136,19 +1985,9 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; /* Symmetry only applies if we've got an active stream. */ - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - if (cpu_dai->active) { - err = soc_pcm_apply_symmetry(fe_substream, - cpu_dai); - if (err < 0) - return err; - } - } - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - if (codec_dai->active) { - err = soc_pcm_apply_symmetry(fe_substream, - codec_dai); + for_each_rtd_dais(rtd, i, dai) { + if (dai->active) { + err = soc_pcm_apply_symmetry(fe_substream, dai); if (err < 0) return err; } From d1eb6d116123b2bcebeefce8bcdc828c80b033b8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Mar 2020 15:37:20 +0900 Subject: [PATCH 320/415] ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new() Now CPU/Codec DAIs are alias for dais. Thus, we can directly use for_each_rtd_dais() macro for soc_dai_pcm_new(). This patch merge CPU/Codec for it. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87r1xsolen.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 23 +++++++++-------------- 1 file changed, 9 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 246d59966795b4..843b8b1c89d410 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1037,20 +1037,20 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_add_pcm_runtime); -static int soc_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, - struct snd_soc_pcm_runtime *rtd) +static int soc_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai; int i, ret = 0; - for (i = 0; i < num_dais; ++i) { - struct snd_soc_dai_driver *drv = dais[i]->driver; + for_each_rtd_dais(rtd, i, dai) { + struct snd_soc_dai_driver *drv = dai->driver; if (drv->pcm_new) - ret = drv->pcm_new(rtd, dais[i]); + ret = drv->pcm_new(rtd, dai); if (ret < 0) { - dev_err(dais[i]->dev, + dev_err(dai->dev, "ASoC: Failed to bind %s with pcm device\n", - dais[i]->name); + dai->name); return ret; } } @@ -1121,13 +1121,8 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, dai_link->stream_name, ret); return ret; } - ret = soc_dai_pcm_new(rtd->cpu_dais, - rtd->num_cpus, rtd); - if (ret < 0) - return ret; - ret = soc_dai_pcm_new(rtd->codec_dais, - rtd->num_codecs, rtd); - return ret; + + return soc_dai_pcm_new(rtd); } static void soc_set_name_prefix(struct snd_soc_card *card, From c208a5335036a332c0248b0e16fca57332b3f1d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2020 18:06:42 +0100 Subject: [PATCH 321/415] ALSA: core: Add snd_device_get_state() helper A new small helper to get the current state of the device registration for the given object. It'll be used for USB-audio driver to check the delayed device registrations. Link: https://lore.kernel.org/r/20200323170643.19181-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/core.h | 1 + sound/core/device.c | 21 +++++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index ac8b692b69b45c..381a010a1bd4d2 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -266,6 +266,7 @@ void snd_device_disconnect(struct snd_card *card, void *device_data); void snd_device_disconnect_all(struct snd_card *card); void snd_device_free(struct snd_card *card, void *device_data); void snd_device_free_all(struct snd_card *card); +int snd_device_get_state(struct snd_card *card, void *device_data); /* isadma.c */ diff --git a/sound/core/device.c b/sound/core/device.c index cdc5af52673936..bf0b04a7ee7975 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -237,3 +237,24 @@ void snd_device_free_all(struct snd_card *card) list_for_each_entry_safe_reverse(dev, next, &card->devices, list) __snd_device_free(dev); } + +/** + * snd_device_get_state - Get the current state of the given device + * @card: the card instance + * @device_data: the data pointer to release + * + * Returns the current state of the given device object. For the valid + * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or + * @SNDRV_DEV_DISCONNECTED is returned. + * Or for a non-existing device, -1 is returned as an error. + */ +int snd_device_get_state(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + dev = look_for_dev(card, device_data); + if (dev) + return dev->state; + return -1; +} +EXPORT_SYMBOL_GPL(snd_device_get_state); From a30b59bffcb728b429288a24518bc0891c1122bb Mon Sep 17 00:00:00 2001 From: Dmitry Osipenko Date: Fri, 20 Mar 2020 23:55:04 +0300 Subject: [PATCH 322/415] ASoC: tegra: tegra_wm8903: Support DAPM events for built-in microphone The enable-GPIO needs to be toggled on a DAPM event in order to turn microphone ON/OFF, otherwise microphone won't work. Signed-off-by: Dmitry Osipenko Acked-by: Stephen Warren Acked-by: Jon Hunter Link: https://lore.kernel.org/r/20200320205504.30466-3-digetx@gmail.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index f08d3489c3cf31..071c7d2de77c70 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -143,14 +143,32 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, return 0; } +static int tegra_wm8903_event_int_mic(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_int_mic_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_int_mic_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + static const struct snd_soc_dapm_widget tegra_wm8903_dapm_widgets[] = { SND_SOC_DAPM_SPK("Int Spk", tegra_wm8903_event_int_spk), SND_SOC_DAPM_HP("Headphone Jack", tegra_wm8903_event_hp), SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("Int Mic", tegra_wm8903_event_int_mic), }; static const struct snd_kcontrol_new tegra_wm8903_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), + SOC_DAPM_PIN_SWITCH("Int Mic"), }; static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) From 7efa128e610fb0a44df23f20364c7278a4ead051 Mon Sep 17 00:00:00 2001 From: Dmitry Osipenko Date: Fri, 20 Mar 2020 23:55:03 +0300 Subject: [PATCH 323/415] ASoC: tegra-wm8903: Document built-in microphone audio source The internal microphone source is needed in order to be able to describe the hardware audio routing for devices that have the built-in microphone in addition to the external Mic Jack. Signed-off-by: Dmitry Osipenko Acked-by: Stephen Warren Acked-by: Jon Hunter Link: https://lore.kernel.org/r/20200320205504.30466-2-digetx@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt index b795d282818d8a..a8f2b0c56c795a 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt @@ -18,6 +18,7 @@ Required properties: * Headphone Jack * Int Spk * Mic Jack + * Int Mic - nvidia,i2s-controller : The phandle of the Tegra I2S1 controller - nvidia,audio-codec : The phandle of the WM8903 audio codec From bc765162f730fe0ddd71a268b58c39eec057c082 Mon Sep 17 00:00:00 2001 From: Lukas Bulwahn Date: Sat, 21 Mar 2020 12:40:22 +0100 Subject: [PATCH 324/415] ASoC: MT6660: make spdxcheck.py happy The SPDX-License-Identifier shall not be suffixed with anything further. This makes ./scripts/spdxcheck.py complain: sound/soc/codecs/mt6660.c: 1:36 Invalid token: // Clean up SPDX-License-Identifier line to make spdxcheck.py happy. Signed-off-by: Lukas Bulwahn Link: https://lore.kernel.org/r/20200321114022.8545-1-lukas.bulwahn@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/mt6660.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index bcec82aa57fb9d..d1797003c83dda 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 // +// SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2019 MediaTek Inc. From 9a74c44a6f675e4e991437eee39496109b601629 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 23 Mar 2020 16:25:45 +0800 Subject: [PATCH 325/415] ASoC: rt5682: Add a property for DMIC clock rate The patch adds a property for DMIC clock rate (hz) and changes the default to the common optimize DMIC clock rate. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200323082547.7898-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 9 +++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index 6bf0e3581056eb..96b268ac96bd61 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -38,6 +38,7 @@ struct rt5682_platform_data { enum rt5682_dmic1_clk_pin dmic1_clk_pin; enum rt5682_jd_src jd_src; unsigned int btndet_delay; + unsigned int dmic_clk_rate; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 513429478d27e4..cc00d47895b523 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1231,10 +1231,13 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - int idx = -EINVAL; + int idx = -EINVAL, dmic_clk_rate = 3072000; static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; - idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div)); + if (rt5682->pdata.dmic_clk_rate) + dmic_clk_rate = rt5682->pdata.dmic_clk_rate; + + idx = rt5682_div_sel(rt5682, dmic_clk_rate, div, ARRAY_SIZE(div)); snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1, RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT); @@ -3231,6 +3234,8 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) &rt5682->pdata.jd_src); device_property_read_u32(dev, "realtek,btndet-delay", &rt5682->pdata.btndet_delay); + device_property_read_u32(dev, "realtek,dmic-clk-rate-hz", + &rt5682->pdata.dmic_clk_rate); rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); From 8b15ee0bf80ece9da8787ed5af160a00eb208bd9 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 23 Mar 2020 16:25:46 +0800 Subject: [PATCH 326/415] ASoC: rt5682: Add a property for DMIC delay The patch adds a property for DMIC delay (ms) to avoid pop noise and changes the default delay setting. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200323082547.7898-2-oder_chiou@realtek.com Signed-off-by: Mark Brown --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 12 +++++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index 96b268ac96bd61..e1f790561ac13b 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -39,6 +39,7 @@ struct rt5682_platform_data { enum rt5682_jd_src jd_src; unsigned int btndet_delay; unsigned int dmic_clk_rate; + unsigned int dmic_delay; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index cc00d47895b523..923541a52504e7 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1544,10 +1544,18 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, static int set_dmic_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int delay = 50; + + if (rt5682->pdata.dmic_delay) + delay = rt5682->pdata.dmic_delay; + switch (event) { case SND_SOC_DAPM_POST_PMU: /*Add delay to avoid pop noise*/ - msleep(150); + msleep(delay); break; default: @@ -3236,6 +3244,8 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) &rt5682->pdata.btndet_delay); device_property_read_u32(dev, "realtek,dmic-clk-rate-hz", &rt5682->pdata.dmic_clk_rate); + device_property_read_u32(dev, "realtek,dmic-delay-ms", + &rt5682->pdata.dmic_delay); rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); From 235eb70034a09596bcf6a2a92e9b6c20c47dfae1 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 23 Mar 2020 16:25:47 +0800 Subject: [PATCH 327/415] ASoC: rt5682: Add the descriptions for the DMIC clock rate and delay settings The patch adds the descriptions for the DMIC clock rate and delay settings. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200323082547.7898-3-oder_chiou@realtek.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5682.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index ac98151d29e4ba..ade1ece8b45f82 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -38,6 +38,12 @@ Optional properties: - clocks : phandle and clock specifier for codec MCLK. - clock-names : Clock name string for 'clocks' attribute, should be "mclk". +- realtek,dmic-clk-rate-hz : Set the clock rate (hz) for the requirement of + the particular DMIC. + +- realtek,dmic-delay-ms : Set the delay time (ms) for the requirement of + the particular DMIC. + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 From eedf8a126629bf9db8ad3a2a5dc9dc1798fb2302 Mon Sep 17 00:00:00 2001 From: Jonghwan Choi Date: Thu, 19 Mar 2020 23:00:44 +0900 Subject: [PATCH 328/415] ASoC: tas2562: Fixed incorrect amp_level setting. According to the tas2562 datasheet,the bits[5:1] represents the amp_level value. So to set the amp_level value correctly,the shift value should be set to 1. Signed-off-by: Jonghwan Choi Acked-by: Dan Murphy Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20200319140043.GA6688@jhbirdchoi-MS-7B79 Signed-off-by: Mark Brown --- sound/soc/codecs/tas2562.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index be52886a5edb21..fb2233ca9103fb 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -409,7 +409,7 @@ static const struct snd_kcontrol_new vsense_switch = 1, 1); static const struct snd_kcontrol_new tas2562_snd_controls[] = { - SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, + SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 1, 0x1c, 0, tas2562_dac_tlv), }; From 6b877cf8bc98e6c574a6d763943c4a92592e431c Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Tue, 24 Mar 2020 15:06:15 +0800 Subject: [PATCH 329/415] ASoC: wm8974: remove unused variables sound/soc/codecs/wm8974.c:200:38: warning: wm8974_aux_boost_controls defined but not used [-Wunused-const-variable=] sound/soc/codecs/wm8974.c:204:38: warning: wm8974_mic_boost_controls defined but not used [-Wunused-const-variable=] commit 8a123ee2a46d ("ASoC: WM8974 DAPM cleanups") left behind this, remove them. Signed-off-by: YueHaibing Link: https://lore.kernel.org/r/20200324070615.16248-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index dc4fe4f5239d6f..06ba36595ddd61 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -196,14 +196,6 @@ SOC_DAPM_SINGLE("MicN Switch", WM8974_INPUT, 1, 1, 0), SOC_DAPM_SINGLE("MicP Switch", WM8974_INPUT, 0, 1, 0), }; -/* AUX Input boost vol */ -static const struct snd_kcontrol_new wm8974_aux_boost_controls = -SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0); - -/* Mic Input boost vol */ -static const struct snd_kcontrol_new wm8974_mic_boost_controls = -SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0); - static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0, &wm8974_speaker_mixer_controls[0], From 472abb80fac66b07b6940a40c6392d3dea5dd2a3 Mon Sep 17 00:00:00 2001 From: Johan Jonker Date: Tue, 24 Mar 2020 10:41:47 +0100 Subject: [PATCH 330/415] dt-bindings: sound: convert rockchip i2s bindings to yaml Current dts files with 'i2s' nodes are manually verified. In order to automate this process rockchip-i2s.txt has to be converted to yaml. Signed-off-by: Johan Jonker Reviewed-by: Rob Herring Link: https://lore.kernel.org/r/20200324094149.6904-1-jbx6244@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/rockchip-i2s.txt | 49 -------- .../bindings/sound/rockchip-i2s.yaml | 106 ++++++++++++++++++ 2 files changed, 106 insertions(+), 49 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt deleted file mode 100644 index 54aefab71f2cb1..00000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ /dev/null @@ -1,49 +0,0 @@ -* Rockchip I2S controller - -The I2S bus (Inter-IC sound bus) is a serial link for digital -audio data transfer between devices in the system. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-i2s": for rk3066 - - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 - - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 - - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328 - - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366 - - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 - - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the I2S interrupt. -- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should contain the following: - - "i2s_hclk": clock for I2S BUS - - "i2s_clk" : clock for I2S controller -- rockchip,playback-channels: max playback channels, if not set, 8 channels default. -- rockchip,capture-channels: max capture channels, if not set, 2 channels default. - -Required properties for controller which support multi channels -playback/capture: - -- rockchip,grf: the phandle of the syscon node for GRF register. - -Example for rk3288 I2S controller: - -i2s@ff890000 { - compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; - reg = <0xff890000 0x10000>; - interrupts = ; - dmas = <&pdma1 0>, <&pdma1 1>; - dma-names = "tx", "rx"; - clock-names = "i2s_hclk", "i2s_clk"; - clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; - rockchip,playback-channels = <8>; - rockchip,capture-channels = <2>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml new file mode 100644 index 00000000000000..eff06b4b51db94 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -0,0 +1,106 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S controller + +description: + The I2S bus (Inter-IC sound bus) is a serial link for digital + audio data transfer between devices in the system. + +maintainers: + - Heiko Stuebner + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-i2s + - items: + - enum: + - rockchip,px30-i2s + - rockchip,rk3036-i2s + - rockchip,rk3188-i2s + - rockchip,rk3228-i2s + - rockchip,rk3288-i2s + - rockchip,rk3328-i2s + - rockchip,rk3366-i2s + - rockchip,rk3368-i2s + - rockchip,rk3399-i2s + - const: rockchip,rk3066-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for I2S controller + - description: clock for I2S BUS + + clock-names: + items: + - const: i2s_clk + - const: i2s_hclk + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + rockchip,capture-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 2 + description: + Max capture channels, if not set, 2 channels default. + + rockchip,playback-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 8 + description: + Max playback channels, if not set, 8 channels default. + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property for controllers which support multi channel + playback/capture. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include + #include + #include + i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = ; + clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>; + clock-names = "i2s_clk", "i2s_hclk"; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "tx", "rx"; + rockchip,capture-channels = <2>; + rockchip,playback-channels = <8>; + }; From 515d2757d31e1c9587f166dba5e828fb8569df84 Mon Sep 17 00:00:00 2001 From: Johan Jonker Date: Tue, 24 Mar 2020 10:41:48 +0100 Subject: [PATCH 331/415] dt-bindings: sound: rockchip-i2s: add #sound-dai-cells property '#sound-dai-cells' is required to properly interpret the list of DAI specified in the 'sound-dai' property, so add them to 'rockchip-i2s.yaml' Signed-off-by: Johan Jonker Reviewed-by: Rob Herring Link: https://lore.kernel.org/r/20200324094149.6904-2-jbx6244@gmail.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rockchip-i2s.yaml | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml index eff06b4b51db94..7cd0e278ed85a1 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -77,6 +77,9 @@ properties: Required property for controllers which support multi channel playback/capture. + "#sound-dai-cells": + const: 0 + required: - compatible - reg @@ -85,6 +88,7 @@ required: - clock-names - dmas - dma-names + - "#sound-dai-cells" additionalProperties: false @@ -103,4 +107,5 @@ examples: dma-names = "tx", "rx"; rockchip,capture-channels = <2>; rockchip,playback-channels = <8>; + #sound-dai-cells = <0>; }; From bde8ca7c87d4388e24195f6c84cd9ac775344d2b Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 6 Mar 2020 23:29:30 +0100 Subject: [PATCH 332/415] ASoC: jz4740-i2s: Add support for the JZ4760 The change of offset for the {rx,tx}_threshold fields in the conf register predates the JZ4780, and was first introduced in the JZ4760. Signed-off-by: Paul Cercueil Link: https://lore.kernel.org/r/20200306222931.39664-5-paul@crapouillou.net Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 3f9b2e1b47474a..253f8d8ba27345 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -49,12 +49,8 @@ #define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12 #define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8 -#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24 -#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16 -#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_MASK \ - (0xf << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) -#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_MASK \ - (0x1f << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) +#define JZ4760_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24 +#define JZ4760_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16 #define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19) #define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16) @@ -90,6 +86,7 @@ enum jz47xx_i2s_version { JZ_I2S_JZ4740, + JZ_I2S_JZ4760, JZ_I2S_JZ4780, }; @@ -403,9 +400,9 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, &i2s->capture_dma_data); - if (i2s->soc_info->version >= JZ_I2S_JZ4780) { - conf = (7 << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | - (8 << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | + if (i2s->soc_info->version >= JZ_I2S_JZ4760) { + conf = (7 << JZ4760_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | + (8 << JZ4760_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | JZ_AIC_CONF_OVERFLOW_PLAY_LAST | JZ_AIC_CONF_I2S | JZ_AIC_CONF_INTERNAL_CODEC; @@ -467,6 +464,11 @@ static const struct i2s_soc_info jz4740_i2s_soc_info = { .dai = &jz4740_i2s_dai, }; +static const struct i2s_soc_info jz4760_i2s_soc_info = { + .version = JZ_I2S_JZ4760, + .dai = &jz4740_i2s_dai, +}; + static struct snd_soc_dai_driver jz4780_i2s_dai = { .probe = jz4740_i2s_dai_probe, .remove = jz4740_i2s_dai_remove, @@ -499,6 +501,7 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { #ifdef CONFIG_OF static const struct of_device_id jz4740_of_matches[] = { { .compatible = "ingenic,jz4740-i2s", .data = &jz4740_i2s_soc_info }, + { .compatible = "ingenic,jz4760-i2s", .data = &jz4760_i2s_soc_info }, { .compatible = "ingenic,jz4780-i2s", .data = &jz4780_i2s_soc_info }, { /* sentinel */ } }; From a3434a497a2f33324e0f47bc1500a400959b4b25 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 6 Mar 2020 23:29:31 +0100 Subject: [PATCH 333/415] ASoC: jz4740-i2s: Add support for the JZ4770 Before the JZ4770, the playback and capture sampling rates had to match. The JZ4770 supports independent sampling rates for both. Signed-off-by: Paul Cercueil Link: https://lore.kernel.org/r/20200306222931.39664-6-paul@crapouillou.net Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 253f8d8ba27345..6f6f8dad03565c 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -87,6 +87,7 @@ enum jz47xx_i2s_version { JZ_I2S_JZ4740, JZ_I2S_JZ4760, + JZ_I2S_JZ4770, JZ_I2S_JZ4780, }; @@ -286,7 +287,7 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; - if (i2s->soc_info->version >= JZ_I2S_JZ4780) { + if (i2s->soc_info->version >= JZ_I2S_JZ4770) { div_reg &= ~I2SDIV_IDV_MASK; div_reg |= (div - 1) << I2SDIV_IDV_SHIFT; } else { @@ -469,7 +470,7 @@ static const struct i2s_soc_info jz4760_i2s_soc_info = { .dai = &jz4740_i2s_dai, }; -static struct snd_soc_dai_driver jz4780_i2s_dai = { +static struct snd_soc_dai_driver jz4770_i2s_dai = { .probe = jz4740_i2s_dai_probe, .remove = jz4740_i2s_dai_remove, .playback = { @@ -487,9 +488,14 @@ static struct snd_soc_dai_driver jz4780_i2s_dai = { .ops = &jz4740_i2s_dai_ops, }; +static const struct i2s_soc_info jz4770_i2s_soc_info = { + .version = JZ_I2S_JZ4770, + .dai = &jz4770_i2s_dai, +}; + static const struct i2s_soc_info jz4780_i2s_soc_info = { .version = JZ_I2S_JZ4780, - .dai = &jz4780_i2s_dai, + .dai = &jz4770_i2s_dai, }; static const struct snd_soc_component_driver jz4740_i2s_component = { @@ -502,6 +508,7 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { static const struct of_device_id jz4740_of_matches[] = { { .compatible = "ingenic,jz4740-i2s", .data = &jz4740_i2s_soc_info }, { .compatible = "ingenic,jz4760-i2s", .data = &jz4760_i2s_soc_info }, + { .compatible = "ingenic,jz4770-i2s", .data = &jz4770_i2s_soc_info }, { .compatible = "ingenic,jz4780-i2s", .data = &jz4780_i2s_soc_info }, { /* sentinel */ } }; From 129a5d4824d5fc4a8c155d4349492caaf1a4ea28 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 6 Mar 2020 23:29:26 +0100 Subject: [PATCH 334/415] ASoC: Convert jz4740-i2s doc to YAML Convert the textual binding documentation for the AIC (AC97/I2S Controller) of Ingenic SoCs to a YAML schema, and add the new compatible strings in the process. Signed-off-by: Paul Cercueil Reviewed-by: Rob Herring Link: https://lore.kernel.org/r/20200306222931.39664-1-paul@crapouillou.net Signed-off-by: Mark Brown --- .../bindings/sound/ingenic,aic.yaml | 92 +++++++++++++++++++ .../bindings/sound/ingenic,jz4740-i2s.txt | 23 ----- 2 files changed, 92 insertions(+), 23 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,aic.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/ingenic,aic.yaml b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml new file mode 100644 index 00000000000000..44f49bebb267c0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,aic.yaml @@ -0,0 +1,92 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ingenic,aic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Ingenic SoCs AC97 / I2S Controller (AIC) DT bindings + +maintainers: + - Paul Cercueil + +properties: + $nodename: + pattern: '^audio-controller@' + + compatible: + oneOf: + - enum: + - ingenic,jz4740-i2s + - ingenic,jz4760-i2s + - ingenic,jz4770-i2s + - ingenic,jz4780-i2s + - items: + - const: ingenic,jz4725b-i2s + - const: ingenic,jz4740-i2s + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: AIC clock + - description: I2S clock + - description: EXT clock + - description: PLL/2 clock + + clock-names: + items: + - const: aic + - const: i2s + - const: ext + - const: pll half + + dmas: + items: + - description: DMA controller phandle and request line for I2S RX + - description: DMA controller phandle and request line for I2S TX + + dma-names: + items: + - const: rx + - const: tx + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - '#sound-dai-cells' + +examples: + - | + #include + aic: audio-controller@10020000 { + compatible = "ingenic,jz4740-i2s"; + reg = <0x10020000 0x38>; + + #sound-dai-cells = <0>; + + interrupt-parent = <&intc>; + interrupts = <18>; + + clocks = <&cgu JZ4740_CLK_AIC>, + <&cgu JZ4740_CLK_I2S>, + <&cgu JZ4740_CLK_EXT>, + <&cgu JZ4740_CLK_PLL_HALF>; + clock-names = "aic", "i2s", "ext", "pll half"; + + dmas = <&dmac 25 0xffffffff>, <&dmac 24 0xffffffff>; + dma-names = "rx", "tx"; + }; diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt deleted file mode 100644 index b623d50004fb01..00000000000000 --- a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt +++ /dev/null @@ -1,23 +0,0 @@ -Ingenic JZ4740 I2S controller - -Required properties: -- compatible : "ingenic,jz4740-i2s" or "ingenic,jz4780-i2s" -- reg : I2S registers location and length -- clocks : AIC and I2S PLL clock specifiers. -- clock-names: "aic" and "i2s" -- dmas: DMA controller phandle and DMA request line for I2S Tx and Rx channels -- dma-names: Must be "tx" and "rx" - -Example: - -i2s: i2s@10020000 { - compatible = "ingenic,jz4740-i2s"; - reg = <0x10020000 0x94>; - - clocks = <&cgu JZ4740_CLK_AIC>, <&cgu JZ4740_CLK_I2SPLL>; - clock-names = "aic", "i2s"; - - dmas = <&dma 2>, <&dma 3>; - dma-names = "tx", "rx"; - -}; From 68999d939dcfb430dd9834df56c71e43b7f37c6e Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 25 Mar 2020 14:16:09 +0100 Subject: [PATCH 335/415] ASoC: Intel: broadwell: Revert back SSP0 link to use dummy components Recent series of patches targeting broadwell boards, while enabling SOF, changed behavior for non-SOF solutions. In essence replacing platform 'dummy' with actual 'platform' causes redundant stream initialization to occur during audio start. hw_params for haswell-pcm destroys initial stream right after its creation - only to recreate it again from proceed from there. While harmless so far, this flow isn't correct and should be corrected. The actual need for dummy components for SSP0 link is questionable but that issue is subject for another series. Link to first message in conversation: https://lkml.org/lkml/2020/3/18/54 Fixes: 64df6afa0dab ("ASoC: Intel: broadwell: change cpu_dai and platform components for SOF") Reported-by: Dominik Brodowski Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325131611.545-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/broadwell.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index b9c12e24c70bc0..ffbb597052aa61 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -167,9 +167,6 @@ SND_SOC_DAILINK_DEF(codec, #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); -#else -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_DUMMY())); #endif /* broadwell digital audio interface glue - connects codec <--> CPU */ @@ -226,7 +223,11 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .ops = &broadwell_rt286_ops, .dpcm_playback = 1, .dpcm_capture = 1, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + SND_SOC_DAILINK_REG(dummy, codec, dummy), +#else SND_SOC_DAILINK_REG(ssp0_port, codec, platform), +#endif }, }; From c031d3de80a4f6127fc7b953fdaeead934e61c1a Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 25 Mar 2020 14:16:10 +0100 Subject: [PATCH 336/415] ASoC: Intel: bdw-rt5677: Revert SSP0 link to use dummy components Recent series of patches targeting broadwell boards, while enabling SOF, changed behavior for non-SOF solutions. In essence replacing platform 'dummy' with actual 'platform' causes redundant stream initialization to occur during audio start. hw_params for haswell-pcm destroys initial stream right after its creation - only to recreate it again from proceed from there. While harmless so far, this flow isn't right and should be corrected. The actual need for dummy components for SSP0 link is questionable but that issue is subject for another series. Fixes: 4865bde187b2 ("ASoC: Intel: bdw-rt5677: change cpu_dai and platform components for SOF") Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325131611.545-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5677.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index bb643c99069dfd..63108e1a0e4d7e 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -298,9 +298,6 @@ SND_SOC_DAILINK_DEF(be, #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); -#else -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_DUMMY())); #endif /* Wake on voice interface */ @@ -350,7 +347,11 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, .init = bdw_rt5677_init, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + SND_SOC_DAILINK_REG(dummy, be, dummy), +#else SND_SOC_DAILINK_REG(ssp0_port, be, platform), +#endif }, }; From f25e203070e5b12e4db366ee99b86f33a968f1ae Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 25 Mar 2020 14:16:11 +0100 Subject: [PATCH 337/415] ASoC: Intel: bdw-rt5650: Revert SSP0 link to use dummy components Recent series of patches targeting broadwell boards, while enabling SOF, changed behavior for non-SOF solutions. In essence replacing platform 'dummy' with actual 'platform' causes redundant stream initialization to occur during audio start. hw_params for haswell-pcm destroys initial stream right after its creation - only to recreate it again from proceed from there. While harmless so far, this flow isn't right and should be corrected. The actual need for dummy components for SSP0 link is questionable but that issue is subject for another series. Fixes: a40acc6bfceb ("ASoC: Intel: bdw-rt5650: change cpu_dai and platform components for SOF") Signed-off-by: Cezary Rojewski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325131611.545-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5650.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index 1a302436d45037..6cff603d465658 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -226,9 +226,6 @@ SND_SOC_DAILINK_DEF(be, #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); -#else -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_DUMMY())); #endif static struct snd_soc_dai_link bdw_rt5650_dais[] = { @@ -264,7 +261,11 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, .init = bdw_rt5650_init, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + SND_SOC_DAILINK_REG(dummy, be, dummy), +#else SND_SOC_DAILINK_REG(ssp0_port, be, platform), +#endif }, }; From 633fddee7355e46a5b5ec471abb58d65e1e41012 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 25 Mar 2020 13:29:13 +0000 Subject: [PATCH 338/415] ASoC: mchp-i2s-mcc: make signed 1 bit bitfields unsigned The signed 1 bit bitfields should be unsigned, so make them unsigned. Signed-off-by: Colin Ian King Reviewed-by: Codrin Ciubotariu Link: https://lore.kernel.org/r/20200325132913.110115-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-i2s-mcc.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index befc2a3a05b050..3cb63886195ff5 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -239,10 +239,10 @@ struct mchp_i2s_mcc_dev { unsigned int frame_length; int tdm_slots; int channels; - int gclk_use:1; - int gclk_running:1; - int tx_rdy:1; - int rx_rdy:1; + unsigned int gclk_use:1; + unsigned int gclk_running:1; + unsigned int tx_rdy:1; + unsigned int rx_rdy:1; }; static irqreturn_t mchp_i2s_mcc_interrupt(int irq, void *dev_id) From d8695bc5b1fe88305396b1f788d3b5f218e28a30 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Mar 2020 11:33:19 +0100 Subject: [PATCH 339/415] ALSA: usb-audio: Rewrite registration quirk handling A slight refactoring of the registration quirk code. Now it uses the table lookup for easy additions in future. Also the return type was changed to bool, and got a few more comments. Link: https://lore.kernel.org/r/20200325103322.2508-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 +- sound/usb/quirks.c | 40 ++++++++++++++++++++++++++++++---------- sound/usb/quirks.h | 3 +-- 3 files changed, 32 insertions(+), 13 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 16bbe2a50fb7b5..55d563a8154dfd 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -665,7 +665,7 @@ static int usb_audio_probe(struct usb_interface *intf, /* we are allowed to call snd_card_register() many times, but first * check to see if a device needs to skip it or do anything special */ - if (snd_usb_registration_quirk(chip, ifnum) == 0) { + if (!snd_usb_registration_quirk(chip, ifnum)) { err = snd_card_register(chip->card); if (err < 0) goto __error; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d605aff801b848..86f192a3043dfe 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1809,16 +1809,36 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, } } -int snd_usb_registration_quirk(struct snd_usb_audio *chip, - int iface) +/* + * registration quirk: + * the registration is skipped if a device matches with the given ID, + * unless the interface reaches to the defined one. This is for delaying + * the registration until the last known interface, so that the card and + * devices appear at the same time. + */ + +struct registration_quirk { + unsigned int usb_id; /* composed via USB_ID() */ + unsigned int interface; /* the interface to trigger register */ +}; + +#define REG_QUIRK_ENTRY(vendor, product, iface) \ + { .usb_id = USB_ID(vendor, product), .interface = (iface) } + +static const struct registration_quirk registration_quirks[] = { + REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ + { 0 } /* terminator */ +}; + +/* return true if skipping registration */ +bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface) { - switch (chip->usb_id) { - case USB_ID(0x0951, 0x16d8): /* Kingston HyperX AMP */ - /* Register only when we reach interface 2 so that streams can - * merge correctly into PCMs from interface 0 - */ - return (iface != 2); - } + const struct registration_quirk *q; + + for (q = registration_quirks; q->usb_id; q++) + if (chip->usb_id == q->usb_id) + return iface != q->interface; + /* Register as normal */ - return 0; + return false; } diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 3afc01eabc7e2a..c76cf24a640a6f 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -51,7 +51,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, struct audioformat *fp, int stream); -int snd_usb_registration_quirk(struct snd_usb_audio *chip, - int iface); +bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface); #endif /* __USBAUDIO_QUIRKS_H */ From b70038ef4feae9c5c9d01de4cd799ae52ccafc08 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Mar 2020 11:33:20 +0100 Subject: [PATCH 340/415] ALSA: usb-audio: Add delayed_register option Add a new option for specifying the quirk for delayed registration of the certain device. A list of devices can be passed in a form ID:IFACE,ID:IFACE,ID:IFACE,.... where ID is the 32bit hex number combo of vendor and device IDs and IFACE is the interface number to trigger the register. When a matching device is probed, the card registration is delayed until the given interface is probed. It's needed for syncing the registration until the last interface when multiple interfaces are provided for the same card. Link: https://lore.kernel.org/r/20200325103322.2508-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 55d563a8154dfd..9511342386691a 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -72,6 +72,7 @@ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; +static char *delayed_register[SNDRV_CARDS]; bool snd_usb_use_vmalloc = true; bool snd_usb_skip_validation; @@ -95,6 +96,8 @@ module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); +module_param_array(delayed_register, charp, NULL, 0444); +MODULE_PARM_DESC(delayed_register, "Quirk for delayed registration, given by id:iface, e.g. 0123abcd:4."); module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444); MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes)."); module_param_named(skip_validation, snd_usb_skip_validation, bool, 0444); @@ -525,6 +528,21 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id) return false; } +static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) +{ + int i; + unsigned int id, inum; + + for (i = 0; i < ARRAY_SIZE(delayed_register); i++) { + if (delayed_register[i] && + sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 && + id == chip->usb_id) + return inum != iface; + } + + return false; +} + static const struct usb_device_id usb_audio_ids[]; /* defined below */ /* look for the corresponding quirk */ @@ -665,7 +683,8 @@ static int usb_audio_probe(struct usb_interface *intf, /* we are allowed to call snd_card_register() many times, but first * check to see if a device needs to skip it or do anything special */ - if (!snd_usb_registration_quirk(chip, ifnum)) { + if (!snd_usb_registration_quirk(chip, ifnum) && + !check_delayed_register_option(chip, ifnum)) { err = snd_card_register(chip->card); if (err < 0) goto __error; From a4aad5636c7298eecfb60fc9d73933336cb0c4d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Mar 2020 11:33:21 +0100 Subject: [PATCH 341/415] ALSA: usb-audio: Inform devices that need delayed registration The USB-audio driver may call snd_card_register() multiple times as its probe function is per USB interface while some USB-audio devices may provide multiple interfaces to assign different streams although they belong to the same device. This works in most cases but the registration is racy, hence it may miss the device recognition, e.g. PA doesn't see certain devices when hotplugged. The recent addition of the delayed registration quirk allows to sync the registration at the last known interface, and the previous commit added a new module option to allow the dynamic setup for that purpose. Now, this patch tries to find out and notifies for such devices that require the delayed registration. It shows a message like: Found post-registration device assignment: 1234abcd:02 If you hit this message, you can pass delayed_register module option like: snd_usb_audio.delayed_register=1234abcd:02 by just copying the last shown entry. If this works, it can be added statically in the quirk list, registration_quirks[] found at the end of sound/usb/quirks.c. Link: https://lore.kernel.org/r/20200325103322.2508-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 7 +++++++ sound/usb/stream.c | 3 +++ sound/usb/usbaudio.h | 1 + 3 files changed, 11 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9511342386691a..fd6fd1726ea0bd 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -680,6 +680,13 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } + if (chip->need_delayed_register) { + dev_info(&dev->dev, + "Found post-registration device assignment: %08x:%02x\n", + chip->usb_id, ifnum); + chip->need_delayed_register = false; /* clear again */ + } + /* we are allowed to call snd_card_register() many times, but first * check to see if a device needs to skip it or do anything special */ diff --git a/sound/usb/stream.c b/sound/usb/stream.c index afd5aa574611e6..15296f2c902cd3 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -502,6 +502,9 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, subs = &as->substream[stream]; if (subs->ep_num) continue; + if (snd_device_get_state(chip->card, as->pcm) != + SNDRV_DEV_BUILD) + chip->need_delayed_register = true; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 6fe3ab582ec6ac..1c892c7f14d78b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -34,6 +34,7 @@ struct snd_usb_audio { unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */ + unsigned int need_delayed_register:1; /* warn for delayed registration */ int num_interfaces; int num_suspended_intf; int sample_rate_read_error; From 2ceb65cd04a9b7d694188b72807f2e1fead16d9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Mar 2020 11:33:22 +0100 Subject: [PATCH 342/415] ALSA: usb-audio: Update the documentation for the new delayed_register option Just adding a brief explanation to alsa-configuration.rst. Link: https://lore.kernel.org/r/20200325103322.2508-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- Documentation/sound/alsa-configuration.rst | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 392875a1b94e74..72f97d4b01a784 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2234,6 +2234,19 @@ use_vmalloc buffers. If mmap is used on such architectures, turn off this option, so that the DMA-coherent buffers are allocated and used instead. +delayed_register + The option is needed for devices that have multiple streams + defined in multiple USB interfaces. The driver may invoke + registrations multiple times (once per interface) and this may + lead to the insufficient device enumeration. + This option receives an array of strings, and you can pass + ID:INTERFACE like ``0123abcd:4`` for performing the delayed + registration to the given device. In this example, when a USB + device 0123:abcd is probed, the driver waits the registration + until the USB interface 4 gets probed. + The driver prints a message like "Found post-registration device + assignment: 1234abcd:04" for such a device, so that user can + notice the need. This module supports multiple devices, autoprobe and hotplugging. From d60b55c9edaed31e8e0c961f42237dcb5c83deb8 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Wed, 25 Mar 2020 16:32:42 -0500 Subject: [PATCH 343/415] ASoC: Intel: Make glk+rt5682 echo ref dynamic Without the dynamic flag to allow runtime routing, the card cannot probe on chromebooks because SOF is constantly waiting for the link. Adding flag back to allow upstream kernels to work on rt5682 based chromebooks since SOF can now ignore the hard coded front end. Signed-off-by: Curtis Malainey Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325213245.28247-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/glk_rt5682_max98357a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index ea1de8b3f3cdd4..3c576b33b9c644 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -409,6 +409,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .init = NULL, .capture_only = 1, .nonatomic = 1, + .dynamic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, [GLK_DPCM_AUDIO_REF_CP] = { From 90c49d6a1f2446ae61b3631b4d8950842a4b3edf Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Wed, 25 Mar 2020 16:32:43 -0500 Subject: [PATCH 344/415] ASoC: intel: sof_da7219_max98373: Add speaker switch Add "Spk Switch" and associated widget, route to max98360a speaker amp for power saving, also remove the speaker_amp_init() callback with complete separated tables for max98373 and max98360a. Signed-off-by: Bhat, Uday M Signed-off-by: Pierre-Louis Bossart Signed-off-by: Yong Zhi Link: https://lore.kernel.org/r/20200325213245.28247-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219_max98373.c | 67 +++++++++++--------- 1 file changed, 37 insertions(+), 30 deletions(-) diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 6d210ba06d195e..239d8ffdbccd9f 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -72,11 +72,17 @@ static const struct snd_kcontrol_new controls[] = { static const struct snd_kcontrol_new m98360a_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), }; +/* For MAX98373 amp */ static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), @@ -90,17 +96,33 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "Platform Clock" }, { "Headset Mic", NULL, "Platform Clock" }, + + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, }; -/* For MAX98373 amp */ -static const struct snd_soc_dapm_widget max98373_widgets[] = { - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), +/* For MAX98360A amp */ +static const struct snd_soc_dapm_widget max98360a_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_SPK("Spk", NULL), + + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU), }; -static const struct snd_soc_dapm_route max98373_map[] = { - { "Left Spk", NULL, "Left BE_OUT" }, - { "Right Spk", NULL, "Right BE_OUT" }, +static const struct snd_soc_dapm_route max98360a_map[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + { "MIC", NULL, "Headset Mic" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, + + {"Spk", NULL, "Speaker"}, }; static struct snd_soc_jack headset; @@ -144,21 +166,6 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static int speaker_amp_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - - /* Add widgets */ - ret = snd_soc_dapm_new_controls(&rtd->card->dapm, max98373_widgets, - ARRAY_SIZE(max98373_widgets)); - if (ret) - return ret; - - /* Add routes */ - return snd_soc_dapm_add_routes(&rtd->card->dapm, max98373_map, - ARRAY_SIZE(max98373_map)); -} - static int ssp1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -249,8 +256,9 @@ SND_SOC_DAILINK_DEF(ssp1_amps, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI), /* Right */ COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI))); -/* For the driver-less spk amp */ -SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); + +SND_SOC_DAILINK_DEF(ssp1_m98360a, + DAILINK_COMP_ARRAY(COMP_CODEC("MX98360A:00", "HiFi"))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -282,7 +290,6 @@ static struct snd_soc_dai_link dais[] = { .id = 0, .ignore_pmdown_time = 1, .no_pcm = 1, - .init = speaker_amp_init, .dpcm_playback = 1, .dpcm_capture = 1, /* IV feedback */ .ops = &ssp1_ops, @@ -356,10 +363,10 @@ static struct snd_soc_card card_da7219_m98360a = { .num_links = ARRAY_SIZE(dais), .controls = m98360a_controls, .num_controls = ARRAY_SIZE(m98360a_controls), - .dapm_widgets = widgets, - .num_dapm_widgets = ARRAY_SIZE(widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), + .dapm_widgets = max98360a_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98360a_widgets), + .dapm_routes = max98360a_map, + .num_dapm_routes = ARRAY_SIZE(max98360a_map), .fully_routed = true, .late_probe = card_late_probe, }; @@ -383,7 +390,7 @@ static int audio_probe(struct platform_device *pdev) .no_pcm = 1, .dpcm_playback = 1, .ignore_pmdown_time = 1, - SND_SOC_DAILINK_REG(ssp1_pin, dummy, platform) }; + SND_SOC_DAILINK_REG(ssp1_pin, ssp1_m98360a, platform) }; } INIT_LIST_HEAD(&ctx->hdmi_pcm_list); From e2e404a6164e526193f78717de060cd9b27b3b90 Mon Sep 17 00:00:00 2001 From: Sathyanarayana Nujella Date: Wed, 25 Mar 2020 16:32:44 -0500 Subject: [PATCH 345/415] ASoC: Intel: sof_rt5682: Add support for tgl-max98373-rt5682 This patch does the below: 1. Adds the driver data and updates quirk info for TGL with Max98373 speaker amp and ALC5682 headset codec. 2. Added max98373 speaker related code to common file for re-use. Signed-off-by: Jairaj Arava Signed-off-by: Sathyanarayana Nujella Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325213245.28247-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 80 +++++++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 24 +++++++ sound/soc/intel/boards/sof_rt5682.c | 21 ++++++ 5 files changed, 127 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/boards/sof_maxim_common.c create mode 100644 sound/soc/intel/boards/sof_maxim_common.h diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index fb8d83518c4733..f18dd9fde97315 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -450,6 +450,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) depends on SND_HDA_CODEC_HDMI + select SND_SOC_MAX98373 select SND_SOC_RT1015 select SND_SOC_RT5682 select SND_SOC_DMIC diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 781e7ec58e47a0..e083ceeccdad63 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -19,7 +19,7 @@ snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o -snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o +snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c new file mode 100644 index 00000000000000..463b39a7ccfd0e --- /dev/null +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -0,0 +1,80 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright(c) 2020 Intel Corporation. All rights reserved. +#include +#include +#include +#include +#include +#include +#include "sof_maxim_common.h" + +static const struct snd_soc_dapm_route max_98373_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static struct snd_soc_codec_conf max_98373_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98373_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98373_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +struct snd_soc_dai_link_component max_98373_components[] = { + { /* For Left */ + .name = MAX_98373_DEV0_NAME, + .dai_name = MAX_98373_CODEC_DAI, + }, + { /* For Right */ + .name = MAX_98373_DEV1_NAME, + .dai_name = MAX_98373_CODEC_DAI, + }, +}; + +static int max98373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) { + /* DEV0 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + } + if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) { + /* DEV1 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + } + } + return 0; +} + +struct snd_soc_ops max_98373_ops = { + .hw_params = max98373_hw_params, +}; + +int max98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, + ARRAY_SIZE(max_98373_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_max98373_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = max_98373_codec_conf; + card->num_configs = ARRAY_SIZE(max_98373_codec_conf); +} diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h new file mode 100644 index 00000000000000..406bf0e8115588 --- /dev/null +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2020 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with Maxim Codecs. + */ +#ifndef __SOF_MAXIM_COMMON_H +#define __SOF_MAXIM_COMMON_H + +#include + +#define MAX_98373_CODEC_DAI "max98373-aif1" +#define MAX_98373_DEV0_NAME "i2c-MX98373:00" +#define MAX_98373_DEV1_NAME "i2c-MX98373:01" + +extern struct snd_soc_dai_link_component max_98373_components[2]; +extern struct snd_soc_ops max_98373_ops; + +int max98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); +void sof_max98373_codec_conf(struct snd_soc_card *card); +#endif /* __SOF_MAXIM_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 6defe7c85c32da..2eeaa14e59c01a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -23,6 +23,7 @@ #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" +#include "sof_maxim_common.h" #define NAME_SIZE 32 @@ -41,6 +42,7 @@ #define SOF_RT5682_NUM_HDMIDEV(quirk) \ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) #define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(14) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -637,6 +639,12 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].num_codecs = ARRAY_SIZE(rt1015_components); links[id].init = speaker_codec_init_lr; links[id].ops = &sof_rt1015_ops; + } else if (sof_rt5682_quirk & + SOF_MAX98373_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98373_components; + links[id].num_codecs = ARRAY_SIZE(max_98373_components); + links[id].init = max98373_spk_codec_init; + links[id].ops = &max_98373_ops; } else { links[id].codecs = max98357a_component; links[id].num_codecs = ARRAY_SIZE(max98357a_component); @@ -745,6 +753,9 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_rt5682.num_links++; + if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) + sof_max98373_codec_conf(&sof_audio_card_rt5682); + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); if (!dai_links) @@ -811,6 +822,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT1015_SPEAKER_AMP_PRESENT | SOF_RT5682_SSP_AMP(1)), }, + { + .name = "tgl_max98373_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; @@ -833,3 +853,4 @@ MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); MODULE_ALIAS("platform:tgl_max98357a_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_rt1015"); +MODULE_ALIAS("platform:tgl_max98373_rt5682"); From eb1006c6ecf931e7f63d551e38569fbe4ebd5c1e Mon Sep 17 00:00:00 2001 From: Sathyanarayana Nujella Date: Wed, 25 Mar 2020 16:32:45 -0500 Subject: [PATCH 346/415] ASoC: Intel: common: Add mach table for tgl-max98373-rt5682 Update tgl mach table with: Maxim98373 Amp and ALC5682 hp codec. Both of the codecs are on I2S bus. Signed-off-by: Jairaj Arava Signed-off-by: Sathyanarayana Nujella Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325213245.28247-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 5984dd151f3e84..c15eae402b18cc 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -46,6 +46,11 @@ static const struct snd_soc_acpi_link_adr tgl_rvp[] = { {} }; +static struct snd_soc_acpi_codecs tgl_max98373_amp = { + .num_codecs = 1, + .codecs = {"MX98373"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", @@ -63,6 +68,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "tgl_max98373_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &tgl_max98373_amp, + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); From 66de6beb933d373224f350834fbab68093d24627 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 25 Mar 2020 16:12:29 -0500 Subject: [PATCH 347/415] ASoC: SOF: Intel: hda: Improve DSP state logging Improve the DSP power state logs with the state names instead of values. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325211233.27394-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 43 +++++++++++++++++++++++++++++++++-- 1 file changed, 41 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 79ce52c32ef1a8..c396b7ef0328d9 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -452,6 +452,46 @@ static int hda_dsp_set_D0_state(struct snd_sof_dev *sdev, return ret; } +/* helper to log DSP state */ +static void hda_dsp_state_log(struct snd_sof_dev *sdev) +{ + switch (sdev->dsp_power_state.state) { + case SOF_DSP_PM_D0: + switch (sdev->dsp_power_state.substate) { + case SOF_HDA_DSP_PM_D0I0: + dev_dbg(sdev->dev, "Current DSP power state: D0I0\n"); + break; + case SOF_HDA_DSP_PM_D0I3: + dev_dbg(sdev->dev, "Current DSP power state: D0I3\n"); + break; + default: + dev_dbg(sdev->dev, "Unknown DSP D0 substate: %d\n", + sdev->dsp_power_state.substate); + break; + } + break; + case SOF_DSP_PM_D1: + dev_dbg(sdev->dev, "Current DSP power state: D1\n"); + break; + case SOF_DSP_PM_D2: + dev_dbg(sdev->dev, "Current DSP power state: D2\n"); + break; + case SOF_DSP_PM_D3_HOT: + dev_dbg(sdev->dev, "Current DSP power state: D3_HOT\n"); + break; + case SOF_DSP_PM_D3: + dev_dbg(sdev->dev, "Current DSP power state: D3\n"); + break; + case SOF_DSP_PM_D3_COLD: + dev_dbg(sdev->dev, "Current DSP power state: D3_COLD\n"); + break; + default: + dev_dbg(sdev->dev, "Unknown DSP power state: %d\n", + sdev->dsp_power_state.state); + break; + } +} + /* * All DSP power state transitions are initiated by the driver. * If the requested state change fails, the error is simply returned. @@ -511,8 +551,7 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev, } sdev->dsp_power_state = *target_state; - dev_dbg(sdev->dev, "New DSP state %d substate %d\n", - target_state->state, target_state->substate); + hda_dsp_state_log(sdev); return ret; } From c688cf1d3a2cc1bca5737e7849325b3ac8e69a41 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 25 Mar 2020 16:12:30 -0500 Subject: [PATCH 348/415] ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load() Use for_each_pcm_streams() to enumerate streams in sof_dai_load() instead of doing that manually. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20200325211233.27394-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 058de94fb8cf63..54437caf9488cc 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2448,7 +2448,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_stream_caps *caps; struct snd_soc_tplg_private *private = &pcm->priv; struct snd_sof_pcm *spcm; - int stream = SNDRV_PCM_STREAM_PLAYBACK; + int stream; int ret = 0; /* nothing to do for BEs atm */ @@ -2460,8 +2460,9 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, return -ENOMEM; spcm->scomp = scomp; - spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id = COMP_ID_UNASSIGNED; - spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id = COMP_ID_UNASSIGNED; + + for_each_pcm_streams(stream) + spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED; spcm->pcm = *pcm; dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name); @@ -2482,8 +2483,10 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, if (!spcm->pcm.playback) goto capture; + stream = SNDRV_PCM_STREAM_PLAYBACK; + dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: playback d0i3:%d\n", - spcm->pcm.pcm_name, spcm->stream[0].d0i3_compatible); + spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible); caps = &spcm->pcm.caps[stream]; @@ -2513,7 +2516,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, return ret; dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: capture d0i3:%d\n", - spcm->pcm.pcm_name, spcm->stream[1].d0i3_compatible); + spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible); caps = &spcm->pcm.caps[stream]; From 9ef91cad92ba75d17d5a5203230746c9d9009705 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 25 Mar 2020 16:12:31 -0500 Subject: [PATCH 349/415] ASoC: SOF: fix uninitialised "work" with VirtIO In the VirtIO case the sof_pcm_open() function isn't called on the host during guest streaming, which then leaves "work" structures uninitialised. However it is then used to handle position update messages from the DSP. Move their initialisation to immediately after allocation of the containing structure. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20200325211233.27394-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 4 +--- sound/soc/sof/sof-audio.h | 3 +++ sound/soc/sof/topology.c | 6 +++++- 3 files changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index f4769e19965a55..47cd741f2a8c74 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -57,7 +57,7 @@ static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream /* * sof pcm period elapse work */ -static void sof_pcm_period_elapsed_work(struct work_struct *work) +void snd_sof_pcm_period_elapsed_work(struct work_struct *work) { struct snd_sof_pcm_stream *sps = container_of(work, struct snd_sof_pcm_stream, @@ -475,8 +475,6 @@ static int sof_pcm_open(struct snd_soc_component *component, dev_dbg(component->dev, "pcm: open stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); - INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work, - sof_pcm_period_elapsed_work); caps = &spcm->pcm.caps[substream->stream]; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index eacd10e4da11a3..bf65f31af85821 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -11,6 +11,8 @@ #ifndef __SOUND_SOC_SOF_AUDIO_H #define __SOUND_SOC_SOF_AUDIO_H +#include + #include #include #include /* needs to be included before control.h */ @@ -189,6 +191,7 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp, struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_soc_component *scomp, unsigned int pcm_id); void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream); +void snd_sof_pcm_period_elapsed_work(struct work_struct *work); /* * Mixer IPC diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 54437caf9488cc..fe8ba3e05e08bd 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -9,6 +9,7 @@ // #include +#include #include #include #include @@ -2461,8 +2462,11 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, spcm->scomp = scomp; - for_each_pcm_streams(stream) + for_each_pcm_streams(stream) { spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED; + INIT_WORK(&spcm->stream[stream].period_elapsed_work, + snd_sof_pcm_period_elapsed_work); + } spcm->pcm = *pcm; dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name); From aae5a6e92f3f3411fdf6abcf41ecadd771abaa4b Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Wed, 25 Mar 2020 16:12:32 -0500 Subject: [PATCH 350/415] ASoC: SOF: Intel: hda: do not leave clock gating off upon error The misc clock gating (MISCBDCGE) is disabled for controller reset and reenabled once reset is complete. Fix the case when error happens during reset, and clock gating is left disabled. The clock gating should be reenabled also in this case. Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20200325211233.27394-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ctrl.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 871b71a15a6331..93be6fc51ccd7c 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -183,7 +183,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) ret = hda_dsp_ctrl_link_reset(sdev, true); if (ret < 0) { dev_err(sdev->dev, "error: failed to reset HDA controller\n"); - return ret; + goto err; } usleep_range(500, 1000); @@ -192,7 +192,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) ret = hda_dsp_ctrl_link_reset(sdev, false); if (ret < 0) { dev_err(sdev->dev, "error: failed to exit HDA controller reset\n"); - return ret; + goto err; } usleep_range(1000, 1200); @@ -202,7 +202,8 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) /* check to see if controller is ready */ if (!snd_hdac_chip_readb(bus, GCTL)) { dev_dbg(bus->dev, "controller not ready!\n"); - return -EBUSY; + ret = -EBUSY; + goto err; } /* Accept unsolicited responses */ @@ -268,6 +269,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) bus->chip_init = true; +err: hda_dsp_ctrl_misc_clock_gating(sdev, true); return ret; From 7e26df0ced1643679922d197e798d469ac3bf9c0 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Wed, 25 Mar 2020 16:12:33 -0500 Subject: [PATCH 351/415] ASoC: SOF: Intel: hda: call codec wake at chip init Further align HDA init sequence to the legacy non-DSP HDA driver by calling snd_hdac_set_codec_wakeup() during the chip init sequence. Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20200325211233.27394-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ctrl.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 93be6fc51ccd7c..f88dbcc4ba66a6 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -18,6 +18,7 @@ #include #include #include +#include #include "../ops.h" #include "hda.h" @@ -176,6 +177,9 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) if (bus->chip_init) return 0; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + snd_hdac_set_codec_wakeup(bus, true); +#endif hda_dsp_ctrl_misc_clock_gating(sdev, false); if (full_reset) { @@ -271,6 +275,9 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) err: hda_dsp_ctrl_misc_clock_gating(sdev, true); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + snd_hdac_set_codec_wakeup(bus, false); +#endif return ret; } From f7cc9b996e7417708b8168697762e4bc97fa6696 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:29:04 -0500 Subject: [PATCH 352/415] ASoC: rt1308-sdw: add set_tdm_slot() support Add ability to select which of the channels is used, or both, in case two RT1308 amplifiers are located on the same link. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325212905.28145-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1308-sdw.c | 23 +++++++++++++++++++++++ sound/soc/codecs/rt1308-sdw.h | 2 ++ 2 files changed, 25 insertions(+) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index d930f60cb79726..8763192434c480 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -507,6 +507,28 @@ static void rt1308_sdw_shutdown(struct snd_pcm_substream *substream, kfree(stream); } +static int rt1308_sdw_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct rt1308_sdw_priv *rt1308 = + snd_soc_component_get_drvdata(component); + + if (tx_mask) + return -EINVAL; + + if (slots > 2) + return -EINVAL; + + rt1308->rx_mask = rx_mask; + rt1308->slots = slots; + /* slot_width is not used since it's irrelevant for SoundWire */ + + return 0; +} + static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -597,6 +619,7 @@ static const struct snd_soc_dai_ops rt1308_aif_dai_ops = { .hw_free = rt1308_sdw_pcm_hw_free, .set_sdw_stream = rt1308_set_sdw_stream, .shutdown = rt1308_sdw_shutdown, + .set_tdm_slot = rt1308_sdw_set_tdm_slot, }; #define RT1308_STEREO_RATES SNDRV_PCM_RATE_48000 diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index c9341e70d6cf56..c5ce75666dcc86 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -160,6 +160,8 @@ struct rt1308_sdw_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + int rx_mask; + int slots; }; struct sdw_stream_data { From 27a18e9e673f03b48a881e133a532c1619b711c7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:29:05 -0500 Subject: [PATCH 353/415] ASoC: rt1308-sdw: use slot and rx_mask to configure stream If the DAI was configured with a set_tdm_slots() call, use the information. A platform or machine driver may configure each amplifier to extract different bitSlots from the frame, or extract the same data and use processing to generate the relevant output. The latter case is easier to handle in case of orientation changes. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325212905.28145-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1308-sdw.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 8763192434c480..a5a7e46de246a4 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -539,7 +539,7 @@ static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, struct sdw_port_config port_config; enum sdw_data_direction direction; struct sdw_stream_data *stream; - int retval, port, num_channels; + int retval, port, num_channels, ch_mask; dev_dbg(dai->dev, "%s %s", __func__, dai->name); stream = snd_soc_dai_get_dma_data(dai, substream); @@ -559,13 +559,20 @@ static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + if (rt1308->slots) { + num_channels = rt1308->slots; + ch_mask = rt1308->rx_mask; + } else { + num_channels = params_channels(params); + ch_mask = (1 << num_channels) - 1; + } + stream_config.frame_rate = params_rate(params); - stream_config.ch_count = params_channels(params); + stream_config.ch_count = num_channels; stream_config.bps = snd_pcm_format_width(params_format(params)); stream_config.direction = direction; - num_channels = params_channels(params); - port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.ch_mask = ch_mask; port_config.num = port; retval = sdw_stream_add_slave(rt1308->sdw_slave, &stream_config, From 60a260169abd78a71d1fa3f0384fbe78aab3ffdb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Mar 2020 15:10:53 +0000 Subject: [PATCH 354/415] ASoC: pxa: Select regmap from AC'97 machines regmap needs to be selected by users which for machine drivers that select AC'97 CODEC drivers means that we need to also select regmap to ensure that the CODEC driver will build if nothing else enables regmap as is likely for such systems. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20200326151053.40806-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 295cfffa4646ea..1f0c08b06c1dd1 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -81,6 +81,7 @@ config SND_PXA2XX_SOC_TOSA depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB depends on AC97_BUS=n + select REGMAP select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -91,6 +92,7 @@ config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 depends on AC97_BUS=n + select REGMAP select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -101,6 +103,7 @@ config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 depends on AC97_BUS=n + select REGMAP select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -111,6 +114,7 @@ config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 depends on AC97_BUS=n + select REGMAP select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -122,6 +126,7 @@ config SND_PXA2XX_SOC_EM_X270 depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) depends on AC97_BUS=n + select REGMAP select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -133,6 +138,7 @@ config SND_PXA2XX_SOC_PALM27X depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) depends on AC97_BUS=n + select REGMAP select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -164,6 +170,7 @@ config SND_SOC_ZYLONITE depends on SND_PXA2XX_SOC && MACH_ZYLONITE depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 + select REGMAP select SND_PXA_SOC_SSP select SND_SOC_WM9713 help @@ -193,6 +200,7 @@ config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n + select REGMAP select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help From 27821f4ddedcaea0f16d03ee3432bddb729daba5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Mar 2020 18:01:16 +0000 Subject: [PATCH 355/415] ASoC: pxa: Enable AC'97 bus support for PXA machines The AC'97 based PXA machines currently don't build reliably as they don't ensure that an AC'97 bus is built, causing at least eseries_pxa_defconfig to fail to build. Add selects to fix this. Reported-by: KernelCI Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20200326180116.21375-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 1f0c08b06c1dd1..d4c0f580a565e6 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -82,6 +82,8 @@ config SND_PXA2XX_SOC_TOSA depends on MFD_TC6393XB depends on AC97_BUS=n select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -93,6 +95,8 @@ config SND_PXA2XX_SOC_E740 depends on SND_PXA2XX_SOC && MACH_E740 depends on AC97_BUS=n select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -116,6 +120,8 @@ config SND_PXA2XX_SOC_E800 depends on AC97_BUS=n select REGMAP select SND_SOC_WM9712 + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 help Say Y if you want to add support for SoC audio on the @@ -127,6 +133,8 @@ config SND_PXA2XX_SOC_EM_X270 MACH_CM_X300) depends on AC97_BUS=n select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -139,6 +147,8 @@ config SND_PXA2XX_SOC_PALM27X MACH_PALMT5 || MACH_PALMTE2) depends on AC97_BUS=n select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -169,6 +179,8 @@ config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE depends on AC97_BUS=n + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select REGMAP select SND_PXA_SOC_SSP @@ -201,6 +213,8 @@ config SND_PXA2XX_SOC_MIOA701 depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help From f5a88b0accc24c4a9021247d7a3124f90aa4c586 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Fri, 27 Mar 2020 12:46:25 +0800 Subject: [PATCH 356/415] ALSA: hda/realtek: Enable mute LED on an HP system The system in question uses ALC285, and it uses GPIO 0x04 to control its mute LED. The mic mute LED can be controlled by GPIO 0x01, however the system uses DMIC so we should use that to control mic mute LED. Signed-off-by: Kai-Heng Feng Cc: Link: https://lore.kernel.org/r/20200327044626.29582-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63e1a56f705b7d..1ad8c2e2d1aff2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4008,6 +4008,12 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10); } +static void alc285_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0x04, 0x00); +} + static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5923,6 +5929,7 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC285_FIXUP_HP_GPIO_LED, }; static const struct hda_fixup alc269_fixups[] = { @@ -7061,6 +7068,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC285_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_gpio_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7208,6 +7219,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), From 04a9af2e03842347269fe1b3e64b94b8226fbba1 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Thu, 26 Mar 2020 22:10:12 +0100 Subject: [PATCH 357/415] ALSA: ppc: keywest: convert to use i2c_new_client_device() Move away from the deprecated API and return the shiny new ERRPTR where useful. Signed-off-by: Wolfram Sang Link: https://lore.kernel.org/r/20200326211013.13531-2-wsa+renesas@sang-engineering.com Signed-off-by: Takashi Iwai --- sound/ppc/keywest.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 093806d735c618..9554a0c506afbc 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -40,6 +40,7 @@ static int keywest_probe(struct i2c_client *client, static int keywest_attach_adapter(struct i2c_adapter *adapter) { struct i2c_board_info info; + struct i2c_client *client; if (! keywest_ctx) return -EINVAL; @@ -50,9 +51,11 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) memset(&info, 0, sizeof(struct i2c_board_info)); strlcpy(info.type, "keywest", I2C_NAME_SIZE); info.addr = keywest_ctx->addr; - keywest_ctx->client = i2c_new_device(adapter, &info); - if (!keywest_ctx->client) - return -ENODEV; + client = i2c_new_client_device(adapter, &info); + if (IS_ERR(client)) + return PTR_ERR(client); + keywest_ctx->client = client; + /* * We know the driver is already loaded, so the device should be * already bound. If not it means binding failed, and then there From 2af69581e1b11152bc42f1122d3c16e177bd77cf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:16:57 +0900 Subject: [PATCH 358/415] ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro Now, snd_soc_pcm_runtime supports multi cpu_dai/codec_dai. It still has cpu_dai/codec_dai for single DAI, and has cpu_dais/codec_dais for multi DAIs. dais = [][][][][][][][][][][][][][][][][][] ^cpu_dais ^codec_dais |--- num_cpus ---|--- num_codecs --| /* for multi DAIs */ rtd->cpu_dais = &rtd->dais[0]; rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; /* for single DAI */ rtd->cpu_dai = rtd->cpu_dais[0]; rtd->codec_dai = rtd->codec_dais[0]; But, these can be replaced by dais. This patch adds asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for it. Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/875zevk5va.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 539211bd0f9493..13458e4fbb132e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1169,6 +1169,10 @@ struct snd_soc_pcm_runtime { int num_components; struct snd_soc_component *components[0]; /* CPU/Codec/Platform */ }; +/* see soc_new_pcm_runtime() */ +#define asoc_rtd_to_cpu(rtd, n) (rtd)->dais[n] +#define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus] + #define for_each_rtd_components(rtd, i, component) \ for ((i) = 0; \ ((i) < rtd->num_components) && ((component) = rtd->components[i]);\ From b09b22fcf9fb8a3186ad0e09aedfa9e119520c43 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:17:13 +0900 Subject: [PATCH 359/415] ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/874kufk5uu.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 ++-- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +++--- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 7a5621e5e2330d..9414d7269c4f58 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -54,7 +54,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index 91abeb92b648e6..73b31f88a6b566 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -48,7 +48,7 @@ static int cz_aif1_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, CZ_PLAT_CLK, params_rate(params) * 512); @@ -73,7 +73,7 @@ static int cz_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card; struct snd_soc_component *codec; - codec = rtd->codec_dai->component; + codec = asoc_rtd_to_codec(rtd, 0)->component; card = rtd->card; ret = snd_soc_card_jack_new(card, "Headset Jack", diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 8f71c3f7ef792e..024a7ee54cd5a6 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -35,7 +35,7 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -183,7 +183,7 @@ static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); machine->cap_i2s_instance = I2S_BT_INSTANCE; @@ -198,7 +198,7 @@ static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); machine->cap_i2s_instance = I2S_BT_INSTANCE; From b434d70788815a4fb2de3051643ccf49cc38db42 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:17:22 +0900 Subject: [PATCH 360/415] ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87369zk5ul.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm-dma.c | 4 ++-- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- 6 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index db67f5ba1e9a32..cb03c4f7324c9c 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -56,7 +56,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pcm_dma_params *prtd; - prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (ssc_sr & prtd->mask->ssc_error) { if (snd_pcm_running(substream)) @@ -83,7 +83,7 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, struct ssc_device *ssc; int ret; - prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); ssc = prtd->ssc; ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 59c1331a698407..a8daebcbf6c818 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -213,7 +213,7 @@ static int atmel_pcm_hw_params(struct snd_soc_component *component, snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); - prtd->params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + prtd->params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); prtd->params->dma_intr_handler = atmel_pcm_dma_irq; prtd->dma_buffer = runtime->dma_addr; diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 776b27d3686e72..148c943cb53843 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -27,7 +27,7 @@ static int atmel_asoc_wm8904_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK, diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index aa6d0d78566f80..f9a85fd01b79e2 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -21,7 +21,7 @@ static int snd_proto_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* Set proto sysclk */ int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index b1bef2bf142dc1..ed1f69b570244f 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -96,7 +96,7 @@ static const struct snd_soc_dapm_route intercon[] = { */ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct device *dev = rtd->dev; int ret; diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 7822425d5e6169..9fbc3c1113cc5e 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -40,7 +40,7 @@ struct sam9x5_drvdata { */ static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct device *dev = rtd->dev; int ret; From 11a828fa8b426a949ce2759e30399695b58114a2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:17:47 +0900 Subject: [PATCH 361/415] ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/871rpjk5tw.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index d6b692fff29a29..d649037bda9b8e 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -95,7 +95,7 @@ static struct snd_soc_card db1550_ac97_machine = { static int db1200_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* WM8731 has its own 12MHz crystal */ snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 8f855644c6b4f3..e82bbf2d1eea78 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -281,7 +281,7 @@ static int au1xpsc_pcm_open(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = substream->private_data; int stype = substream->stream, *dmaids; - dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dmaids) return -ENODEV; /* whoa, has ordering changed? */ diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index c9a038a5e2d34d..4e246c7e78f247 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -195,7 +195,7 @@ static int alchemy_pcm_open(struct snd_soc_component *component, int *dmaids, s = substream->stream; char *name; - dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dmaids) return -ENODEV; /* whoa, has ordering changed? */ diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 0227993c5da8ab..05eb36991f147a 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -58,7 +58,7 @@ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x) { struct snd_soc_card *c = x->bus->card->private_data; - return snd_soc_dai_get_drvdata(c->rtd->cpu_dai); + return snd_soc_dai_get_drvdata(c->asoc_rtd_to_cpu(rtd, 0)); } #else From fc3923644867e4bf2ccd3b9a1daca269c0bc8ff1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:17:59 +0900 Subject: [PATCH 362/415] ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87zhc7ir94.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/bcm/bcm63xx-pcm-whistler.c | 16 ++++++++-------- sound/soc/bcm/cygnus-pcm.c | 22 +++++++++++----------- 2 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index 55c760f1cf4da2..e46c390683e785 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -55,7 +55,7 @@ static int bcm63xx_pcm_hw_params(struct snd_soc_component *component, if (!dma_desc) return -ENOMEM; - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_desc); + snd_soc_dai_set_dma_data(asoc_rtd_to_cpu(rtd, 0), substream, dma_desc); return 0; } @@ -66,7 +66,7 @@ static int bcm63xx_pcm_hw_free(struct snd_soc_component *component, struct i2s_dma_desc *dma_desc; struct snd_soc_pcm_runtime *rtd = substream->private_data; - dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); kfree(dma_desc); snd_pcm_set_runtime_buffer(substream, NULL); @@ -82,7 +82,7 @@ static int bcm63xx_pcm_trigger(struct snd_soc_component *component, struct regmap *regmap_i2s; rtd = substream->private_data; - i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev); regmap_i2s = i2s_priv->regmap_i2s; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -152,7 +152,7 @@ static int bcm63xx_pcm_prepare(struct snd_soc_component *component, struct snd_pcm_runtime *runtime = substream->runtime; uint32_t regaddr_desclen, regaddr_descaddr; - dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); dma_desc->dma_len = snd_pcm_lib_period_bytes(substream); dma_desc->dma_addr = runtime->dma_addr; dma_desc->dma_area = runtime->dma_area; @@ -165,7 +165,7 @@ static int bcm63xx_pcm_prepare(struct snd_soc_component *component, regaddr_descaddr = I2S_RX_DESC_IFF_ADDR; } - i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev); regmap_i2s = i2s_priv->regmap_i2s; regmap_write(regmap_i2s, regaddr_desclen, dma_desc->dma_len); @@ -269,7 +269,7 @@ static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv) runtime = substream->runtime; rtd = substream->private_data; prtd = runtime->private_data; - dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); offlevel = (int_status & I2S_RX_DESC_OFF_LEVEL_MASK) >> I2S_RX_DESC_OFF_LEVEL_SHIFT; @@ -317,7 +317,7 @@ static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv) runtime = substream->runtime; rtd = substream->private_data; prtd = runtime->private_data; - dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); offlevel = (int_status & I2S_TX_DESC_OFF_LEVEL_MASK) >> I2S_TX_DESC_OFF_LEVEL_SHIFT; @@ -388,7 +388,7 @@ static int bcm63xx_soc_pcm_new(struct snd_soc_component *component, struct bcm_i2s_priv *i2s_priv; int ret; - i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev); of_dma_configure(pcm->card->dev, pcm->card->dev->of_node, 1); diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c index 3a80c613bc3f9f..f96d27c8b30182 100644 --- a/sound/soc/bcm/cygnus-pcm.c +++ b/sound/soc/bcm/cygnus-pcm.c @@ -209,7 +209,7 @@ static struct cygnus_aio_port *cygnus_dai_get_dma_data( { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - return snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); + return snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(soc_runtime, 0), substream); } static void ringbuf_set_initial(void __iomem *audio_io, @@ -359,7 +359,7 @@ static void disable_intr(struct snd_pcm_substream *substream) aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s on port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s on port %d\n", __func__, aio->portnum); /* The port number maps to the bit position to be set */ set_mask = BIT(aio->portnum); @@ -590,7 +590,7 @@ static int cygnus_pcm_open(struct snd_soc_component *component, if (!aio) return -ENODEV; - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw); @@ -623,7 +623,7 @@ static int cygnus_pcm_close(struct snd_soc_component *component, aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aio->play_stream = NULL; @@ -631,7 +631,7 @@ static int cygnus_pcm_close(struct snd_soc_component *component, aio->capture_stream = NULL; if (!aio->play_stream && !aio->capture_stream) - dev_dbg(rtd->cpu_dai->dev, "freed port %d\n", aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "freed port %d\n", aio->portnum); return 0; } @@ -645,7 +645,7 @@ static int cygnus_pcm_hw_params(struct snd_soc_component *component, struct cygnus_aio_port *aio; aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); @@ -660,7 +660,7 @@ static int cygnus_pcm_hw_free(struct snd_soc_component *component, struct cygnus_aio_port *aio; aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -678,12 +678,12 @@ static int cygnus_pcm_prepare(struct snd_soc_component *component, struct ringbuf_regs *p_rbuf = NULL; aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); bufsize = snd_pcm_lib_buffer_bytes(substream); periodsize = snd_pcm_lib_period_bytes(substream); - dev_dbg(rtd->cpu_dai->dev, "%s (buf_size %lu) (period_size %lu)\n", + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s (buf_size %lu) (period_size %lu)\n", __func__, bufsize, periodsize); configure_ringbuf_regs(substream); @@ -745,11 +745,11 @@ static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%zx @ %pK\n", + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: size 0x%zx @ %pK\n", __func__, size, buf->area); if (!buf->area) { - dev_err(rtd->cpu_dai->dev, "%s: dma_alloc failed\n", __func__); + dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: dma_alloc failed\n", __func__); return -ENOMEM; } buf->bytes = size; From 07c497a621c50c28ade05891dcb0333a501e6a21 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:18:11 +0900 Subject: [PATCH 363/415] ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87y2rrir8s.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/cirrus/edb93xx.c | 4 ++-- sound/soc/cirrus/snappercl15.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 10961190068e87..ccf65f087ea618 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -23,8 +23,8 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int err; unsigned int mclk_rate; unsigned int rate = params_rate(params); diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 70c2f3e08d6d40..cb133e80b7c34a 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -23,8 +23,8 @@ static int snappercl15_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int err; err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, From e42b2047cd56236d3811f900db3fbaf7f88c7065 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:18:20 +0900 Subject: [PATCH 364/415] ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87wo7bir8j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/dwc/dwc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/dwc-pcm.c b/sound/soc/dwc/dwc-pcm.c index 4b25aca3804f6f..9868e7373d369a 100644 --- a/sound/soc/dwc/dwc-pcm.c +++ b/sound/soc/dwc/dwc-pcm.c @@ -140,7 +140,7 @@ static int dw_pcm_open(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); snd_soc_set_runtime_hwparams(substream, &dw_pcm_hardware); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); From 17198ae76e0ffcb891d34f59ad3725d9c536ac99 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:18:30 +0900 Subject: [PATCH 365/415] ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87v9mvir89.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 4 ++-- sound/soc/fsl/fsl-asoc-card.c | 10 +++++----- sound/soc/fsl/fsl_asrc_dma.c | 6 +++--- sound/soc/fsl/fsl_spdif.c | 10 +++++----- sound/soc/fsl/fsl_ssi.c | 8 ++++---- sound/soc/fsl/imx-audmix.c | 8 ++++---- sound/soc/fsl/imx-mc13783.c | 4 ++-- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 10 +++++----- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 ++-- sound/soc/fsl/mx27vis-aic32x4.c | 4 ++-- sound/soc/fsl/p1022_ds.c | 4 ++-- sound/soc/fsl/p1022_rdk.c | 4 ++-- sound/soc/fsl/wm1133-ev1.c | 6 +++--- 15 files changed, 43 insertions(+), 43 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 6f3b768489f6f1..4ff2d21bb32fe2 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -31,8 +31,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, 0, diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 9ce55feaac22d0..bb33601fab844a 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -159,7 +159,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, return 0; /* Specific configurations of DAIs starts from here */ - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx], cpu_priv->sysclk_freq[tx], cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { @@ -168,7 +168,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } if (cpu_priv->slot_width) { - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); @@ -257,7 +257,7 @@ static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -446,14 +446,14 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); struct snd_soc_pcm_runtime *rtd = list_first_entry( &card->rtd_list, struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = card->dev; int ret; if (fsl_asoc_card_is_ac97(priv)) { #if IS_ENABLED(CONFIG_SND_AC97_CODEC) - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); /* diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 44e5924be87056..e7178817d7a756 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -152,7 +152,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, for_each_dpcm_be(rtd, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *substream_be; - struct snd_soc_dai *dai = be->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(be, 0); if (dpcm->fe != rtd) continue; @@ -169,7 +169,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, } /* Override dma_data of the Front-End and config its dmaengine */ - dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params_fe = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index); dma_params_fe->maxburst = dma_params_be->maxburst; @@ -328,7 +328,7 @@ static int fsl_asrc_dma_startup(struct snd_soc_component *component, goto dma_chan_err; } - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); /* Refine the snd_imx_hardware according to caps of DMA. */ ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 7858a5499ac5d0..c711d2d9328094 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -370,7 +370,7 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, int sample_rate) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; struct regmap *regmap = spdif_priv->regmap; struct platform_device *pdev = spdif_priv->pdev; @@ -458,7 +458,7 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct platform_device *pdev = spdif_priv->pdev; struct regmap *regmap = spdif_priv->regmap; u32 scr, mask; @@ -534,7 +534,7 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct regmap *regmap = spdif_priv->regmap; u32 scr, mask, i; @@ -569,7 +569,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; struct platform_device *pdev = spdif_priv->pdev; u32 sample_rate = params_rate(params); @@ -597,7 +597,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct regmap *regmap = spdif_priv->regmap; bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 intr = SIE_INTR_FOR(tx); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 5c97269be34604..bad89b0d129e7f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -631,7 +631,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); int ret; ret = clk_prepare_enable(ssi->clk); @@ -655,7 +655,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); clk_disable_unprepare(ssi->clk); } @@ -854,7 +854,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); if (fsl_ssi_is_i2s_master(ssi) && ssi->baudclk_streams & BIT(substream->stream)) { @@ -1059,7 +1059,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; switch (cmd) { diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 5ef6881395e0f4..e09b45de0efd4b 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -85,13 +85,13 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; /* set DAI configuration */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret) { dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); return ret; } - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir); + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), FSL_SAI_CLK_MAST1, 0, dir); if (ret) { dev_err(dev, "failed to set cpu sysclk: %d\n", ret); return ret; @@ -101,7 +101,7 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, * Per datasheet, AUDMIX expects 8 slots and 32 bits * for every slot in TDM mode. */ - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1, + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), BIT(channels) - 1, BIT(channels) - 1, 8, 32); if (ret) dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); @@ -125,7 +125,7 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, fmt |= SND_SOC_DAIFMT_CBM_CFM; /* set AUDMIX DAI configuration */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret) dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret); diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 2b679680c93f1d..fab2d6c56653ec 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -27,8 +27,8 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 4, 16); diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 15e8b9343c3544..f45cb4bbb6c4d1 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -30,7 +30,7 @@ static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) struct device *dev = rtd->card->dev; int ret; - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), SGTL5000_SYSCLK, data->clk_frequency, SND_SOC_CLOCK_IN); if (ret) { dev_err(dev, "could not set codec driver clock params\n"); diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index ed7211d744b3ab..3b8c796d7829f5 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -115,7 +115,7 @@ static int psc_dma_trigger(struct snd_soc_component *component, struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct snd_pcm_runtime *runtime = substream->runtime; struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; @@ -217,7 +217,7 @@ static int psc_dma_open(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct psc_dma_stream *s; int rc; @@ -245,7 +245,7 @@ static int psc_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct psc_dma_stream *s; dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream); @@ -271,7 +271,7 @@ psc_dma_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct psc_dma_stream *s; dma_addr_t count; @@ -298,7 +298,7 @@ static int psc_dma_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); struct snd_pcm *pcm = rtd->pcm; size_t size = psc_dma_hardware.buffer_bytes_max; int rc; diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 9bc01f374b39a5..1ab4fbda08cb02 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -39,7 +39,7 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); u32 mode; dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 23617eb09ba1bc..f7bd90051ce70b 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -105,7 +105,7 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, machine_data->dai_format); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), machine_data->dai_format); if (ret < 0) { dev_err(dev, "could not set codec driver audio format\n"); return ret; @@ -115,7 +115,7 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 0, machine_data->clk_frequency, machine_data->codec_clk_direction); if (ret < 0) { diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index 38ac4a397742a8..a36d4e8cd55cc3 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -37,8 +37,8 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, 0, diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 6114b01b90f7b8..fe3091590f2075 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -128,7 +128,7 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format); if (ret < 0) { dev_err(dev, "could not set codec driver audio format\n"); return ret; @@ -138,7 +138,7 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream) * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 0, mdata->clk_frequency, mdata->codec_clk_direction); if (ret < 0) { dev_err(dev, "could not set codec driver clock params\n"); diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index 72687235c0ae37..f5374fe354abc5 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -134,14 +134,14 @@ static int p1022_rdk_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format); if (ret < 0) { dev_err(dev, "could not set codec driver audio format (ret=%i)\n", ret); return ret; } - ret = snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, mdata->clk_frequency, + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0, mdata->clk_frequency, mdata->clk_frequency); if (ret < 0) { dev_err(dev, "could not set codec PLL frequency (ret=%i)\n", diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 52d321bede9c15..8b1551c55452bc 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -76,8 +76,8 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, found = 0; snd_pcm_format_t format = params_format(params); unsigned int rate = params_rate(params); @@ -196,7 +196,7 @@ static struct snd_soc_jack_pin mic_jack_pins[] = { static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* Headphone jack detection */ snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, From e7718a72653603722e1b0e06f6a4e76bcafdc431 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:18:41 +0900 Subject: [PATCH 366/415] ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87tv2fir7y.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index abbdf1054f6fa2..8c54dc6710fe56 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -213,8 +213,8 @@ EXPORT_SYMBOL_GPL(asoc_simple_startup); void asoc_simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); @@ -249,8 +249,8 @@ int asoc_simple_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); @@ -381,12 +381,12 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_init_dai(rtd->codec_dai, + ret = asoc_simple_init_dai(asoc_rtd_to_codec(rtd, 0), dai_props->codec_dai); if (ret < 0) return ret; - ret = asoc_simple_init_dai(rtd->cpu_dai, + ret = asoc_simple_init_dai(asoc_rtd_to_cpu(rtd, 0), dai_props->cpu_dai); if (ret < 0) return ret; From 4d3801d5f49d6d06cabad7afad97d3a155de4a84 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:18:53 +0900 Subject: [PATCH 367/415] ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87sghzir7m.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index fdd2c73fd2fac0..a495d1050d490f 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -397,7 +397,7 @@ static int img_i2s_in_dma_prepare_slave_config(struct snd_pcm_substream *st, struct snd_dmaengine_dai_dma_data *dma_data; int ret; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, st); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), st); ret = snd_hwparams_to_dma_slave_config(st, params, sc); if (ret) diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 4b18534096336b..db052ec17d5d8c 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -403,7 +403,7 @@ static int img_i2s_out_dma_prepare_slave_config(struct snd_pcm_substream *st, struct snd_dmaengine_dai_dma_data *dma_data; int ret; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, st); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), st); ret = snd_hwparams_to_dma_slave_config(st, params, sc); if (ret) From 0d1571c197a920eb8567e9376da04f4fb7965f23 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:19:05 +0900 Subject: [PATCH 368/415] ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87r1xjir7a.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 ++--- sound/soc/intel/boards/bdw-rt5650.c | 6 ++--- sound/soc/intel/boards/bdw-rt5677.c | 6 ++--- sound/soc/intel/boards/broadwell.c | 4 +-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 8 +++--- sound/soc/intel/boards/bxt_rt298.c | 8 +++--- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 4 +-- sound/soc/intel/boards/bytcht_cx2072x.c | 10 +++---- sound/soc/intel/boards/bytcht_da7213.c | 8 +++--- sound/soc/intel/boards/bytcht_es8316.c | 8 +++--- sound/soc/intel/boards/bytcht_nocodec.c | 4 +-- sound/soc/intel/boards/bytcr_rt5640.c | 8 +++--- sound/soc/intel/boards/bytcr_rt5651.c | 8 +++--- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 ++--- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 +-- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +++++----- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 +++--- sound/soc/intel/boards/cml_rt1011_rt5682.c | 6 ++--- sound/soc/intel/boards/glk_rt5682_max98357a.c | 10 +++---- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 8 +++--- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 ++--- sound/soc/intel/boards/kbl_rt5660.c | 6 ++--- sound/soc/intel/boards/kbl_rt5663_max98927.c | 8 +++--- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 8 +++--- .../soc/intel/boards/skl_nau88l25_max98357a.c | 12 ++++----- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 ++++++------ sound/soc/intel/boards/skl_rt286.c | 8 +++--- sound/soc/intel/boards/sof_da7219_max98373.c | 8 +++--- sound/soc/intel/boards/sof_pcm512x.c | 8 +++--- sound/soc/intel/boards/sof_rt5682.c | 6 ++--- sound/soc/intel/haswell/sst-haswell-pcm.c | 26 +++++++++---------- sound/soc/intel/skylake/skl-pcm.c | 10 +++---- 34 files changed, 135 insertions(+), 135 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 340bd2be39a7e4..82f2b6357778d7 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -649,7 +649,7 @@ static snd_pcm_uframes_t sst_soc_pointer(struct snd_soc_component *component, static int sst_soc_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); struct snd_pcm *pcm = rtd->pcm; if (dai->driver->playback.channels_min || @@ -741,7 +741,7 @@ static int sst_soc_prepare(struct device *dev) /* set the SSPs to idle */ for_each_card_rtds(drv->soc_card, rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); if (dai->active) { send_ssp_cmd(dai, dai->name, 0); @@ -762,7 +762,7 @@ static void sst_soc_complete(struct device *dev) /* restart SSPs */ for_each_card_rtds(drv->soc_card, rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); if (dai->active) { sst_handle_vb_timer(dai, true); diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index 058abf3eec501f..53ca3d8da5de20 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -107,7 +107,7 @@ static int bdw_rt5650_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* Workaround: set codec PLL to 19.2MHz that PLL source is @@ -166,8 +166,8 @@ static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd) { struct bdw_rt5650_priv *bdw_rt5650 = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; int ret; /* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1. diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index a94f498388e145..06a5396082dd02 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -157,7 +157,7 @@ static int bdw_rt5677_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, 24576000, @@ -174,7 +174,7 @@ static int bdw_rt5677_dsp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_PLL1, 24576000, @@ -226,7 +226,7 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) { struct bdw_rt5677_priv *bdw_rt5677 = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); int ret; diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 25178000c6a55c..8318914227e5ae 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -70,7 +70,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret = 0; ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, @@ -104,7 +104,7 @@ static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 061462248bce02..44016c16f25e2e 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -179,8 +179,8 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int clk_freq; /* Configure sysclk for codec */ @@ -226,7 +226,7 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct bxt_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -244,7 +244,7 @@ static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int broxton_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 4b5e7f6dbdf1cf..7a4decf341918e 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -155,7 +155,7 @@ static const struct snd_soc_dapm_route geminilake_rt298_map[] = { static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -165,7 +165,7 @@ static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd) static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret = 0; ret = snd_soc_card_jack_new(rtd->card, "Headset", @@ -186,7 +186,7 @@ static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct bxt_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -225,7 +225,7 @@ static int broxton_rt298_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 01739ad75b12bf..f5097da2882850 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -89,7 +89,7 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) card->dapm.idle_bias_off = true; - ret = snd_soc_dai_set_sysclk(runtime->codec_dai, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(runtime, 0), M98090_REG_SYSTEM_CLOCK, 25000000, SND_SOC_CLOCK_IN); if (ret < 0) { diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index 0c76dafdd57266..ace232f8aed659 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -73,7 +73,7 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, @@ -123,7 +123,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; struct snd_soc_card *card = runtime->card; const struct snd_soc_dapm_route *custom_map; int num_routes; diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 67f06c95eec531..3b3df7c9008c7d 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -70,7 +70,7 @@ static const struct acpi_gpio_mapping byt_cht_cx2072x_acpi_gpios[] = { static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct snd_soc_component *codec = rtd->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; int ret; if (devm_acpi_dev_add_driver_gpios(codec->dev, @@ -80,7 +80,7 @@ static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd) card->dapm.idle_bias_off = true; /* set the default PLL rate, the clock is handled by the codec driver */ - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, CX2072X_MCLK_EXTERNAL_PLL, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), CX2072X_MCLK_EXTERNAL_PLL, 19200000, SND_SOC_CLOCK_IN); if (ret) { dev_err(rtd->dev, "Could not set sysclk\n"); @@ -97,7 +97,7 @@ static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_component_set_jack(codec, &byt_cht_cx2072x_headset, NULL); - snd_soc_dai_set_bclk_ratio(rtd->codec_dai, 50); + snd_soc_dai_set_bclk_ratio(asoc_rtd_to_codec(rtd, 0), 50); return ret; } @@ -123,7 +123,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -132,7 +132,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index d6b912c013fc56..5e96e7d02733c1 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -78,7 +78,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -87,7 +87,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; @@ -106,7 +106,7 @@ static int aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK, @@ -127,7 +127,7 @@ static int aif1_hw_params(struct snd_pcm_substream *substream, static int aif1_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_pll(codec_dai, 0, diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 0adc5a5e134a8a..ddcd070100ef46 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -157,7 +157,7 @@ static struct snd_soc_jack_pin byt_cht_es8316_jack_pins[] = { static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) { - struct snd_soc_component *codec = runtime->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(runtime, 0)->component; struct snd_soc_card *card = runtime->card; struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card); const struct snd_soc_dapm_route *custom_map; @@ -212,7 +212,7 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) if (ret) dev_err(card->dev, "unable to enable MCLK\n"); - ret = snd_soc_dai_set_sysclk(runtime->codec_dai, 0, 19200000, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(runtime, 0), 0, 19200000, SND_SOC_CLOCK_IN); if (ret < 0) { dev_err(card->dev, "can't set codec clock %d\n", ret); @@ -262,7 +262,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS @@ -272,7 +272,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c index 479af808ef4360..8c0dab1f40308b 100644 --- a/sound/soc/intel/boards/bytcht_nocodec.c +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -58,7 +58,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -68,7 +68,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6bd9ae813be289..33fb8ea4e5cb5c 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -381,7 +381,7 @@ static int byt_rt5640_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); return byt_rt5640_prepare_and_enable_pll1(dai, params_rate(params)); } @@ -805,7 +805,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map; int num_routes; int ret; @@ -962,7 +962,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -971,7 +971,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 5074bb53f98e83..214ef41e23e68a 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -348,7 +348,7 @@ static int byt_rt5651_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); snd_pcm_format_t format = params_format(params); int rate = params_rate(params); int bclk_ratio; @@ -540,7 +540,7 @@ static int byt_rt5651_add_codec_device_props(struct device *i2c_dev) static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *codec = runtime->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(runtime, 0)->component; struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); const struct snd_soc_dapm_route *custom_map; int num_routes; @@ -685,7 +685,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS @@ -696,7 +696,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index ea119d52392659..135701738a4449 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -113,7 +113,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK, @@ -257,7 +257,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, int ret = 0; unsigned int fmt = 0; - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 16); if (ret < 0) { dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret); return ret; @@ -266,7 +266,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret < 0) { dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 34d4e17e329594..f456150f89c269 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -73,7 +73,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, NAU8824_CLK_FLL_FS, 0, @@ -96,7 +96,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); struct snd_soc_jack *jack = &ctx->jack; - struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); struct snd_soc_component *component = codec_dai->component; int ret, jack_type; diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 452691db12ccb7..e64eca56e42604 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -208,7 +208,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ @@ -252,7 +252,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; int jack_type; int ret; @@ -359,7 +359,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 16-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS @@ -369,7 +369,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_fmt(rtd->codec_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS @@ -379,7 +379,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 16); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; @@ -393,7 +393,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, /* * Default mode for SSP configuration is TDM 4 slot */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -403,7 +403,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, } /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24); if (ret < 0) { dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 9d657421730a7d..097023a3ec14a2 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -144,7 +144,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ @@ -176,7 +176,7 @@ static const struct acpi_gpio_mapping cht_rt5672_gpios[] = { static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); struct snd_soc_component *component = codec_dai->component; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); @@ -255,7 +255,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, /* * Default mode for SSP configuration is TDM 4 slot */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -265,7 +265,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, } /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24); if (ret < 0) { dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index ed6c26a256e7a3..8167b2977e1d3e 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -85,7 +85,7 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; int ret; @@ -125,7 +125,7 @@ static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int clk_id, clk_freq, pll_out, ret; clk_id = RT5682_PLL1_S_MCLK; @@ -274,7 +274,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) static int hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 3c576b33b9c644..f13158e4a1fcd6 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -136,8 +136,8 @@ static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_jack *jack; int ret; @@ -188,7 +188,7 @@ static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* Set valid bitmask & configuration for I2S in 24 bit */ @@ -208,7 +208,7 @@ static struct snd_soc_ops geminilake_rt5682_ops = { static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct glk_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -225,7 +225,7 @@ static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; struct snd_soc_dapm_context *dapm; int ret; diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 6589fa56873f32..3ed53d7db4e696 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -56,7 +56,7 @@ static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index bc7f9a9ce9afd4..32cd90b8d4c4ec 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -159,8 +159,8 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_jack *jack; int ret; @@ -203,7 +203,7 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -236,7 +236,7 @@ static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 0ceb1748a26225..abd4e3839678e8 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -331,7 +331,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; struct snd_soc_card *card = rtd->card; int ret; @@ -381,7 +381,7 @@ static int kabylake_dmic_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -414,7 +414,7 @@ static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index e23dea9ab79a3f..6460e3f0c9749c 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -157,7 +157,7 @@ static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); ret = devm_acpi_dev_add_driver_gpios(component->dev, acpi_rt5660_gpios); @@ -210,7 +210,7 @@ static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -244,7 +244,7 @@ static int kabylake_rt5660_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 20d566e9dd9d80..658a9da3a40fd1 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -242,7 +242,7 @@ static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -258,7 +258,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; /* @@ -305,7 +305,7 @@ static int kabylake_rt5663_max98927_codec_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -431,7 +431,7 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6493ede8930014..1b1f8d7a4ea3fc 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -206,7 +206,7 @@ static struct snd_soc_codec_conf max98927_codec_conf[] = { static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; int ret; dapm = snd_soc_component_get_dapm(component); @@ -221,7 +221,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; /* @@ -255,7 +255,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -372,7 +372,7 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 8216c15fc8da17..d7b8154c43a4b8 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* * Headset buttons map to the google Reference headset. @@ -182,7 +182,7 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -200,7 +200,7 @@ static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -218,7 +218,7 @@ static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -236,7 +236,7 @@ static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -296,7 +296,7 @@ static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 40f8eb53e8223f..4b317bcf6ea0a2 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -161,12 +161,12 @@ static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; /* Slot 1 for left */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[0], 0x01, 0x01, 2, 48); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0x01, 0x01, 2, 48); if (ret < 0) return ret; /* Slot 2 for right */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[1], 0x02, 0x02, 2, 48); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 1), 0x02, 0x02, 2, 48); if (ret < 0) return ret; @@ -176,7 +176,7 @@ static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* * 4 buttons here map to the google Reference headset @@ -201,7 +201,7 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -219,7 +219,7 @@ static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -238,7 +238,7 @@ static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -256,7 +256,7 @@ static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -348,7 +348,7 @@ static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index a9aec66a2351d0..903ae1b28ec96e 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -112,7 +112,7 @@ static const struct snd_soc_dapm_route skylake_rt286_map[] = { static int skylake_rt286_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -122,7 +122,7 @@ static int skylake_rt286_fe_init(struct snd_soc_pcm_runtime *rtd) static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret; ret = snd_soc_card_jack_new(rtd->card, "Headset", @@ -143,7 +143,7 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -229,7 +229,7 @@ static int skylake_rt286_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 239d8ffdbccd9f..b707dd3b562568 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -129,8 +129,8 @@ static struct snd_soc_jack headset; static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; struct snd_soc_jack *jack; int ret; @@ -173,7 +173,7 @@ static int ssp1_hw_params(struct snd_pcm_substream *substream, int ret, j; for (j = 0; j < runtime->num_codecs; j++) { - struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, j); if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { /* vmon_slot_no = 0 imon_slot_no = 1 for TX slots */ @@ -214,7 +214,7 @@ static struct snd_soc_codec_conf max98373_codec_conf[] = { static int hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 4ce707b6eb7995..fb781189999924 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -66,7 +66,7 @@ static const struct dmi_system_id sof_pcm512x_quirk_table[] = { static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct sof_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -84,7 +84,7 @@ static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int sof_pcm512x_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *codec = rtd->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; snd_soc_component_update_bits(codec, PCM512x_GPIO_EN, 0x08, 0x08); snd_soc_component_update_bits(codec, PCM512x_GPIO_OUTPUT_4, 0x0f, 0x02); @@ -97,7 +97,7 @@ static int sof_pcm512x_codec_init(struct snd_soc_pcm_runtime *rtd) static int aif1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *codec = rtd->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1, 0x08, 0x08); @@ -108,7 +108,7 @@ static int aif1_startup(struct snd_pcm_substream *substream) static void aif1_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *codec = rtd->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1, 0x08, 0x00); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 2eeaa14e59c01a..8c29431b584760 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -124,7 +124,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct sof_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -143,7 +143,7 @@ static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; int ret; @@ -211,7 +211,7 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int clk_id, clk_freq, pll_out, ret; if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) { diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 033d7c05d7fb1c..c183f8e94ee4eb 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -476,7 +476,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, u8 channels; int ret, dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; /* check if we are being called a subsequent time */ @@ -494,7 +494,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, } pcm_data->allocated = false; - pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id, + pcm_data->stream = sst_hsw_stream_new(hsw, asoc_rtd_to_cpu(rtd, 0)->id, hsw_notify_pointer, pcm_data); if (pcm_data->stream == NULL) { dev_err(rtd->dev, "error: failed to create stream\n"); @@ -509,7 +509,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, path_id = SST_HSW_STREAM_PATH_SSP0_IN; /* DSP stream type depends on DAI ID */ - switch (rtd->cpu_dai->id) { + switch (asoc_rtd_to_cpu(rtd, 0)->id) { case 0: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { stream_type = SST_HSW_STREAM_TYPE_SYSTEM; @@ -533,7 +533,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, break; default: dev_err(rtd->dev, "error: invalid DAI ID %d\n", - rtd->cpu_dai->id); + asoc_rtd_to_cpu(rtd, 0)->id); return -EINVAL; } @@ -595,7 +595,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, dmab = snd_pcm_get_dma_buf(substream); ret = create_adsp_page_table(substream, pdata, rtd, runtime->dma_area, - runtime->dma_bytes, rtd->cpu_dai->id); + runtime->dma_bytes, asoc_rtd_to_cpu(rtd, 0)->id); if (ret < 0) return ret; @@ -608,7 +608,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, pages = runtime->dma_bytes / PAGE_SIZE; ret = sst_hsw_stream_buffer(hsw, pcm_data->stream, - pdata->dmab[rtd->cpu_dai->id][substream->stream].addr, + pdata->dmab[asoc_rtd_to_cpu(rtd, 0)->id][substream->stream].addr, pages, runtime->dma_bytes, 0, snd_sgbuf_get_addr(dmab, 0) >> PAGE_SHIFT); if (ret < 0) { @@ -661,7 +661,7 @@ static int hsw_pcm_trigger(struct snd_soc_component *component, snd_pcm_uframes_t pos; int dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; sst_stream = pcm_data->stream; @@ -770,7 +770,7 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_soc_component *component, u32 position; int dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); @@ -791,7 +791,7 @@ static int hsw_pcm_open(struct snd_soc_component *component, struct sst_hsw *hsw = pdata->hsw; int dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); @@ -801,7 +801,7 @@ static int hsw_pcm_open(struct snd_soc_component *component, snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware); - pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id, + pcm_data->stream = sst_hsw_stream_new(hsw, asoc_rtd_to_cpu(rtd, 0)->id, hsw_notify_pointer, pcm_data); if (pcm_data->stream == NULL) { dev_err(rtd->dev, "error: failed to create stream\n"); @@ -824,7 +824,7 @@ static int hsw_pcm_close(struct snd_soc_component *component, struct sst_hsw *hsw = pdata->hsw; int ret, dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); @@ -923,9 +923,9 @@ static int hsw_pcm_new(struct snd_soc_component *component, hsw_pcm_hardware.buffer_bytes_max); } if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) - priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm; + priv_data->pcm[asoc_rtd_to_cpu(rtd, 0)->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm; if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) - priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm; + priv_data->pcm[asoc_rtd_to_cpu(rtd, 0)->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm; return 0; } diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 05a9677c5a53f2..89dcccdfb1cdce 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -545,7 +545,7 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *link_dev; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct skl_pipe_params p_params = {0}; struct hdac_ext_link *link; int stream_tag; @@ -644,7 +644,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, link_dev->link_prepared = 0; - link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name); if (!link) return -EINVAL; @@ -1074,7 +1074,7 @@ static int skl_platform_soc_open(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai_link = rtd->dai_link; - dev_dbg(rtd->cpu_dai->dev, "In %s:%s\n", __func__, + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "In %s:%s\n", __func__, dai_link->cpus->dai_name); snd_soc_set_runtime_hwparams(substream, &azx_pcm_hw); @@ -1226,7 +1226,7 @@ static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); u64 codec_frames, codec_nsecs; if (!codec_dai->driver->ops->delay) @@ -1281,7 +1281,7 @@ static int skl_platform_soc_get_time_info( static int skl_platform_soc_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_pcm *pcm = rtd->pcm; unsigned int size; From f844705f15201a9c6c2b4be0a45b3ed76de16b88 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:19:14 +0900 Subject: [PATCH 369/415] ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87pnd3ir71.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c index 8c3c808bda9a9e..4f66b011f1b4de 100644 --- a/sound/soc/kirkwood/armada-370-db.c +++ b/sound/soc/kirkwood/armada-370-db.c @@ -19,7 +19,7 @@ static int a370db_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int freq; switch (params_rate(params)) { diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index f882b4003edf90..e037826b24517c 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -20,7 +20,7 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) { struct snd_soc_pcm_runtime *soc_runtime = subs->private_data; - return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai); + return snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(soc_runtime, 0)); } static const struct snd_pcm_hardware kirkwood_dma_snd_hw = { From c8ac82127c83634847bf63cff00fb29ff2f7140a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:19:23 +0900 Subject: [PATCH 370/415] ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87o8snir6s.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 10 +++++----- sound/soc/mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 ++-- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 ++-- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 ++-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 4 ++-- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 6 +++--- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 4 ++-- .../mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- 14 files changed, 25 insertions(+), 25 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index 4254f3a954dd5f..375e3b492922ef 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -40,7 +40,7 @@ int mtk_afe_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct snd_pcm_runtime *runtime = substream->runtime; - int memif_num = rtd->cpu_dai->id; + int memif_num = asoc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[memif_num]; const struct snd_pcm_hardware *mtk_afe_hardware = afe->mtk_afe_hardware; int ret; @@ -100,7 +100,7 @@ void mtk_afe_fe_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); - struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; + struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id]; int irq_id; irq_id = memif->irq_usage; @@ -122,7 +122,7 @@ int mtk_afe_fe_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[id]; int ret; unsigned int channels = params_channels(params); @@ -199,7 +199,7 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime * const runtime = substream->runtime; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[id]; struct mtk_base_afe_irq *irqs = &afe->irqs[memif->irq_usage]; const struct mtk_base_irq_data *irq_data = irqs->irq_data; @@ -265,7 +265,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; int pbuf_size; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 44dfef71390573..0a1a65c86f0e8a 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -82,7 +82,7 @@ snd_pcm_uframes_t mtk_afe_pcm_pointer(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; + struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id]; const struct mtk_base_memif_data *memif_data = memif->data; struct regmap *regmap = afe->regmap; struct device *dev = afe->dev; diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 488603a0c4b1ba..f0250b0dd734ce 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -497,7 +497,7 @@ static int mt2701_memif_fs(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; int fs; - if (rtd->cpu_dai->id != MT2701_MEMIF_ULBT) + if (asoc_rtd_to_cpu(rtd, 0)->id != MT2701_MEMIF_ULBT) fs = mt2701_afe_i2s_fs(rate); else fs = (rate == 16000 ? 1 : 0); diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index b6941796efcaa9..c47af9b6949b31 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -128,8 +128,8 @@ static int mt2701_cs42448_be_ops_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int mclk_rate; unsigned int rate = params_rate(params); unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4; diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 8c4c89e4c616ff..0122e7df067f57 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -25,8 +25,8 @@ static int mt2701_wm8960_be_ops_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int mclk_rate; unsigned int rate = params_rate(params); unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4; diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 378bfc16ef52b0..7f930556d961c0 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -143,7 +143,7 @@ static int mt6797_memif_fs(struct snd_pcm_substream *substream, struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; return mt6797_rate_transform(afe->dev, rate, id); } diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 461e4de8c918e0..1e3f2d7860666e 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -485,7 +485,7 @@ static int mt8173_memif_fs(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; + struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id]; int fs; if (memif->data->id == MT8173_AFE_MEMIF_DAI || diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 22c00600c999fe..37693d354e66ee 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -53,7 +53,7 @@ static int mt8173_max98090_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); return snd_soc_dai_set_sysclk(codec_dai, 0, params_rate(params) * 256, SND_SOC_CLOCK_IN); @@ -67,7 +67,7 @@ static int mt8173_max98090_init(struct snd_soc_pcm_runtime *runtime) { int ret; struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headphone", SND_JACK_HEADPHONE, diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 5d82159f4f2ea0..51009a1727778b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -73,7 +73,7 @@ static struct snd_soc_jack mt8173_rt5650_rt5514_jack; static int mt8173_rt5650_rt5514_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dais[0]->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; int ret; rt5645_sel_asrc_clk_src(component, diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index f65e3ebe38b8a4..247ac7690805a7 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -77,8 +77,8 @@ static struct snd_soc_jack mt8173_rt5650_rt5676_jack; static int mt8173_rt5650_rt5676_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dais[0]->component; - struct snd_soc_component *component_sub = runtime->codec_dais[1]->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_component *component_sub = asoc_rtd_to_codec(runtime, 1)->component; int ret; rt5645_sel_asrc_clk_src(component, diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index bbc4ad749892c9..2065c94dbf99de 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -104,8 +104,8 @@ static struct snd_soc_jack mt8173_rt5650_jack, mt8173_rt5650_hdmi_jack; static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dais[0]->component; - const char *codec_capture_dai = runtime->codec_dais[1]->name; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + const char *codec_capture_dai = asoc_rtd_to_codec(runtime, 1)->name; int ret; rt5645_sel_asrc_clk_src(component, @@ -154,7 +154,7 @@ static int mt8173_rt5650_hdmi_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - return hdmi_codec_set_jack_detect(rtd->codec_dai->component, + return hdmi_codec_set_jack_detect(asoc_rtd_to_codec(rtd, 0)->component, &mt8173_rt5650_hdmi_jack); } diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 6e2270bbb10e58..c8ded53bde1dbd 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -146,7 +146,7 @@ static int mt8183_memif_fs(struct snd_pcm_substream *substream, struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; return mt8183_rate_transform(afe->dev, rate, id); } diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c4e4f1f99ddeef..5b3dfa79b4aea4 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -28,7 +28,7 @@ static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int mclk_fs_ratio = 128; unsigned int mclk_fs = rate * mclk_fs_ratio; - return snd_soc_dai_set_sysclk(rtd->cpu_dai, + return snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk_fs, SND_SOC_CLOCK_OUT); } @@ -47,7 +47,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int freq; int ret = 0, j; - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk_fs, SND_SOC_CLOCK_OUT); if (ret < 0) dev_err(rtd->dev, "failed to set cpu dai sysclk\n"); diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 0555f7d73d0501..1fca8df109b42f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -41,7 +41,7 @@ static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int mclk_fs_ratio = 128; unsigned int mclk_fs = rate * mclk_fs_ratio; - return snd_soc_dai_set_sysclk(rtd->cpu_dai, + return snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk_fs, SND_SOC_CLOCK_OUT); } From 385a5c60ad7ac778a24c2715a2085241b2d6a7f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:19:32 +0900 Subject: [PATCH 371/415] ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87mu87ir6j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-card.c | 8 ++++---- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- 4 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c index da8c098e875029..d9cede4c33ffc6 100644 --- a/sound/soc/meson/aiu-fifo.c +++ b/sound/soc/meson/aiu-fifo.c @@ -26,7 +26,7 @@ static struct snd_soc_dai *aiu_fifo_dai(struct snd_pcm_substream *ss) { struct snd_soc_pcm_runtime *rtd = ss->private_data; - return rtd->cpu_dai; + return asoc_rtd_to_cpu(rtd, 0); } snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component, diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 77a7d5f36ebf61..af46845f4ef2ba 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -72,10 +72,10 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) } } - ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask, + ret = axg_tdm_set_tdm_slots(asoc_rtd_to_cpu(rtd, 0), be->tx_mask, be->rx_mask, be->slots, be->slot_width); if (ret) { - dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "setting tdm link slots failed\n"); return ret; } @@ -90,10 +90,10 @@ static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) int ret; /* The loopback rx_mask is the pad tx_mask */ - ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask, + ret = axg_tdm_set_tdm_slots(asoc_rtd_to_cpu(rtd, 0), NULL, be->tx_mask, be->slots, be->slot_width); if (ret) { - dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "setting tdm link slots failed\n"); return ret; } diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index c12b0d5e8ebf3a..2e9b56b29d3131 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -47,7 +47,7 @@ static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss) { struct snd_soc_pcm_runtime *rtd = ss->private_data; - return rtd->cpu_dai; + return asoc_rtd_to_cpu(rtd, 0); } static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss) diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index b5d3c9f56bac0b..2ca8c98e204f2b 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -30,7 +30,7 @@ int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, SND_SOC_CLOCK_OUT); if (ret && ret != -ENOTSUPP) return ret; From 84a41e069d164231aa979c8dc5aa17f8ff9b90da Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:19:39 +0900 Subject: [PATCH 372/415] ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87lfnrir6c.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 9841e1da97826e..f46d7aca8cf6b7 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -20,8 +20,8 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int rate = params_rate(params); u32 mclk; int ret; From 8d8fef280c94869a4a96c2ac77aea435516fd838 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:19:49 +0900 Subject: [PATCH 373/415] ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87k13bir62.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/pxa/brownstone.c | 4 ++-- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/hx4700.c | 4 ++-- sound/soc/pxa/imote2.c | 4 ++-- sound/soc/pxa/magician.c | 8 ++++---- sound/soc/pxa/mioa701_wm9713.c | 4 ++-- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 ++-- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- sound/soc/pxa/ttc-dkb.c | 2 +- sound/soc/pxa/z2.c | 4 ++-- sound/soc/pxa/zylonite.c | 6 +++--- 14 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 53b1435ced3f75..016a9119948515 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -44,8 +44,8 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int freq_out, sspa_mclk, sysclk; if (params_rate(params) > 11025) { diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index d81082323fb4da..6fbef9a0afa730 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -116,8 +116,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 0139343dbcce71..b4da9a9a65210e 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -54,8 +54,8 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret = 0; /* set the I2S system clock as output */ diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 514e17724fc3db..3014e8244ab466 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -12,8 +12,8 @@ static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret; diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 6483cff5b73da1..1b926a5bfb50e5 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -83,8 +83,8 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int width; int ret = 0; @@ -121,8 +121,8 @@ static int magician_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret = 0; /* set codec DAI configuration */ diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 76e054d514a8fe..bf27b277c01f9e 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -73,7 +73,7 @@ static int rear_amp_event(struct snd_soc_dapm_widget *widget, struct snd_soc_component *component; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - component = rtd->codec_dai->component; + component = asoc_rtd_to_codec(rtd, 0)->component; return rear_amp_power(component, SND_SOC_DAPM_EVENT_ON(event)); } @@ -117,7 +117,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* Prepare GPIO8 for rear speaker amplifier */ snd_soc_component_update_bits(component, AC97_GPIO_CFG, 0x100, 0x100); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 287b5da739e521..3fe6c4c5a3ab24 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -112,7 +112,7 @@ static int mmp_pcm_open(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct platform_device *pdev = to_platform_device(component->dev); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct mmp_dma_data dma_data; struct resource *r; diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index e701637a9ae96b..3548a2634a635f 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -251,7 +251,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); struct ssp_device *sspa = sspa_priv->sspa; struct snd_dmaengine_dai_dma_data *dma_params; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 59ef04d0467a62..287984a564c819 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -90,8 +90,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 5f1c477b58330b..9a32bf72127a69 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -96,7 +96,7 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index f7babffb7228e0..6d8174f6293580 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -117,8 +117,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index d8f79e2266b12b..d5f2961b1a3e1b 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -61,7 +61,7 @@ static const struct snd_soc_dapm_route ttc_audio_map[] = { static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* Headset jack detection */ snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE | diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index f9a33cb36f5b8e..6eee1aefc89a86 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -34,8 +34,8 @@ static int z2_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 567dc133ea925e..447b59b8bd33cc 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -66,7 +66,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd) { if (clk_pout) - snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, + snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0, clk_get_rate(pout), 0); return 0; @@ -76,8 +76,8 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); From 6e3a98bcc8678545ad69b200f6f35740bfc70d3a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:20:01 +0900 Subject: [PATCH 374/415] ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87imivir5q.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/apq8096.c | 6 +++--- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 22 +++++++++++----------- sound/soc/qcom/storm.c | 2 +- 7 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 7647af3e51f6b7..2ef090f4af9e92 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -33,7 +33,7 @@ struct apq8016_sbc_data { static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai; struct snd_soc_component *component; struct snd_soc_card *card = rtd->card; diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 94363fd6846ab4..d55e3ad9671608 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -31,8 +31,8 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0; @@ -66,7 +66,7 @@ static struct snd_soc_ops apq8096_ops = { static int apq8096_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* * Codec SLIMBUS configuration diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 5d1bc57571698c..34f7fd1bab1cfb 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -55,7 +55,7 @@ static int lpass_platform_pcmops_open(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); struct lpass_variant *v = drvdata->variant; int ret, dma_ch, dir = substream->stream; diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 8b5d86be9acec7..f6c7cddf08e8a9 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -333,7 +333,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_prtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0); struct q6asm_dai_rtd *prtd; struct q6asm_dai_data *pdata; struct device *dev = component->dev; @@ -545,7 +545,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) struct snd_soc_pcm_runtime *rtd = stream->private_data; struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct snd_compr_runtime *runtime = stream->runtime; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct q6asm_dai_data *pdata; struct device *dev = c->dev; struct q6asm_dai_rtd *prtd; diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 4d5915b9a06d52..46e50612b92c10 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -926,7 +926,7 @@ static int routing_hw_params(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct msm_routing_data *data = dev_get_drvdata(component->dev); - unsigned int be_id = rtd->cpu_dai->id; + unsigned int be_id = asoc_rtd_to_cpu(rtd, 0)->id; struct session_data *session; int path_type; diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 67a55edf755fe2..b2de65c7f95cb7 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -46,7 +46,7 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card); u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; @@ -86,7 +86,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai; int ret = 0, j; int channels, slot_width; @@ -171,8 +171,8 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret = 0; switch (cpu_dai->id) { @@ -220,8 +220,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); struct snd_jack *jack; /* @@ -304,8 +304,8 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int j; int ret; @@ -394,7 +394,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); switch (cpu_dai->id) { case PRIMARY_MI2S_RX: @@ -439,7 +439,7 @@ static int sdm845_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; int ret; @@ -478,7 +478,7 @@ static int sdm845_snd_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; if (sruntime && data->stream_prepared[cpu_dai->id]) { diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index e6666e597265a6..3a6e18709b9e24 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -39,7 +39,7 @@ static int storm_ops_hw_params(struct snd_pcm_substream *substream, */ sysclk_freq = rate * bitwidth * 2 * STORM_SYSCLK_MULT; - ret = snd_soc_dai_set_sysclk(soc_runtime->cpu_dai, 0, sysclk_freq, 0); + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(soc_runtime, 0), 0, sysclk_freq, 0); if (ret) { dev_err(card->dev, "error setting sysclk to %u: %d\n", sysclk_freq, ret); From a7ff526814d5be3cf38bafdbdb1217225cd12922 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:20:09 +0900 Subject: [PATCH 375/415] ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87h7yfir5i.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 ++-- sound/soc/rockchip/rk3399_gru_sound.c | 16 ++++++++-------- sound/soc/rockchip/rockchip_max98090.c | 6 +++--- sound/soc/rockchip/rockchip_rt5645.c | 6 +++--- 4 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c index 767700c34ee247..01078155a91486 100644 --- a/sound/soc/rockchip/rk3288_hdmi_analog.c +++ b/sound/soc/rockchip/rk3288_hdmi_analog.c @@ -67,8 +67,8 @@ static int rk_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int mclk; switch (params_rate(params)) { diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index d951100bf77007..f45e5aaa4b3028 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -57,7 +57,7 @@ static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substrea mclk = params_rate(params) * SOUND_FS; - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, 0); if (ret) { dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", __func__, mclk, ret); @@ -71,8 +71,8 @@ static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int mclk; int ret; @@ -103,8 +103,8 @@ static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int mclk, ret; /* in bypass mode, the mclk has to be one of the frequencies below */ @@ -153,8 +153,8 @@ static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dais[0]->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* We need default MCLK and PLL settings for the accessory detection */ @@ -206,7 +206,7 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream, mclk = params_rate(params) * SOUND_FS; - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, 0); if (ret) { dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", __func__, mclk, ret); diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 60930fa85aa495..1f527d3763ce12 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -146,8 +146,8 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int mclk; switch (params_rate(params)) { @@ -227,7 +227,7 @@ static struct snd_soc_jack rk_hdmi_jack; static int rk_hdmi_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; int ret; /* enable jack detection */ diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 26b67b24548467..0617ccf4e42c15 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -56,8 +56,8 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int mclk; switch (params_rate(params)) { @@ -113,7 +113,7 @@ static int rk_init(struct snd_soc_pcm_runtime *runtime) return ret; } - return rt5645_set_jack_detect(runtime->codec_dai->component, + return rt5645_set_jack_detect(asoc_rtd_to_codec(runtime, 0)->component, &headset_jack, &headset_jack, &headset_jack); From 7de6b6bc1a58ec3118ca825d8b48faac3a956a85 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:20:20 +0900 Subject: [PATCH 376/415] ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87ftdzir57.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/samsung/arndale.c | 6 +++--- sound/soc/samsung/bells.c | 16 ++++++++-------- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 4 ++-- sound/soc/samsung/littlemill.c | 14 +++++++------- sound/soc/samsung/lowland.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 10 +++++----- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 ++-- sound/soc/samsung/s3c24xx_uda134x.c | 6 +++--- sound/soc/samsung/smartq_wm8987.c | 4 ++-- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 4 ++-- sound/soc/samsung/snow.c | 4 ++-- sound/soc/samsung/spdif.c | 8 ++++---- sound/soc/samsung/speyside.c | 8 ++++---- sound/soc/samsung/tm2_wm5110.c | 16 ++++++++-------- sound/soc/samsung/tobermory.c | 8 ++++---- 24 files changed, 68 insertions(+), 68 deletions(-) diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index 6e6d67d6e0ab4c..c81ece78e0360f 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -21,8 +21,8 @@ static int arndale_rt5631_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int rfs, ret; unsigned long rclk; @@ -56,7 +56,7 @@ static int arndale_wm1811_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int rfs, rclk; /* Ensure AIF1CLK is >= 3 MHz for optimal performance */ diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 5de633497f839b..8b83f39c3ac918 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -60,7 +60,7 @@ static int bells_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); component = codec_dai->component; if (dapm->dev != codec_dai->dev) @@ -106,7 +106,7 @@ static int bells_set_bias_level_post(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); component = codec_dai->component; if (dapm->dev != codec_dai->dev) @@ -152,11 +152,11 @@ static int bells_late_probe(struct snd_soc_card *card) int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_AP_DSP]); - wm0010 = rtd->codec_dai->component; + wm0010 = asoc_rtd_to_codec(rtd, 0)->component; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]); - component = rtd->codec_dai->component; - aif1_dai = rtd->codec_dai; + component = asoc_rtd_to_codec(rtd, 0)->component; + aif1_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_component_set_sysclk(component, ARIZONA_CLK_SYSCLK, ARIZONA_CLK_SRC_FLL1, @@ -195,7 +195,7 @@ static int bells_late_probe(struct snd_soc_card *card) } rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_CODEC_CP]); - aif2_dai = rtd->cpu_dai; + aif2_dai = asoc_rtd_to_cpu(rtd, 0); ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); if (ret != 0) { @@ -207,8 +207,8 @@ static int bells_late_probe(struct snd_soc_card *card) return 0; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_CODEC_SUB]); - aif3_dai = rtd->cpu_dai; - wm9081_dai = rtd->codec_dai; + aif3_dai = asoc_rtd_to_cpu(rtd, 0); + wm9081_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0); if (ret != 0) { diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index a95c34e53a2b90..9139a1e7e20062 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -68,7 +68,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int div; int ret; unsigned int rate = params_rate(params); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a57bb989a0efd5..f86e3028b40240 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -932,7 +932,7 @@ static int i2s_trigger(struct snd_pcm_substream *substream, struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct i2s_dai *i2s = to_info(rtd->cpu_dai); + struct i2s_dai *i2s = to_info(asoc_rtd_to_cpu(rtd, 0)); unsigned long flags; switch (cmd) { diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 949d2e02996255..30899016cf08a5 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -33,8 +33,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct s3c_i2sv2_rate_calc div; unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 2f2f83a8c23a35..f4375c49f7f49d 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -23,7 +23,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - aif1_dai = rtd->codec_dai; + aif1_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != aif1_dai->dev) return 0; @@ -70,7 +70,7 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - aif1_dai = rtd->codec_dai; + aif1_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != aif1_dai->dev) return 0; @@ -105,7 +105,7 @@ static int littlemill_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; sample_rate = params_rate(params); @@ -181,7 +181,7 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]); - aif2_dai = rtd->cpu_dai; + aif2_dai = asoc_rtd_to_cpu(rtd, 0); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -264,11 +264,11 @@ static int littlemill_late_probe(struct snd_soc_card *card) int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - component = rtd->codec_dai->component; - aif1_dai = rtd->codec_dai; + component = asoc_rtd_to_codec(rtd, 0)->component; + aif1_dai = asoc_rtd_to_codec(rtd, 0); rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]); - aif2_dai = rtd->cpu_dai; + aif2_dai = asoc_rtd_to_cpu(rtd, 0); ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index fcc7897ee7d05e..998d10cf8c9476 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -32,7 +32,7 @@ static struct snd_soc_jack_pin lowland_headset_pins[] = { static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret; ret = snd_soc_component_set_sysclk(component, WM5100_CLK_SYSCLK, @@ -65,7 +65,7 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; snd_soc_dapm_nc_pin(&rtd->card->dapm, "LINEOUT"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 1339e41e986072..b7ce1da854cee0 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -26,8 +26,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int pll_out = 0, bclk = 0; int ret = 0; unsigned long iis_clkrate; @@ -100,7 +100,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); @@ -118,7 +118,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int pcmdiv = 0; int ret = 0; unsigned long iis_clkrate; @@ -155,7 +155,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 30c7e1bc2a30b2..6eda5af989fe38 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -98,7 +98,7 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, return ret; if (rtd->num_codecs > 1) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[1]; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 1); ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk_freq, SND_SOC_CLOCK_IN); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index f6e67d0e7882b3..a5b1a12b349660 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -212,7 +212,7 @@ static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); unsigned long flags; dev_dbg(pcm->dev, "Entered %s\n", __func__); @@ -256,7 +256,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *socdai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 4b247e91ae5b2d..3afe63c0923ebd 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -149,7 +149,7 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int div; int ret; unsigned int rate = params_rate(params); diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 593be1b668d644..35888784829388 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -380,7 +380,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_i2sv2_info *i2s = to_info(rtd->cpu_dai); + struct s3c_i2sv2_info *i2s = to_info(asoc_rtd_to_cpu(rtd, 0)); int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 4543705b8d87d2..fd2a4da086f308 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -160,8 +160,8 @@ static int simtec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, 0, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 55d2a802a6cb6a..abb5c4713c534e 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -51,7 +51,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c24xx_uda134x *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret = 0; mutex_lock(&priv->clk_lock); @@ -119,8 +119,8 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; int clk_source, fs_mode; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index fab3db9fdb98f2..36bef136d57f81 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -25,8 +25,8 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret; diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index 4baef84d29ee22..776a270261bf1b 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -101,7 +101,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned long pll_out, rclk_rate; int ret, ratio; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index d096ff9122607b..02074c34a2b2e1 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -23,7 +23,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int pll_out; int rfs, ret; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 8fa5f6b387adcf..a9f345f19a8a1c 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -45,7 +45,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int pll_out; int ret; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 6e44f7927852e3..746930dde5d791 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -44,8 +44,8 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned long mclk_freq; int rfs, ret; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index bebcf0a4d608e7..40c5de8df0ffe7 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -110,9 +110,9 @@ static int snow_late_probe(struct snd_soc_card *card) /* In the multi-codec case codec_dais 0 is MAX98095 and 1 is HDMI. */ if (rtd->num_codecs > 1) - codec_dai = rtd->codec_dais[0]; + codec_dai = asoc_rtd_to_codec(rtd, 0); else - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); /* Set the MCLK rate for the codec */ return snd_soc_dai_set_sysclk(codec_dai, 0, diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 1a9f08a50394bf..759fc66443298a 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -142,7 +142,7 @@ static int spdif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai); + struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0)); unsigned long flags; dev_dbg(spdif->dev, "Entered %s\n", __func__); @@ -178,7 +178,7 @@ static int spdif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *socdai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai); + struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0)); void __iomem *regs = spdif->regs; struct snd_dmaengine_dai_dma_data *dma_data; u32 con, clkcon, cstas; @@ -194,7 +194,7 @@ static int spdif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); + snd_soc_dai_set_dma_data(asoc_rtd_to_cpu(rtd, 0), substream, dma_data); spin_lock_irqsave(&spdif->lock, flags); @@ -280,7 +280,7 @@ static void spdif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai); + struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0)); void __iomem *regs = spdif->regs; u32 con, clkcon; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 8f175f204eb70a..f5f6ba00d0731f 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -25,7 +25,7 @@ static int speyside_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -61,7 +61,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -131,7 +131,7 @@ static void speyside_set_polarity(struct snd_soc_component *component, static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(dai, 0, MCLK_AUDIO_RATE, 0); @@ -143,7 +143,7 @@ static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd) static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = dai->component; int ret; diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 043a287728b36b..6dfd540e2d743a 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -93,7 +93,7 @@ static int tm2_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card); switch (params_rate(params)) { @@ -134,7 +134,7 @@ static int tm2_aif2_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; unsigned int asyncclk_rate; int ret; @@ -188,7 +188,7 @@ static int tm2_aif2_hw_params(struct snd_pcm_substream *substream, static int tm2_aif2_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret; /* disable FLL2 */ @@ -209,7 +209,7 @@ static int tm2_hdmi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int bfs; int bitwidth, ret; @@ -284,7 +284,7 @@ static int tm2_set_bias_level(struct snd_soc_card *card, rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - if (dapm->dev != rtd->codec_dai->dev) + if (dapm->dev != asoc_rtd_to_codec(rtd, 0)->dev) return 0; switch (level) { @@ -315,8 +315,8 @@ static int tm2_late_probe(struct snd_soc_card *card) int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[TM2_DAI_AIF1]); - aif1_dai = rtd->codec_dai; - priv->component = rtd->codec_dai->component; + aif1_dai = asoc_rtd_to_codec(rtd, 0); + priv->component = asoc_rtd_to_codec(rtd, 0)->component; ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0); if (ret < 0) { @@ -325,7 +325,7 @@ static int tm2_late_probe(struct snd_soc_card *card) } rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[TM2_DAI_AIF2]); - aif2_dai = rtd->codec_dai; + aif2_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); if (ret < 0) { diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 1aa3fdb4b15218..c962d2c2a7f789 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -23,7 +23,7 @@ static int tobermory_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -66,7 +66,7 @@ static int tobermory_set_bias_level_post(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -181,8 +181,8 @@ static int tobermory_late_probe(struct snd_soc_card *card) int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - component = rtd->codec_dai->component; - codec_dai = rtd->codec_dai; + component = asoc_rtd_to_codec(rtd, 0)->component; + codec_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, 32768, SND_SOC_CLOCK_IN); From 34a43780622ace5a495d1dd661e5d493123d4e3f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:20:30 +0900 Subject: [PATCH 377/415] ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87eetjir4x.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/dma-sh7760.c | 16 ++++++++-------- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 6 +++--- sound/soc/sh/rcar/core.c | 2 +- 4 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index eee1a1e994cbb5..a35de78f14a9cc 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -119,7 +119,7 @@ static int camelot_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; int ret, dmairq; @@ -132,7 +132,7 @@ static int camelot_pcm_open(struct snd_soc_component *component, ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); if (unlikely(ret)) { pr_debug("audio unit %d irqs already taken!\n", - rtd->cpu_dai->id); + asoc_rtd_to_cpu(rtd, 0)->id); return -EBUSY; } (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); @@ -141,7 +141,7 @@ static int camelot_pcm_open(struct snd_soc_component *component, ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); if (unlikely(ret)) { pr_debug("audio unit %d irqs already taken!\n", - rtd->cpu_dai->id); + asoc_rtd_to_cpu(rtd, 0)->id); return -EBUSY; } (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); @@ -153,7 +153,7 @@ static int camelot_pcm_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; int dmairq; @@ -175,7 +175,7 @@ static int camelot_hw_params(struct snd_soc_component *component, struct snd_pcm_hw_params *hw_params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; int ret; @@ -194,7 +194,7 @@ static int camelot_prepare(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); @@ -242,7 +242,7 @@ static int camelot_trigger(struct snd_soc_component *component, struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; switch (cmd) { @@ -270,7 +270,7 @@ static snd_pcm_uframes_t camelot_pos(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; unsigned long pos; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5ef4221be6c396..1c3c4fdc9befdf 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -408,7 +408,7 @@ static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - return rtd->cpu_dai; + return asoc_rtd_to_cpu(rtd, 0); } static struct fsi_priv *fsi_get_priv_frm_dai(struct snd_soc_dai *dai) diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 991557e25eba66..d5702fbf176bea 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -46,7 +46,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; unsigned int rate = params_rate(params); @@ -67,7 +67,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream, clk_set_rate(&siumckb_clk, codec_freq); dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq); - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, SIU_CLKB_EXT, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), SIU_CLKB_EXT, codec_freq / 2, SND_SOC_CLOCK_IN); if (!ret) @@ -79,7 +79,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream, static int migor_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); if (use_count) { use_count--; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 0bfcb77e5f6587..4349f2fb823ff1 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -696,7 +696,7 @@ struct snd_soc_dai *rsnd_substream_to_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - return rtd->cpu_dai; + return asoc_rtd_to_cpu(rtd, 0); } static From be3e8de706b9219c0074eb780400a167ed7633e3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:20:37 +0900 Subject: [PATCH 378/415] ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87d093ir4q.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 6 +++--- sound/soc/sof/intel/hda-dsp.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index b9e3ce65e778e3..833dc303b39410 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -204,7 +204,7 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, struct hdac_bus *bus = hstream->bus; struct hdac_ext_stream *link_dev; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct sof_intel_hda_stream *hda_stream; struct hda_pipe_params p_params = {0}; struct hdac_ext_link *link; @@ -293,7 +293,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, bus = hstream->bus; rtd = snd_pcm_substream_chip(substream); - link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name); if (!link) return -EINVAL; @@ -374,7 +374,7 @@ static int hda_link_hw_free(struct snd_pcm_substream *substream, if (ret < 0) return ret; - link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name); if (!link) return -EINVAL; diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index c396b7ef0328d9..725be6ccd71049 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -845,7 +845,7 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) */ if (stream->link_substream) { rtd = snd_pcm_substream_chip(stream->link_substream); - name = rtd->codec_dai->component->name; + name = asoc_rtd_to_codec(rtd, 0)->component->name; link = snd_hdac_ext_bus_get_link(bus, name); if (!link) return -EINVAL; From 82d4c713335381651a5e2cff085ab150ad5de03d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:20:44 +0900 Subject: [PATCH 379/415] ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87blonir4j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sprd/sprd-pcm-compress.c | 4 ++-- sound/soc/sprd/sprd-pcm-dma.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c index 6cddf551bc110e..74d48340cade5b 100644 --- a/sound/soc/sprd/sprd-pcm-compress.c +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -135,7 +135,7 @@ static int sprd_platform_compr_dma_config(struct snd_compr_stream *cstream, struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct device *dev = component->dev; - struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sprd_compr_data *data = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct sprd_pcm_dma_params *dma_params = data->dma_params; struct sprd_compr_dma *dma = &stream->dma[channel]; struct dma_slave_config config = { }; @@ -321,7 +321,7 @@ static int sprd_platform_compr_open(struct snd_compr_stream *cstream) struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct device *dev = component->dev; - struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sprd_compr_data *data = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct sprd_compr_stream *stream; struct sprd_compr_callback cb; int stream_id = cstream->direction, ret; diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c index 2284558684bc87..d12d3cad8cbd79 100644 --- a/sound/soc/sprd/sprd-pcm-dma.c +++ b/sound/soc/sprd/sprd-pcm-dma.c @@ -200,7 +200,7 @@ static int sprd_pcm_hw_params(struct snd_soc_component *component, unsigned long flags; int ret, i, j, sg_num; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dma_params) { dev_warn(component->dev, "no dma parameters setting\n"); dma_private->params = NULL; From b1bee67c327de14bc7e9a84c91892747ee1f9ab0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:20:52 +0900 Subject: [PATCH 380/415] ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87a747ir4b.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 12 ++++++------ sound/soc/stm/stm32_sai_sub.c | 2 +- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 51407a21c44098..16ff02953015cf 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -215,7 +215,7 @@ static int stm32_adfsdm_trigger(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -235,7 +235,7 @@ static int stm32_adfsdm_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); int ret; ret = snd_soc_set_runtime_hwparams(substream, &stm32_adfsdm_pcm_hw); @@ -250,7 +250,7 @@ static int stm32_adfsdm_pcm_close(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); priv->substream = NULL; @@ -263,7 +263,7 @@ static snd_pcm_uframes_t stm32_adfsdm_pcm_pointer( { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); return bytes_to_frames(substream->runtime, priv->pos); } @@ -274,7 +274,7 @@ static int stm32_adfsdm_pcm_hw_params(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); priv->pcm_buff = substream->runtime->dma_area; @@ -287,7 +287,7 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_component *component, { struct snd_pcm *pcm = rtd->pcm; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); unsigned int size = DFSDM_MAX_PERIODS * DFSDM_MAX_PERIOD_SIZE; snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index fe4903260d4e40..2bd280c01c3316 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1238,7 +1238,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); int *ptr = (int *)(runtime->dma_area + hwoff + channel * (runtime->dma_bytes / runtime->channels)); From 2dc5fd034142426de1a4c60f63149b3ea8fc0b88 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:00 +0900 Subject: [PATCH 381/415] ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/878sjrir43.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index 98a9fe64552138..86779a99df751d 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -244,7 +244,7 @@ static int sun4i_spdif_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -EINVAL; From 0b25cffb2c117186ee6c9701c6680afa1a1748f2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:07 +0900 Subject: [PATCH 382/415] ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/877dzbir3w.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 6 +++--- sound/soc/tegra/trimslice.c | 2 +- 8 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 9e8b1497efd3e5..ec39ecba1e8b84 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -37,7 +37,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 4954a33ff46bcb..d800b62b36f830 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -38,7 +38,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index d46915a3ec4cba..9878bc3eb89e9c 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -40,7 +40,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 81cb6cc6236e44..5821313db977a3 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -42,7 +42,7 @@ static int tegra_rt5677_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); int srate, mclk, err; diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index e13b81d29cf359..dc411ba2e36d5e 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -36,7 +36,7 @@ static int tegra_sgtl5000_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_sgtl5000 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f6dd790dad71cc..0d653a605358ce 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -40,7 +40,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 071c7d2de77c70..9b5651502f1274 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -45,7 +45,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -173,7 +173,7 @@ static const struct snd_kcontrol_new tegra_wm8903_controls[] = { static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); @@ -205,7 +205,7 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; wm8903_mic_detect(component, NULL, 0, 0); diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 3f67ddd13674e5..f9834afaa2e8b7 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -35,7 +35,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; From 2842b87148af9ab7a0f3913022f2935e47cedd97 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:14 +0900 Subject: [PATCH 383/415] ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/875zevir3p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/ti/ams-delta.c | 4 ++-- sound/soc/ti/davinci-evm.c | 4 ++-- sound/soc/ti/davinci-vcif.c | 4 ++-- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +++--- sound/soc/ti/omap-mcbsp-st.c | 2 +- sound/soc/ti/omap-mcbsp.c | 4 ++-- sound/soc/ti/omap-mcpdm.c | 2 +- sound/soc/ti/omap3pandora.c | 4 ++-- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- 11 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 8e2fb81ad05c72..e17cd5e939f080 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -460,14 +460,14 @@ static void ams_delta_shutdown(struct snd_pcm_substream *substream) static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct snd_soc_dapm_context *dapm = &card->dapm; int ret; /* Codec is ready, now add/activate board specific controls */ /* Store a pointer to the codec structure for tty ldisc use */ - cx20442_codec = rtd->codec_dai->component; + cx20442_codec = asoc_rtd_to_codec(rtd, 0)->component; /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 686b23d7a90d98..2cfbeebdfb41b6 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -54,8 +54,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_card *soc_card = rtd->card; int ret = 0; unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c index c84650e4a7aa7e..ee4d3ef821a133 100644 --- a/sound/soc/ti/davinci-vcif.c +++ b/sound/soc/ti/davinci-vcif.c @@ -43,7 +43,7 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_vcif_dev *davinci_vcif_dev = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; u32 w; @@ -62,7 +62,7 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_vcif_dev *davinci_vcif_dev = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; u32 w; diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c index 3ad2b6daf31eba..a1672b479cb78e 100644 --- a/sound/soc/ti/n810.c +++ b/sound/soc/ti/n810.c @@ -101,7 +101,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int err; /* Set the codec system clock for DAC and ADC */ diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c index 6d564ab5e437a6..61e45fea5dd8f6 100644 --- a/sound/soc/ti/omap-abe-twl6040.c +++ b/sound/soc/ti/omap-abe-twl6040.c @@ -46,7 +46,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int clk_id, freq; @@ -78,7 +78,7 @@ static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret = 0; ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS, @@ -166,7 +166,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_card *card = rtd->card; struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 1a3fe854e8568a..5a32b54bbf3bb5 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -489,7 +489,7 @@ OMAP_MCBSP_ST_CONTROLS(3); int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); if (!mcbsp->st_data) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 26b503bbdb5fef..ff24546a10ee5f 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -737,7 +737,7 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream, unsigned int packet_size) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); int words; @@ -902,7 +902,7 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay( struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); u16 fifo_use; snd_pcm_sframes_t delay; diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index a726cd7a82526e..0ff33fe165f20f 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -532,7 +532,7 @@ static const struct snd_soc_component_driver omap_mcpdm_component = { void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd, u8 rx1, u8 rx2) { - struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2); } diff --git a/sound/soc/ti/omap3pandora.c b/sound/soc/ti/omap3pandora.c index 545f8dac9bd52d..b04146311b31ac 100644 --- a/sound/soc/ti/omap3pandora.c +++ b/sound/soc/ti/omap3pandora.c @@ -32,8 +32,8 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; /* Set the codec system clock for DAC and ADC */ diff --git a/sound/soc/ti/osk5912.c b/sound/soc/ti/osk5912.c index 1ca466bc402591..e01485cc51a151 100644 --- a/sound/soc/ti/osk5912.c +++ b/sound/soc/ti/osk5912.c @@ -39,7 +39,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int err; /* Set the codec system clock for DAC and ADC */ diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c index fdb0dc85fe672e..2a714a0041636f 100644 --- a/sound/soc/ti/rx51.c +++ b/sound/soc/ti/rx51.c @@ -103,7 +103,7 @@ static int rx51_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* Set the codec system clock for DAC and ADC */ return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000, From f7c4880113abde39053967eea378200127d69c9d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:21 +0900 Subject: [PATCH 384/415] ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/874kufir3i.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 985487cc3a5571..4b1cd4da3e368e 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -269,7 +269,7 @@ static int txx9aclc_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); struct snd_pcm *pcm = rtd->pcm; struct platform_device *pdev = to_platform_device(component->dev); struct txx9aclc_soc_device *dev; From 41759f4c43f1152d785ae74e31991a4ad8dd7ad8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:28 +0900 Subject: [PATCH 385/415] ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87369zir3b.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/uniphier/aio-compress.c | 22 +++++++++++----------- sound/soc/uniphier/aio-dma.c | 6 +++--- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/uniphier/aio-compress.c b/sound/soc/uniphier/aio-compress.c index 17f773ac5ca189..232d3cc5bce015 100644 --- a/sound/soc/uniphier/aio-compress.c +++ b/sound/soc/uniphier/aio-compress.c @@ -23,7 +23,7 @@ static int uniphier_aio_comprdma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_compr *compr = rtd->compr; struct device *dev = compr->card->dev; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[compr->direction]; size_t size = AUD_RING_SIZE; int dma_dir = DMA_FROM_DEVICE, ret; @@ -56,7 +56,7 @@ static int uniphier_aio_comprdma_free(struct snd_soc_pcm_runtime *rtd) { struct snd_compr *compr = rtd->compr; struct device *dev = compr->card->dev; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[compr->direction]; int dma_dir = DMA_FROM_DEVICE; @@ -73,7 +73,7 @@ static int uniphier_aio_comprdma_free(struct snd_soc_pcm_runtime *rtd) static int uniphier_aio_compr_open(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; int ret; @@ -98,7 +98,7 @@ static int uniphier_aio_compr_open(struct snd_compr_stream *cstream) static int uniphier_aio_compr_free(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; int ret; @@ -118,7 +118,7 @@ static int uniphier_aio_compr_get_params(struct snd_compr_stream *cstream, struct snd_codec *params) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; *params = sub->cparams.codec; @@ -130,7 +130,7 @@ static int uniphier_aio_compr_set_params(struct snd_compr_stream *cstream, struct snd_compr_params *params) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; struct device *dev = &aio->chip->pdev->dev; int ret; @@ -165,7 +165,7 @@ static int uniphier_aio_compr_set_params(struct snd_compr_stream *cstream, static int uniphier_aio_compr_hw_free(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; sub->setting = 0; @@ -177,7 +177,7 @@ static int uniphier_aio_compr_prepare(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *runtime = cstream->runtime; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; int bytes = runtime->fragment_size; unsigned long flags; @@ -215,7 +215,7 @@ static int uniphier_aio_compr_trigger(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *runtime = cstream->runtime; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; struct device *dev = &aio->chip->pdev->dev; int bytes = runtime->fragment_size, ret = 0; @@ -248,7 +248,7 @@ static int uniphier_aio_compr_pointer(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *runtime = cstream->runtime; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; int bytes = runtime->fragment_size; unsigned long flags; @@ -322,7 +322,7 @@ static int uniphier_aio_compr_copy(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *runtime = cstream->runtime; struct device *carddev = rtd->compr->card->dev; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; size_t cnt = min_t(size_t, count, aio_rb_space_to_end(sub) / 2); int bytes = runtime->fragment_size; diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c index da83423c52e291..4bbcb007df41e6 100644 --- a/sound/soc/uniphier/aio-dma.c +++ b/sound/soc/uniphier/aio-dma.c @@ -109,7 +109,7 @@ static int uniphier_aiodma_prepare(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[substream->stream]; int bytes = runtime->period_size * runtime->channels * samples_to_bytes(runtime, 1); @@ -136,7 +136,7 @@ static int uniphier_aiodma_trigger(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[substream->stream]; struct device *dev = &aio->chip->pdev->dev; int bytes = runtime->period_size * @@ -172,7 +172,7 @@ static snd_pcm_uframes_t uniphier_aiodma_pointer( { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[substream->stream]; int bytes = runtime->period_size * runtime->channels * samples_to_bytes(runtime, 1); From 99396e3883672a9073e7458b40504d647351b54f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:35 +0900 Subject: [PATCH 386/415] ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/871rpjir34.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/ux500/mop500_ab8500.c | 6 +++--- sound/soc/ux500/ux500_pcm.c | 8 ++++---- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 77655084bbdeeb..6aaa19829a73a3 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -215,8 +215,8 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct device *dev = rtd->card->dev; unsigned int fmt; int channels, ret = 0, driver_mode, slots; @@ -339,7 +339,7 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); mutex_lock(&mop500_ab8500_params_lock); __clear_bit(cpu_dai->id, &mop500_ab8500_usage); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 9445dbe8e039ff..39b96c132bc800 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -46,7 +46,7 @@ static const struct snd_pcm_hardware ux500_pcm_hw = { static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); u16 per_data_width, mem_data_width; struct stedma40_chan_cfg *dma_cfg; struct ux500_msp_dma_params *dma_params; @@ -86,7 +86,7 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct dma_slave_config *slave_config) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct msp_i2s_platform_data *pdata = rtd->cpu_dai->dev->platform_data; + struct msp_i2s_platform_data *pdata = asoc_rtd_to_cpu(rtd, 0)->dev->platform_data; struct snd_dmaengine_dai_dma_data *snd_dma_params; struct ux500_msp_dma_params *ste_dma_params; dma_addr_t dma_addr; @@ -94,11 +94,11 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, if (pdata) { ste_dma_params = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); dma_addr = ste_dma_params->tx_rx_addr; } else { snd_dma_params = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); dma_addr = snd_dma_params->addr; } From aafa4ef55c80f3c5f216f6b9b76dc469c917fe63 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:42 +0900 Subject: [PATCH 387/415] ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87zhc7hcih.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index bcf442faff7cbb..68af2176b19c95 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -373,7 +373,7 @@ static int xtfpga_pcm_open(struct snd_soc_component *component, void *p; snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); - p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + p = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); runtime->private_data = p; return 0; From 575be8838dcad6f127d8bbcb69cf0b8128342d5b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:49 +0900 Subject: [PATCH 388/415] ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87y2rrhcia.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/arm/pxa2xx-pcm-lib.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index a86c95d898246c..e81083e1bc68b2 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -38,7 +38,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct dma_slave_config config; int ret; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dma_params) return 0; @@ -47,7 +47,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, return ret; snd_dmaengine_pcm_set_config_from_dai_data(substream, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream), + snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream), &config); ret = dmaengine_slave_config(chan, &config); @@ -95,7 +95,7 @@ int pxa2xx_pcm_open(struct snd_pcm_substream *substream) runtime->hw = pxa2xx_pcm_hardware; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dma_params) return 0; @@ -120,7 +120,7 @@ int pxa2xx_pcm_open(struct snd_pcm_substream *substream) return ret; return snd_dmaengine_pcm_open( - substream, dma_request_slave_channel(rtd->cpu_dai->dev, + substream, dma_request_slave_channel(asoc_rtd_to_cpu(rtd, 0)->dev, dma_params->chan_name)); } EXPORT_SYMBOL(pxa2xx_pcm_open); From b5cb8558e53d28db571f4ae79b9e9590ed30b280 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 23 Mar 2020 14:21:56 +0900 Subject: [PATCH 389/415] ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87wo7bhci3.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l15.c | 4 ++-- sound/soc/codecs/cs47l24.c | 6 +++--- sound/soc/codecs/cs47l35.c | 6 +++--- sound/soc/codecs/cs47l85.c | 6 +++--- sound/soc/codecs/cs47l90.c | 6 +++--- sound/soc/codecs/cs47l92.c | 4 ++-- sound/soc/codecs/wm5110.c | 6 +++--- sound/soc/codecs/wm_adsp.c | 10 +++++----- 8 files changed, 24 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index e8840dc142ef36..8d1869bf7f9ce4 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -1239,12 +1239,12 @@ static int cs47l15_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l15-dsp-trace") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l15-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 25bffc2968f039..6b0570f596304b 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1076,14 +1076,14 @@ static int cs47l24_open(struct snd_compr_stream *stream) struct arizona *arizona = priv->core.arizona; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l24-dsp-voicectrl") == 0) { n_adsp = 2; - } else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l24-dsp-trace") == 0) { n_adsp = 1; } else { dev_err(arizona->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 3d48a0d9ecc581..18839807c9d15d 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -1514,14 +1514,14 @@ static int cs47l35_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l35-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l35-dsp-voicectrl") == 0) { n_adsp = 2; - } else if (strcmp(rtd->codec_dai->name, "cs47l35-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l35-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l85.c b/sound/soc/codecs/cs47l85.c index bef3471f482d49..a575113207f0b9 100644 --- a/sound/soc/codecs/cs47l85.c +++ b/sound/soc/codecs/cs47l85.c @@ -2457,14 +2457,14 @@ static int cs47l85_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l85-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l85-dsp-voicectrl") == 0) { n_adsp = 5; - } else if (strcmp(rtd->codec_dai->name, "cs47l85-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l85-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c index 266eade8276488..81a1311b14e643 100644 --- a/sound/soc/codecs/cs47l90.c +++ b/sound/soc/codecs/cs47l90.c @@ -2368,14 +2368,14 @@ static int cs47l90_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l90-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l90-dsp-voicectrl") == 0) { n_adsp = 5; - } else if (strcmp(rtd->codec_dai->name, "cs47l90-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l90-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index 942040fd354f61..15fc213d178dd1 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -1840,12 +1840,12 @@ static int cs47l92_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l92-dsp-trace") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l92-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9dc215b5c5045b..499e87d1dfcc7e 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2245,14 +2245,14 @@ static int wm5110_open(struct snd_compr_stream *stream) struct arizona *arizona = priv->core.arizona; int n_adsp; - if (strcmp(rtd->codec_dai->name, "wm5110-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "wm5110-dsp-voicectrl") == 0) { n_adsp = 2; - } else if (strcmp(rtd->codec_dai->name, "wm5110-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "wm5110-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(arizona->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ffb9836e05383a..1ef69409ccd1e0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3467,22 +3467,22 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) if (wm_adsp_fw[dsp->fw].num_caps == 0) { adsp_err(dsp, "%s: Firmware does not support compressed API\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); ret = -ENXIO; goto out; } if (wm_adsp_fw[dsp->fw].compr_direction != stream->direction) { adsp_err(dsp, "%s: Firmware does not support stream direction\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); ret = -EINVAL; goto out; } list_for_each_entry(tmp, &dsp->compr_list, list) { - if (!strcmp(tmp->name, rtd->codec_dai->name)) { + if (!strcmp(tmp->name, asoc_rtd_to_codec(rtd, 0)->name)) { adsp_err(dsp, "%s: Only a single stream supported per dai\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); ret = -EBUSY; goto out; } @@ -3496,7 +3496,7 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) compr->dsp = dsp; compr->stream = stream; - compr->name = rtd->codec_dai->name; + compr->name = asoc_rtd_to_codec(rtd, 0)->name; list_add_tail(&compr->list, &dsp->compr_list); From 004bd4163104e4d8b6c1433b31ead10a69c69845 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:50:17 -0500 Subject: [PATCH 390/415] ASoC: soc-acpi: expand description of _ADR-based devices For SoundWire, we need to know if endpoints needs to be 'aggregated' (MIPI parlance, meaning logically grouped), e.g. when two speaker amplifiers need to be handled as a single logical output. We don't necessarily have the information at the firmware (BIOS) level, so add a notion of endpoints and specify if a device/endpoint is part of a group, with a position. This may be expanded in future solutions, for now only provide a group and position information. Since we modify the header file, change all existing upstream tables as well to avoid breaking compilation/bisect. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325215027.28716-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 39 ++++++-- .../intel/common/soc-acpi-intel-cml-match.c | 87 +++++++++++++---- .../intel/common/soc-acpi-intel-icl-match.c | 97 +++++++++++++++---- .../intel/common/soc-acpi-intel-tgl-match.c | 49 ++++++++-- 4 files changed, 221 insertions(+), 51 deletions(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index a217a87cae86fb..392e953d561e42 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -75,18 +75,45 @@ struct snd_soc_acpi_mach_params { }; /** - * snd_soc_acpi_link_adr: ACPI-based list of _ADR, with a variable - * number of devices per link - * + * snd_soc_acpi_endpoint - endpoint descriptor + * @num: endpoint number (mandatory, unique per device) + * @aggregated: 0 (independent) or 1 (logically grouped) + * @group_position: zero-based order (only when @aggregated is 1) + * @group_id: platform-unique group identifier (only when @aggregrated is 1) + */ +struct snd_soc_acpi_endpoint { + u8 num; + u8 aggregated; + u8 group_position; + u8 group_id; +}; + +/** + * snd_soc_acpi_adr_device - descriptor for _ADR-enumerated device + * @adr: 64 bit ACPI _ADR value + * @num_endpoints: number of endpoints for this device + * @endpoints: array of endpoints + */ +struct snd_soc_acpi_adr_device { + const u64 adr; + const u8 num_endpoints; + const struct snd_soc_acpi_endpoint *endpoints; +}; + +/** + * snd_soc_acpi_link_adr - ACPI-based list of _ADR enumerated devices * @mask: one bit set indicates the link this list applies to - * @num_adr: ARRAY_SIZE of adr - * @adr: array of _ADR (represented as u64). + * @num_adr: ARRAY_SIZE of devices + * @adr_d: array of devices + * + * The number of devices per link can be more than 1, e.g. in SoundWire + * multi-drop configurations. */ struct snd_soc_acpi_link_adr { const u32 mask; const u32 num_adr; - const u64 *adr; + const struct snd_soc_acpi_adr_device *adr_d; }; /** diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index f55634c4c2e8eb..3525da79c68ac3 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -59,42 +59,95 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); -static const u64 rt711_0_adr[] = { - 0x000010025D071100 +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, }; -static const u64 rt1308_1_adr[] = { - 0x000110025D130800 +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, }; -static const u64 rt1308_2_adr[] = { - 0x000210025D130800 +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, }; -static const u64 rt715_3_adr[] = { - 0x000310025D071500 +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000010025D071100, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { + { + .adr = 0x000110025D130800, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { + { + .adr = 0x000210025D130800, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { + { + .adr = 0x000110025D130800, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_2_group1_adr[] = { + { + .adr = 0x000210025D130800, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt715_3_adr[] = { + { + .adr = 0x000310025D071500, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; static const struct snd_soc_acpi_link_adr cml_3_in_1_default[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), - .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .num_adr = ARRAY_SIZE(rt1308_1_group1_adr), + .adr_d = rt1308_1_group1_adr, }, { .mask = BIT(2), - .num_adr = ARRAY_SIZE(rt1308_2_adr), - .adr = rt1308_2_adr, + .num_adr = ARRAY_SIZE(rt1308_2_group1_adr), + .adr_d = rt1308_2_group1_adr, }, { .mask = BIT(3), .num_adr = ARRAY_SIZE(rt715_3_adr), - .adr = rt715_3_adr, + .adr_d = rt715_3_adr, }, {} }; @@ -103,17 +156,17 @@ static const struct snd_soc_acpi_link_adr cml_3_in_1_mono_amp[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .adr_d = rt1308_1_adr, }, { .mask = BIT(3), .num_adr = ARRAY_SIZE(rt715_3_adr), - .adr = rt715_3_adr, + .adr_d = rt715_3_adr, }, {} }; diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index 752733013d54a9..a05fc083829e34 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -33,55 +33,112 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines); -static const u64 rt700_0_adr[] = { - 0x000010025D070000 +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, +}; + +static const struct snd_soc_acpi_adr_device rt700_0_adr[] = { + { + .adr = 0x000010025D070000, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; static const struct snd_soc_acpi_link_adr icl_rvp[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt700_0_adr), - .adr = rt700_0_adr, + .adr_d = rt700_0_adr, }, {} }; -static const u64 rt711_0_adr[] = { - 0x000010025D071100 +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000010025D071100, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { + { + .adr = 0x000110025D130800, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; -static const u64 rt1308_1_adr[] = { - 0x000110025D130800 +static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { + { + .adr = 0x000210025D130800, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; -static const u64 rt1308_2_adr[] = { - 0x000210025D130800 +static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { + { + .adr = 0x000110025D130800, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + } }; -static const u64 rt715_3_adr[] = { - 0x000310025D071500 +static const struct snd_soc_acpi_adr_device rt1308_2_group1_adr[] = { + { + .adr = 0x000210025D130800, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt715_3_adr[] = { + { + .adr = 0x000310025D071500, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; static const struct snd_soc_acpi_link_adr icl_3_in_1_default[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), - .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .num_adr = ARRAY_SIZE(rt1308_1_group1_adr), + .adr_d = rt1308_1_group1_adr, }, { .mask = BIT(2), - .num_adr = ARRAY_SIZE(rt1308_2_adr), - .adr = rt1308_2_adr, + .num_adr = ARRAY_SIZE(rt1308_2_group1_adr), + .adr_d = rt1308_2_group1_adr, }, { .mask = BIT(3), .num_adr = ARRAY_SIZE(rt715_3_adr), - .adr = rt715_3_adr, + .adr_d = rt715_3_adr, }, {} }; @@ -90,17 +147,17 @@ static const struct snd_soc_acpi_link_adr icl_3_in_1_mono_amp[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .adr_d = rt1308_1_adr, }, { .mask = BIT(3), .num_adr = ARRAY_SIZE(rt715_3_adr), - .adr = rt715_3_adr, + .adr_d = rt715_3_adr, }, {} }; diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index c15eae402b18cc..3153b44f9053f1 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -14,20 +14,53 @@ static struct snd_soc_acpi_codecs tgl_codecs = { .codecs = {"MX98357A"} }; -static const u64 rt711_0_adr[] = { - 0x000010025D071100 +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, }; -static const u64 rt1308_1_adr[] = { - 0x000120025D130800, - 0x000122025D130800 +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, +}; + +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000010025D071100, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { + { + .adr = 0x000120025D130800, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + }, + { + .adr = 0x000122025D130800, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + } }; static const struct snd_soc_acpi_link_adr tgl_i2s_rt1308[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, {} }; @@ -36,12 +69,12 @@ static const struct snd_soc_acpi_link_adr tgl_rvp[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .adr_d = rt1308_1_adr, }, {} }; From 51dfed1e178a38202960b98f6e29df009a06050f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:50:18 -0500 Subject: [PATCH 391/415] ASoC: SOF: Intel: add SoundWire configuration interface Now that the SoundWire core supports the multi-step initialization, call the relevant APIs. The actual hardware enablement can be done in two places, ideally we'd want to startup the SoundWire IP as soon as possible (while still taking power rail dependencies into account) However when suspend/resume is implemented, the DSP device will be resumed first, and only when the DSP firmware is downloaded/booted would the SoundWire child devices be resumed, so there are only marginal benefits in starting the IP earlier for the first probe. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 13 ++++ sound/soc/sof/intel/hda.c | 120 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 44 ++++++++++++ 3 files changed, 177 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 0633b76dab4910..2ae94ea53122ff 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -399,6 +399,19 @@ int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) /* post fw run operations */ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) { + int ret; + + if (sdev->first_boot) { + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: could not startup SoundWire links\n"); + return ret; + } + } + + hda_sdw_int_enable(sdev, true); + /* re-enable clock gating and power gating */ return hda_dsp_ctrl_clock_power_gating(sdev, true); } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee1edb53840c48..c1fe94800da1fd 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -18,7 +18,9 @@ #include #include +#include #include +#include #include #include #include @@ -34,6 +36,98 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ + sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); +} + +static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + acpi_handle handle; + int ret; + + handle = ACPI_HANDLE(sdev->dev); + + /* save ACPI info for the probe step */ + hdev = sdev->pdata->hw_pdata; + + ret = sdw_intel_acpi_scan(handle, &hdev->info); + if (ret < 0) { + dev_err(sdev->dev, "%s failed\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + struct sdw_intel_res res; + acpi_handle handle; + void *sdw; + + handle = ACPI_HANDLE(sdev->dev); + + hdev = sdev->pdata->hw_pdata; + + memset(&res, 0, sizeof(res)); + + res.mmio_base = sdev->bar[HDA_DSP_BAR]; + res.irq = sdev->ipc_irq; + res.handle = hdev->info.handle; + res.parent = sdev->dev; + + /* + * ops and arg fields are not populated for now, + * they will be needed when the DAI callbacks are + * provided + */ + + /* we could filter links here if needed, e.g for quirks */ + res.count = hdev->info.count; + res.link_mask = hdev->info.link_mask; + + sdw = sdw_intel_probe(&res); + if (!sdw) { + dev_err(sdev->dev, "error: SoundWire probe failed\n"); + return -EINVAL; + } + + /* save context */ + hdev->sdw = sdw; + + return 0; +} + +int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + return sdw_intel_startup(hdev->sdw); +} + +static int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + hda_sdw_int_enable(sdev, false); + + if (hdev->sdw) + sdw_intel_exit(hdev->sdw); + hdev->sdw = NULL; + + return 0; +} +#endif + /* * Debug */ @@ -346,9 +440,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev, static int hda_init_caps(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); + struct snd_sof_pdata *pdata = sdev->pdata; #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hdac_ext_link *hlink; #endif + struct sof_intel_hda_dev *hdev = pdata->hw_pdata; + u32 link_mask; int ret = 0; device_disable_async_suspend(bus->dev); @@ -365,6 +462,27 @@ static int hda_init_caps(struct snd_sof_dev *sdev) return ret; } + /* scan SoundWire capabilities exposed by DSDT */ + ret = hda_sdw_acpi_scan(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire ACPI scan error\n"); + return ret; + } + + link_mask = hdev->info.link_mask; + if (!link_mask) { + /* + * probe/allocated SoundWire resources. + * The hardware configuration takes place in hda_sdw_startup + * after power rails are enabled. + */ + ret = hda_sdw_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire probe error\n"); + return ret; + } + } + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -622,6 +740,8 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) snd_hdac_ext_bus_device_remove(bus); #endif + hda_sdw_exit(sdev); + if (!IS_ERR_OR_NULL(hda->dmic_dev)) platform_device_unregister(hda->dmic_dev); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 2a57fc9f3e5895..928a3432e9e6e1 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -11,6 +11,8 @@ #ifndef __SOF_INTEL_HDA_H #define __SOF_INTEL_HDA_H +#include +#include #include #include #include @@ -436,6 +438,12 @@ struct sof_intel_hda_dev { /* delayed work to enter D0I3 opportunistically */ struct delayed_work d0i3_work; + + /* ACPI information stored between scan and probe steps */ + struct sdw_intel_acpi_info info; + + /* sdw context allocated by SoundWire driver */ + struct sdw_intel_ctx *sdw; }; static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s) @@ -654,6 +662,42 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag); int hda_dsp_trace_release(struct snd_sof_dev *sdev); int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); +/* + * SoundWire support + */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +int hda_sdw_startup(struct snd_sof_dev *sdev); +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); + +#else + +static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ +} + +#endif + /* common dai driver */ extern struct snd_soc_dai_driver skl_dai[]; From f8e25018801548314b3d908315cbf271e8eceba8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:50:19 -0500 Subject: [PATCH 392/415] ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ALH was inserted in the wrong place during integration, add after DMIC to mirror the file used by SOF firmware. No functional change, just text move in the same file to better track changes, if any. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325215027.28716-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/dai-intel.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h index 5f1ef5565be69f..04e48227f542b5 100644 --- a/include/sound/sof/dai-intel.h +++ b/include/sound/sof/dai-intel.h @@ -87,6 +87,15 @@ struct sof_ipc_dai_hda_params { uint32_t link_dma_ch; } __packed; +/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ +struct sof_ipc_dai_alh_params { + struct sof_ipc_hdr hdr; + uint32_t stream_id; + + /* reserved for future use */ + uint32_t reserved[15]; +} __packed; + /* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */ /* This struct is defined per 2ch PDM controller available in the platform. @@ -179,13 +188,4 @@ struct sof_ipc_dai_dmic_params { struct sof_ipc_dai_dmic_pdm_ctrl pdm[0]; } __packed; -/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ -struct sof_ipc_dai_alh_params { - struct sof_ipc_hdr hdr; - uint32_t stream_id; - - /* reserved for future use */ - uint32_t reserved[15]; -} __packed; - #endif From d2c383aa49dce9c5b59930c999099ecc580857fe Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:50:20 -0500 Subject: [PATCH 393/415] ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks These callbacks are invoked when a matching hw_params/hw_free() DAI operation takes place, and will result in IPC operations with the SOF firmware. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 71 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 71 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index c1fe94800da1fd..1f93124a63cd02 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -24,6 +24,7 @@ #include #include #include +#include "../sof-audio.h" #include "../ops.h" #include "hda.h" @@ -38,6 +39,74 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +static int sdw_params_stream(struct device *dev, + struct sdw_intel_stream_params_data *params_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = params_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = params_data->link_id; + int alh_stream_id = params_data->alh_stream_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | (d->id); + config.alh.stream_id = alh_stream_id; + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n", + link_id, d->id, alh_stream_id); + } + + return ret; +} + +static int sdw_free_stream(struct device *dev, + struct sdw_intel_stream_free_data *free_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = free_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = free_data->link_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | d->id; + config.alh.stream_id = 0xFFFF; /* invalid value on purpose */ + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to free stream for link %d dai->id %d\n", + link_id, d->id); + } + + return ret; +} + +static const struct sdw_intel_ops sdw_callback = { + .params_stream = sdw_params_stream, + .free_stream = sdw_free_stream, +}; + void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); @@ -80,6 +149,8 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.irq = sdev->ipc_irq; res.handle = hdev->info.handle; res.parent = sdev->dev; + res.ops = &sdw_callback; + res.dev = sdev->dev; /* * ops and arg fields are not populated for now, From b9ddd81bad19ef7b0955156e7590130127cfdaae Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:50:21 -0500 Subject: [PATCH 394/415] ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect For now we have a limited number of machine driver configurations, and we can detect them based on the link configuration returned after checking hardware and firmware (BIOS) configurations. The link configuration is checked with a link_mask as well as a list of _ADR descriptors for each link. There is a chance that in extreme cases where the BIOS contains too much information we would need to detect which Slave devices actually report as 'attached'. This would be more accurate than static table-based solutions, but it also introduces timing dependencies since we don't know when those devices might become attached, so will only be only be looked at if we see limitations with static methods and the usual quirks based e.g. on DMI information. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Signed-off-by: Rander Wang Link: https://lore.kernel.org/r/20200325215027.28716-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 165 ++++++++++++++++++++++++++++++++++---- 1 file changed, 150 insertions(+), 15 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 1f93124a63cd02..dea3c385b6643d 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -20,6 +20,7 @@ #include #include +#include #include #include #include @@ -136,11 +137,8 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hdev; struct sdw_intel_res res; - acpi_handle handle; void *sdw; - handle = ACPI_HANDLE(sdev->dev); - hdev = sdev->pdata->hw_pdata; memset(&res, 0, sizeof(res)); @@ -180,6 +178,9 @@ int hda_sdw_startup(struct snd_sof_dev *sdev) hdev = sdev->pdata->hw_pdata; + if (!hdev->sdw) + return 0; + return sdw_intel_startup(hdev->sdw); } @@ -536,24 +537,31 @@ static int hda_init_caps(struct snd_sof_dev *sdev) /* scan SoundWire capabilities exposed by DSDT */ ret = hda_sdw_acpi_scan(sdev); if (ret < 0) { - dev_err(sdev->dev, "error: SoundWire ACPI scan error\n"); - return ret; + dev_dbg(sdev->dev, "skipping SoundWire, ACPI scan error\n"); + goto skip_soundwire; } link_mask = hdev->info.link_mask; if (!link_mask) { - /* - * probe/allocated SoundWire resources. - * The hardware configuration takes place in hda_sdw_startup - * after power rails are enabled. - */ - ret = hda_sdw_probe(sdev); - if (ret < 0) { - dev_err(sdev->dev, "error: SoundWire probe error\n"); - return ret; - } + dev_dbg(sdev->dev, "skipping SoundWire, no links enabled\n"); + goto skip_soundwire; } + /* + * probe/allocate SoundWire resources. + * The hardware configuration takes place in hda_sdw_startup + * after power rails are enabled. + * It's entirely possible to have a mix of I2S/DMIC/SoundWire + * devices, so we allocate the resources in all cases. + */ + ret = hda_sdw_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire probe error\n"); + return ret; + } + +skip_soundwire: + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -949,6 +957,123 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) } #endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +/* Check if all Slaves defined on the link can be found */ +static bool link_slaves_found(struct snd_sof_dev *sdev, + const struct snd_soc_acpi_link_adr *link, + struct sdw_intel_ctx *sdw) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct sdw_intel_slave_id *ids = sdw->ids; + int num_slaves = sdw->num_slaves; + unsigned int part_id, link_id, unique_id, mfg_id; + int i, j; + + for (i = 0; i < link->num_adr; i++) { + u64 adr = link->adr_d[i].adr; + + mfg_id = SDW_MFG_ID(adr); + part_id = SDW_PART_ID(adr); + link_id = SDW_DISCO_LINK_ID(adr); + for (j = 0; j < num_slaves; j++) { + if (ids[j].link_id != link_id || + ids[j].id.part_id != part_id || + ids[j].id.mfg_id != mfg_id) + continue; + /* + * we have to check unique id + * if there is more than one + * Slave on the link + */ + unique_id = SDW_UNIQUE_ID(adr); + if (link->num_adr == 1 || + ids[j].id.unique_id == SDW_IGNORED_UNIQUE_ID || + ids[j].id.unique_id == unique_id) { + dev_dbg(bus->dev, + "found %x at link %d\n", + part_id, link_id); + break; + } + } + if (j == num_slaves) { + dev_dbg(bus->dev, + "Slave %x not found\n", + part_id); + return false; + } + } + return true; +} + +static int hda_sdw_machine_select(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *pdata = sdev->pdata; + const struct snd_soc_acpi_link_adr *link; + struct hdac_bus *bus = sof_to_bus(sdev); + struct snd_soc_acpi_mach *mach; + struct sof_intel_hda_dev *hdev; + u32 link_mask; + int i; + + hdev = pdata->hw_pdata; + link_mask = hdev->info.link_mask; + + /* + * Select SoundWire machine driver if needed using the + * alternate tables. This case deals with SoundWire-only + * machines, for mixed cases with I2C/I2S the detection relies + * on the HID list. + */ + if (link_mask && !pdata->machine) { + for (mach = pdata->desc->alt_machines; + mach && mach->link_mask; mach++) { + if (mach->link_mask != link_mask) + continue; + + /* No need to match adr if there is no links defined */ + if (!mach->links) + break; + + link = mach->links; + for (i = 0; i < hdev->info.count && link->num_adr; + i++, link++) { + /* + * Try next machine if any expected Slaves + * are not found on this link. + */ + if (!link_slaves_found(sdev, link, hdev->sdw)) + break; + } + /* Found if all Slaves are checked */ + if (i == hdev->info.count || !link->num_adr) + break; + } + if (mach && mach->link_mask) { + dev_dbg(bus->dev, + "SoundWire machine driver %s topology %s\n", + mach->drv_name, + mach->sof_tplg_filename); + pdata->machine = mach; + mach->mach_params.links = mach->links; + mach->mach_params.link_mask = mach->link_mask; + mach->mach_params.platform = dev_name(sdev->dev); + pdata->fw_filename = mach->sof_fw_filename; + pdata->tplg_filename = mach->sof_tplg_filename; + } else { + dev_info(sdev->dev, + "No SoundWire machine driver found\n"); + } + } + + return 0; +} +#else +static int hda_sdw_machine_select(struct snd_sof_dev *sdev) +{ + return 0; +} +#endif + void hda_set_mach_params(const struct snd_soc_acpi_mach *mach, struct device *dev) { @@ -968,8 +1093,18 @@ void hda_machine_select(struct snd_sof_dev *sdev) if (mach) { sof_pdata->tplg_filename = mach->sof_tplg_filename; sof_pdata->machine = mach; + + if (mach->link_mask) { + mach->mach_params.links = mach->links; + mach->mach_params.link_mask = mach->link_mask; + } } + /* + * If I2S fails, try SoundWire + */ + hda_sdw_machine_select(sdev); + /* * Choose HDA generic machine driver if mach is NULL. * Otherwise, set certain mach params. From 3eadff5639b01c17f5f2ffeb209d05cc19706687 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:50:22 -0500 Subject: [PATCH 395/415] ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend Doing this avoid conflicts and errors reported on the bus. The interrupts are only re-enabled on resume after the firmware is downloaded, so the behavior is not fully symmetric Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index c396b7ef0328d9..1aff90042694cd 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -594,6 +594,8 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #endif int ret; + hda_sdw_int_enable(sdev, false); + /* disable IPC interrupts */ hda_dsp_ipc_int_disable(sdev); From 722ba5f1f530a919d28a0dd9e8e0ec63af18270d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 25 Mar 2020 16:50:23 -0500 Subject: [PATCH 396/415] ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers We have a single irq handler for SOF interrupts. We can further merge SoundWire ones to completely remove MSI interrupts handling issues leading to timeouts. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20200325215027.28716-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++++++++++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index dea3c385b6643d..ee4f1ceca88396 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -198,6 +198,38 @@ static int hda_sdw_exit(struct snd_sof_dev *sdev) return 0; } + +static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + bool ret = false; + u32 irq_status; + + hdev = sdev->pdata->hw_pdata; + + if (!hdev->sdw) + return ret; + + /* store status */ + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2); + + /* invalid message ? */ + if (irq_status == 0xffffffff) + goto out; + + /* SDW message ? */ + if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW) + ret = true; + +out: + return ret; +} + +static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return sdw_intel_thread(irq, context); +} + #endif /* @@ -618,6 +650,7 @@ static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; /* deal with streams and controller first */ if (hda_dsp_check_stream_irq(sdev)) @@ -626,6 +659,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_dsp_check_ipc_irq(sdev)) sof_ops(sdev)->irq_thread(irq, sdev); + if (hda_dsp_check_sdw_irq(sdev)) + hda_dsp_sdw_thread(irq, hdev->sdw); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 928a3432e9e6e1..fc104c5ba006e0 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -232,6 +232,8 @@ #define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10) #define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) +#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5) + /* Intel HD Audio Inter-Processor Communication Registers */ #define HDA_DSP_IPC_BASE 0x40 #define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00) @@ -696,6 +698,15 @@ static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { } +static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + return false; +} + +static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return IRQ_HANDLED; +} #endif /* common dai driver */ From 02df8f4364b070428af0e5b6c8739c884c8ad4e7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:50:24 -0500 Subject: [PATCH 397/415] ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks Add module parameter so that the different modes can be quickly tested. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325215027.28716-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee4f1ceca88396..1e69cfcee8e0aa 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -40,6 +40,16 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +/* + * The default for SoundWire clock stop quirks is to power gate the IP + * and do a Bus Reset, this will need to be modified when the DSP + * needs to remain in D0i3 so that the Master does not lose context + * and enumeration is not required on clock restart + */ +static int sdw_clock_stop_quirks = SDW_INTEL_CLK_STOP_BUS_RESET; +module_param(sdw_clock_stop_quirks, int, 0444); +MODULE_PARM_DESC(sdw_clock_stop_quirks, "SOF SoundWire clock stop quirks"); + static int sdw_params_stream(struct device *dev, struct sdw_intel_stream_params_data *params_data) { @@ -149,6 +159,7 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.parent = sdev->dev; res.ops = &sdw_callback; res.dev = sdev->dev; + res.clock_stop_quirks = sdw_clock_stop_quirks; /* * ops and arg fields are not populated for now, From bbd19cdca8279cf244a301c6a13ae5ec9e4ef976 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Wed, 25 Mar 2020 16:50:25 -0500 Subject: [PATCH 398/415] ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire When a SoundWire link is in clock stop state, a Slave device may wake up the Master for some events such as jack detection. The WAKEEN interrupt will be triggered and processed by the audio pci device. If audio device is in D3, the interrupt will be routed to PME, or aggregated at cAVS level as interrupt when audio device is in D0. This patch only supports D3 case, where the audio pci device will be resumed by a PME event and the WAKEEN interrupt will be processed after audio pci device is powered up and ROM is initialized successfully. The WAKEEN handling is only enabled after the first boot due to dependencies on a shim_lock mutex being initialized. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Rander Wang Link: https://lore.kernel.org/r/20200325215027.28716-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 18 ++++++++++++++++++ sound/soc/sof/intel/hda.c | 11 +++++++++++ sound/soc/sof/intel/hda.h | 5 +++++ 3 files changed, 34 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 2ae94ea53122ff..e1550ccd0a49b4 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -346,6 +346,24 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) goto cleanup; } + /* + * When a SoundWire link is in clock stop state, a Slave + * device may trigger in-band wakes for events such as jack + * insertion or acoustic event detection. This event will lead + * to a WAKEEN interrupt, handled by the PCI device and routed + * to PME if the PCI device is in D3. The resume function in + * audio PCI driver will be invoked by ACPI for PME event and + * initialize the device and process WAKEEN interrupt. + * + * The WAKEEN interrupt should be processed ASAP to prevent an + * interrupt flood, otherwise other interrupts, such IPC, + * cannot work normally. The WAKEEN is handled after the ROM + * is initialized successfully, which ensures power rails are + * enabled before accessing the SoundWire SHIM registers + */ + if (!sdev->first_boot) + hda_sdw_process_wakeen(sdev); + /* * at this point DSP ROM has been initialized and * should be ready for code loading and firmware boot diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 1e69cfcee8e0aa..7d1aa4c7d82c89 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -241,6 +241,17 @@ static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) return sdw_intel_thread(irq, context); } +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + if (!hdev->sdw) + return; + + sdw_intel_process_wakeen_event(hdev->sdw); +} + #endif /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index fc104c5ba006e0..6f1765b1ed1de5 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -671,6 +671,7 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); int hda_sdw_startup(struct snd_sof_dev *sdev); void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev); #else @@ -707,6 +708,10 @@ static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) { return IRQ_HANDLED; } + +static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ +} #endif /* common dai driver */ From 90de3281c86ae5378e951e84c76c4759390ff34d Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Wed, 25 Mar 2020 16:50:26 -0500 Subject: [PATCH 399/415] Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread If pci device is in D0, wakeen interrupt will be aggregated at cAVS level as interrupt. This commit check the wakeen status and process it in irq thread Signed-off-by: Pierre-Louis Bossart Signed-off-by: Rander Wang Link: https://lore.kernel.org/r/20200325215027.28716-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 16 ++++++++++++++++ sound/soc/sof/intel/hda.h | 6 ++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7d1aa4c7d82c89..211e91e79eae02 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -241,6 +241,19 @@ static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) return sdw_intel_thread(irq, context); } +static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + if (hdev->sdw && + snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_SNDW_WAKE_STS)) + return true; + + return false; +} + void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hdev; @@ -684,6 +697,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_dsp_check_sdw_irq(sdev)) hda_dsp_sdw_thread(irq, hdev->sdw); + if (hda_sdw_check_wakeen_irq(sdev)) + hda_sdw_process_wakeen(sdev); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 6f1765b1ed1de5..e9825798de7754 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -233,6 +233,7 @@ #define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) #define HDA_DSP_REG_ADSPIS2_SNDW BIT(5) +#define HDA_DSP_REG_SNDW_WAKE_STS 0x2C192 /* Intel HD Audio Inter-Processor Communication Registers */ #define HDA_DSP_IPC_BASE 0x40 @@ -709,6 +710,11 @@ static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) return IRQ_HANDLED; } +static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) +{ + return false; +} + static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) { } From f09e9c7f6331a5a8a5f48ac3d118b641210cbd16 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 16:50:27 -0500 Subject: [PATCH 400/415] ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities Without this cycle, HDaudio capability parsing fails on some devices. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325215027.28716-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ctrl.c | 25 +++++++++++++++++++++---- 1 file changed, 21 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index f88dbcc4ba66a6..6288b2f9954047 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -65,15 +65,32 @@ int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev) struct hdac_bus *bus = sof_to_bus(sdev); u32 cap, offset, feature; int count = 0; + int ret; + + /* + * On some devices, one reset cycle is necessary before reading + * capabilities + */ + ret = hda_dsp_ctrl_link_reset(sdev, true); + if (ret < 0) + return ret; + ret = hda_dsp_ctrl_link_reset(sdev, false); + if (ret < 0) + return ret; offset = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_LLCH); do { - cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset); - dev_dbg(sdev->dev, "checking for capabilities at offset 0x%x\n", offset & SOF_HDA_CAP_NEXT_MASK); + cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset); + + if (cap == -1) { + dev_dbg(bus->dev, "Invalid capability reg read\n"); + break; + } + feature = (cap & SOF_HDA_CAP_ID_MASK) >> SOF_HDA_CAP_ID_OFF; switch (feature) { @@ -106,8 +123,8 @@ int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev) bus->mlcap = bus->remap_addr + offset; break; default: - dev_vdbg(sdev->dev, "found capability %d at 0x%x\n", - feature, offset); + dev_dbg(sdev->dev, "found capability %d at 0x%x\n", + feature, offset); break; } From 17fb5433150e8b0b4000a77a21055359a2eab534 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Thu, 26 Mar 2020 22:10:10 +0100 Subject: [PATCH 401/415] ASoC: pxa: magician: convert to use i2c_new_client_device() Move away from the deprecated API and return the shiny new ERRPTR where useful. Signed-off-by: Wolfram Sang Link: https://lore.kernel.org/r/20200326211010.13471-2-wsa+renesas@sang-engineering.com Signed-off-by: Mark Brown --- sound/soc/pxa/magician.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 6483cff5b73da1..3bafd86bfb938a 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -358,10 +358,10 @@ static int __init magician_init(void) adapter = i2c_get_adapter(0); if (!adapter) return -ENODEV; - client = i2c_new_device(adapter, i2c_board_info); + client = i2c_new_client_device(adapter, i2c_board_info); i2c_put_adapter(adapter); - if (!client) - return -ENODEV; + if (IS_ERR(client)) + return PTR_ERR(client); ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); if (ret) From 914f674bec6efe42f9d6b036850a618fd1698290 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Fri, 27 Mar 2020 15:38:49 +0800 Subject: [PATCH 402/415] ASoC: rt5682: move DAI clock registry to I2S mode The SoundWire mode doesn't need the DAI clocks. Therefore, the DAI clock registry moves to I2S mode case. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20200327073849.18291-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 39 +++++++++++++++++++-------------------- 1 file changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 923541a52504e7..ce4fe7a683f93f 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2856,26 +2856,6 @@ static int rt5682_probe(struct snd_soc_component *component) #endif rt5682->component = component; -#ifdef CONFIG_COMMON_CLK - /* Check if MCLK provided */ - rt5682->mclk = devm_clk_get(component->dev, "mclk"); - if (IS_ERR(rt5682->mclk)) { - if (PTR_ERR(rt5682->mclk) != -ENOENT) { - ret = PTR_ERR(rt5682->mclk); - return ret; - } - rt5682->mclk = NULL; - } - - /* Register CCF DAI clock control */ - ret = rt5682_register_dai_clks(component); - if (ret) - return ret; - - /* Initial setup for CCF */ - rt5682->lrck[RT5682_AIF1] = CLK_48; -#endif - if (rt5682->is_sdw) { slave = rt5682->slave; time = wait_for_completion_timeout( @@ -2885,6 +2865,25 @@ static int rt5682_probe(struct snd_soc_component *component) dev_err(&slave->dev, "Initialization not complete, timed out\n"); return -ETIMEDOUT; } + } else { +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) { + if (PTR_ERR(rt5682->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682->mclk); + return ret; + } + rt5682->mclk = NULL; + } else { + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(component); + if (ret) + return ret; + } + /* Initial setup for CCF */ + rt5682->lrck[RT5682_AIF1] = CLK_48; +#endif } return 0; From ba762e67c3f34660f6ac8299296dbc9e96720939 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Wed, 25 Mar 2020 17:07:43 -0500 Subject: [PATCH 403/415] ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms Update topology and reflect change to unified machine driver for SoundWire. Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200325220746.29601-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-cml-match.c | 24 ++++++++++++++++--- .../intel/common/soc-acpi-intel-icl-match.c | 6 ++--- .../intel/common/soc-acpi-intel-tgl-match.c | 6 ++--- 3 files changed, 27 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 3525da79c68ac3..bcedec6c6117cb 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -80,6 +80,23 @@ static const struct snd_soc_acpi_endpoint spk_r_endpoint = { .group_id = 1, }; +static const struct snd_soc_acpi_adr_device rt700_1_adr[] = { + { + .adr = 0x000110025D070000, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_link_adr cml_rvp[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt700_1_adr), + .adr_d = rt700_1_adr, + }, + {} +}; + static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { { .adr = 0x000010025D071100, @@ -175,7 +192,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[] = { { .link_mask = 0xF, /* 4 active links required */ .links = cml_3_in_1_default, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt711-rt1308-rt715.tplg", }, @@ -187,13 +204,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[] = { */ .link_mask = 0xF, .links = cml_3_in_1_mono_amp, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt711-rt1308-mono-rt715.tplg", }, { .link_mask = 0x2, /* RT700 connected on Link1 */ - .drv_name = "sdw_rt700", + .links = cml_rvp, + .drv_name = "sof_sdw", .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt700.tplg", }, diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index a05fc083829e34..ef8500349f2f2f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -166,21 +166,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[] = { { .link_mask = 0xF, /* 4 active links required */ .links = icl_3_in_1_default, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-icl.ri", .sof_tplg_filename = "sof-icl-rt711-rt1308-rt715.tplg", }, { .link_mask = 0xB, /* 3 active links required */ .links = icl_3_in_1_mono_amp, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-icl.ri", .sof_tplg_filename = "sof-icl-rt711-rt1308-rt715-mono.tplg", }, { .link_mask = 0x1, /* rt700 connected on link0 */ .links = icl_rvp, - .drv_name = "sdw_rt700", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-icl.ri", .sof_tplg_filename = "sof-icl-rt700.tplg", }, diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 3153b44f9053f1..db360c9a8e5bfe 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -87,11 +87,11 @@ static struct snd_soc_acpi_codecs tgl_max98373_amp = { struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", - .drv_name = "rt711_rt1308", + .drv_name = "sof_sdw", .link_mask = 0x1, /* RT711 on SoundWire link0 */ .links = tgl_i2s_rt1308, .sof_fw_filename = "sof-tgl.ri", - .sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg", + .sof_tplg_filename = "sof-tgl-rt711-i2s-rt1308.tplg", }, { .id = "10EC5682", @@ -118,7 +118,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { { .link_mask = 0x3, /* rt711 on link 0 and 2 rt1308s on link 1 */ .links = tgl_rvp, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg", }, From 52db12d193d45728e738df6119702cba08fd46a2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 25 Mar 2020 17:07:44 -0500 Subject: [PATCH 404/415] ASoC: Intel: boards: add sof_sdw machine driver This machine driver provides support for different configurations: RT700, RT711, RT1308 (1x and 2x, I2S or SoundWire mode), and RT715 CometLake, Icelake, TigerLake. PDM digital microphones HDMI To avoid introducing one driver per configuration, this common machine driver relies on platform-specific information, tables and quirks to dynamically create the relevant dailinks. Unlike a lot of machine drivers, we use different DAI links for SoundWire capture and playback since the Cadence PDIs can do capture OR playback, not both simultaneously. For each configuration, the card component string is updated so that UCM can select the relevant parts. Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20200325220746.29601-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 23 + sound/soc/intel/boards/Makefile | 7 +- sound/soc/intel/boards/sof_sdw.c | 944 ++++++++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 108 +++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 ++ sound/soc/intel/boards/sof_sdw_hdmi.c | 97 +++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 ++++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 ++++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 ++++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 ++ 10 files changed, 1693 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f18dd9fde97315..4110ae5db65f22 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -524,4 +524,27 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH endif ## SND_SOC_SOF_JASPERLAKE +if SND_SOC_SOF_INTEL_SOUNDWIRE + +config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH + tristate "SoundWire generic machine driver" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST + depends on SOUNDWIRE + depends on SND_HDA_CODEC_HDMI + select SND_SOC_RT700_SDW + select SND_SOC_RT711_SDW + select SND_SOC_RT1308_SDW + select SND_SOC_RT1308 + select SND_SOC_RT715_SDW + select SND_SOC_DMIC + help + Add support for Intel SoundWire-based platforms connected to + RT700, RT711, RT1308 and RT715 + If unsure select "N". + +endif + + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index e083ceeccdad63..c4ff5166a04269 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -31,7 +31,10 @@ snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o hda_dsp_c snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o snd-soc-sof_da7219_max98373-objs := sof_da7219_max98373.o hda_dsp_common.o - +snd-soc-sof-sdw-objs += sof_sdw.o \ + sof_sdw_rt711.o sof_sdw_rt700.o \ + sof_sdw_rt1308.o sof_sdw_rt715.o \ + sof_sdw_dmic.o sof_sdw_hdmi.o hda_dsp_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -64,4 +67,4 @@ obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max9 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH) += snd-soc-sof_da7219_max98373.o - +obj-$(CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH) += snd-soc-sof-sdw.o diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c new file mode 100644 index 00000000000000..8ed6d2079deecf --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw.c @@ -0,0 +1,944 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw - ASOC Machine driver for Intel SoundWire platforms + */ + +#include +#include +#include +#include +#include +#include +#include +#include "sof_sdw_common.h" + +unsigned long sof_sdw_quirk = SOF_RT711_JD_SRC_JD1; + +#define INC_ID(BE, CPU, LINK) do { (BE)++; (CPU)++; (LINK)++; } while (0) + +static int sof_sdw_quirk_cb(const struct dmi_system_id *id) +{ + sof_sdw_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_sdw_quirk_table[] = { + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "09C6") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX), + }, + { + /* early version of SKU 09C6 */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0983") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, + "Tiger Lake Client Platform"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 | + SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | + SOF_SSP_PORT(SOF_I2S_SSP2)), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), + }, + .driver_data = (void *)SOF_SDW_PCH_DMIC, + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "CometLake Client"), + }, + .driver_data = (void *)SOF_SDW_PCH_DMIC, + }, + + {} +}; + +static struct snd_soc_codec_conf codec_conf[] = { + { + .dlc = COMP_CODEC_CONF("sdw:0:25d:711:0"), + .name_prefix = "rt711", + }, + /* rt1308 w/ I2S connection */ + { + .dlc = COMP_CODEC_CONF("i2c-10EC1308:00"), + .name_prefix = "rt1308-1", + }, + /* rt1308 left on link 1 */ + { + .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0"), + .name_prefix = "rt1308-1", + }, + /* two 1308s on link1 with different unique id */ + { + .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0:0"), + .name_prefix = "rt1308-1", + }, + { + .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0:2"), + .name_prefix = "rt1308-2", + }, + /* rt1308 right on link 2 */ + { + .dlc = COMP_CODEC_CONF("sdw:2:25d:1308:0"), + .name_prefix = "rt1308-2", + }, + { + .dlc = COMP_CODEC_CONF("sdw:3:25d:715:0"), + .name_prefix = "rt715", + }, +}; + +static struct snd_soc_dai_link_component dmic_component[] = { + { + .name = "dmic-codec", + .dai_name = "dmic-hifi", + } +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +/* these wrappers are only needed to avoid typecast compilation errors */ +static int sdw_startup(struct snd_pcm_substream *substream) +{ + return sdw_startup_stream(substream); +} + +static void sdw_shutdown(struct snd_pcm_substream *substream) +{ + sdw_shutdown_stream(substream); +} + +static const struct snd_soc_ops sdw_ops = { + .startup = sdw_startup, + .shutdown = sdw_shutdown, +}; + +static struct sof_sdw_codec_info codec_info_list[] = { + { + .id = 0x700, + .direction = {true, true}, + .dai_name = "rt700-aif1", + .init = sof_sdw_rt700_init, + }, + { + .id = 0x711, + .direction = {true, true}, + .dai_name = "rt711-aif1", + .init = sof_sdw_rt711_init, + }, + { + .id = 0x1308, + .acpi_id = "10EC1308", + .direction = {true, false}, + .dai_name = "rt1308-aif", + .ops = &sof_sdw_rt1308_i2s_ops, + .init = sof_sdw_rt1308_init, + }, + { + .id = 0x715, + .direction = {false, true}, + .dai_name = "rt715-aif2", + .init = sof_sdw_rt715_init, + }, +}; + +static inline int find_codec_info_part(unsigned int part_id) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + if (part_id == codec_info_list[i].id) + break; + + if (i == ARRAY_SIZE(codec_info_list)) + return -EINVAL; + + return i; +} + +static inline int find_codec_info_acpi(const u8 *acpi_id) +{ + int i; + + if (!acpi_id[0]) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + if (!memcmp(codec_info_list[i].acpi_id, acpi_id, + ACPI_ID_LEN)) + break; + + if (i == ARRAY_SIZE(codec_info_list)) + return -EINVAL; + + return i; +} + +/* + * get BE dailink number and CPU DAI number based on sdw link adr. + * Since some sdw slaves may be aggregated, the CPU DAI number + * may be larger than the number of BE dailinks. + */ +static int get_sdw_dailink_info(const struct snd_soc_acpi_link_adr *links, + int *sdw_be_num, int *sdw_cpu_dai_num) +{ + const struct snd_soc_acpi_link_adr *link; + bool group_visited[SDW_MAX_GROUPS]; + bool no_aggregation; + int i; + + no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION; + *sdw_cpu_dai_num = 0; + *sdw_be_num = 0; + + if (!links) + return -EINVAL; + + for (i = 0; i < SDW_MAX_GROUPS; i++) + group_visited[i] = false; + + for (link = links; link->num_adr; link++) { + const struct snd_soc_acpi_endpoint *endpoint; + int part_id, codec_index; + int stream; + u64 adr; + + adr = link->adr_d->adr; + part_id = SDW_PART_ID(adr); + codec_index = find_codec_info_part(part_id); + if (codec_index < 0) + return codec_index; + + endpoint = link->adr_d->endpoints; + + /* count DAI number for playback and capture */ + for_each_pcm_streams(stream) { + if (!codec_info_list[codec_index].direction[stream]) + continue; + + (*sdw_cpu_dai_num)++; + + /* count BE for each non-aggregated slave or group */ + if (!endpoint->aggregated || no_aggregation || + !group_visited[endpoint->group_id]) + (*sdw_be_num)++; + } + + if (endpoint->aggregated) + group_visited[endpoint->group_id] = true; + } + + return 0; +} + +static void init_dai_link(struct snd_soc_dai_link *dai_links, int be_id, + char *name, int playback, int capture, + struct snd_soc_dai_link_component *cpus, + int cpus_num, + struct snd_soc_dai_link_component *codecs, + int codecs_num, + int (*init)(struct snd_soc_pcm_runtime *rtd), + const struct snd_soc_ops *ops) +{ + dai_links->id = be_id; + dai_links->name = name; + dai_links->platforms = platform_component; + dai_links->num_platforms = ARRAY_SIZE(platform_component); + dai_links->nonatomic = true; + dai_links->no_pcm = 1; + dai_links->cpus = cpus; + dai_links->num_cpus = cpus_num; + dai_links->codecs = codecs; + dai_links->num_codecs = codecs_num; + dai_links->dpcm_playback = playback; + dai_links->dpcm_capture = capture; + dai_links->init = init; + dai_links->ops = ops; +} + +static bool is_unique_device(const struct snd_soc_acpi_link_adr *link, + unsigned int sdw_version, + unsigned int mfg_id, + unsigned int part_id, + unsigned int class_id, + int index_in_link + ) +{ + int i; + + for (i = 0; i < link->num_adr; i++) { + unsigned int sdw1_version, mfg1_id, part1_id, class1_id; + u64 adr; + + /* skip itself */ + if (i == index_in_link) + continue; + + adr = link->adr_d[i].adr; + + sdw1_version = SDW_VERSION(adr); + mfg1_id = SDW_MFG_ID(adr); + part1_id = SDW_PART_ID(adr); + class1_id = SDW_CLASS_ID(adr); + + if (sdw_version == sdw1_version && + mfg_id == mfg1_id && + part_id == part1_id && + class_id == class1_id) + return false; + } + + return true; +} + +static int create_codec_dai_name(struct device *dev, + const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link_component *codec, + int offset) +{ + int i; + + for (i = 0; i < link->num_adr; i++) { + unsigned int sdw_version, unique_id, mfg_id; + unsigned int link_id, part_id, class_id; + int codec_index, comp_index; + char *codec_str; + u64 adr; + + adr = link->adr_d[i].adr; + + sdw_version = SDW_VERSION(adr); + link_id = SDW_DISCO_LINK_ID(adr); + unique_id = SDW_UNIQUE_ID(adr); + mfg_id = SDW_MFG_ID(adr); + part_id = SDW_PART_ID(adr); + class_id = SDW_CLASS_ID(adr); + + comp_index = i + offset; + if (is_unique_device(link, sdw_version, mfg_id, part_id, + class_id, i)) { + codec_str = "sdw:%x:%x:%x:%x"; + codec[comp_index].name = + devm_kasprintf(dev, GFP_KERNEL, codec_str, + link_id, mfg_id, part_id, + class_id); + } else { + codec_str = "sdw:%x:%x:%x:%x:%x"; + codec[comp_index].name = + devm_kasprintf(dev, GFP_KERNEL, codec_str, + link_id, mfg_id, part_id, + class_id, unique_id); + } + + if (!codec[comp_index].name) + return -ENOMEM; + + codec_index = find_codec_info_part(part_id); + if (codec_index < 0) + return codec_index; + + codec[comp_index].dai_name = + codec_info_list[codec_index].dai_name; + } + + return 0; +} + +static int set_codec_init_func(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + bool playback) +{ + int i; + + for (i = 0; i < link->num_adr; i++) { + unsigned int part_id; + int codec_index; + + part_id = SDW_PART_ID(link->adr_d[i].adr); + codec_index = find_codec_info_part(part_id); + + if (codec_index < 0) + return codec_index; + + if (codec_info_list[codec_index].init) + codec_info_list[codec_index].init(link, dai_links, + &codec_info_list[codec_index], + playback); + } + + return 0; +} + +/* + * check endpoint status in slaves and gather link ID for all slaves in + * the same group to generate different CPU DAI. Now only support + * one sdw link with all slaves set with only single group id. + * + * one slave on one sdw link with aggregated = 0 + * one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI + * + * two or more slaves on one sdw link with aggregated = 0 + * one sdw BE DAI <---> one-cpu DAI <---> multi-codec DAIs + * + * multiple links with multiple slaves with aggregated = 1 + * one sdw BE DAI <---> 1 .. N CPU DAIs <----> 1 .. N codec DAIs + */ +static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, + struct device *dev, int *cpu_dai_id, int *cpu_dai_num, + int *codec_num, int *group_id, + bool *group_generated) +{ + const struct snd_soc_acpi_adr_device *adr_d; + const struct snd_soc_acpi_link_adr *adr_next; + bool no_aggregation; + int index = 0; + + no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION; + *codec_num = adr_link->num_adr; + adr_d = adr_link->adr_d; + + /* make sure the link mask has a single bit set */ + if (!is_power_of_2(adr_link->mask)) + return -EINVAL; + + cpu_dai_id[index++] = ffs(adr_link->mask) - 1; + if (!adr_d->endpoints->aggregated || no_aggregation) { + *cpu_dai_num = 1; + *group_id = 0; + return 0; + } + + *group_id = adr_d->endpoints->group_id; + + /* gather other link ID of slaves in the same group */ + for (adr_next = adr_link + 1; adr_next && adr_next->num_adr; + adr_next++) { + const struct snd_soc_acpi_endpoint *endpoint; + + endpoint = adr_next->adr_d->endpoints; + if (!endpoint->aggregated || + endpoint->group_id != *group_id) + continue; + + /* make sure the link mask has a single bit set */ + if (!is_power_of_2(adr_next->mask)) + return -EINVAL; + + if (index >= SDW_MAX_CPU_DAIS) { + dev_err(dev, " cpu_dai_id array overflows"); + return -EINVAL; + } + + cpu_dai_id[index++] = ffs(adr_next->mask) - 1; + *codec_num += adr_next->num_adr; + } + + /* + * indicate CPU DAIs for this group have been generated + * to avoid generating CPU DAIs for this group again. + */ + group_generated[*group_id] = true; + *cpu_dai_num = index; + + return 0; +} + +static int create_sdw_dailink(struct device *dev, int *be_index, + struct snd_soc_dai_link *dai_links, + int sdw_be_num, int sdw_cpu_dai_num, + struct snd_soc_dai_link_component *cpus, + const struct snd_soc_acpi_link_adr *link, + int *cpu_id, bool *group_generated) +{ + const struct snd_soc_acpi_link_adr *link_next; + struct snd_soc_dai_link_component *codecs; + int cpu_dai_id[SDW_MAX_CPU_DAIS]; + int cpu_dai_num, cpu_dai_index; + unsigned int part_id, group_id; + int codec_idx = 0; + int i = 0, j = 0; + int codec_index; + int codec_num; + int stream; + int ret; + int k; + + ret = get_slave_info(link, dev, cpu_dai_id, &cpu_dai_num, &codec_num, + &group_id, group_generated); + if (ret) + return ret; + + codecs = devm_kcalloc(dev, codec_num, sizeof(*codecs), GFP_KERNEL); + if (!codecs) + return -ENOMEM; + + /* generate codec name on different links in the same group */ + for (link_next = link; link_next && link_next->num_adr && + i < cpu_dai_num; link_next++) { + const struct snd_soc_acpi_endpoint *endpoints; + + endpoints = link_next->adr_d->endpoints; + if (group_id && (!endpoints->aggregated || + endpoints->group_id != group_id)) + continue; + + /* skip the link excluded by this processed group */ + if (cpu_dai_id[i] != ffs(link_next->mask) - 1) + continue; + + ret = create_codec_dai_name(dev, link_next, codecs, codec_idx); + if (ret < 0) + return ret; + + /* check next link to create codec dai in the processed group */ + i++; + codec_idx += link_next->num_adr; + } + + /* find codec info to create BE DAI */ + part_id = SDW_PART_ID(link->adr_d[0].adr); + codec_index = find_codec_info_part(part_id); + if (codec_index < 0) + return codec_index; + + cpu_dai_index = *cpu_id; + for_each_pcm_streams(stream) { + char *name, *cpu_name; + int playback, capture; + static const char * const sdw_stream_name[] = { + "SDW%d-Playback", + "SDW%d-Capture", + }; + + if (!codec_info_list[codec_index].direction[stream]) + continue; + + /* create stream name according to first link id */ + name = devm_kasprintf(dev, GFP_KERNEL, + sdw_stream_name[stream], cpu_dai_id[0]); + if (!name) + return -ENOMEM; + + /* + * generate CPU DAI name base on the sdw link ID and + * PIN ID with offset of 2 according to sdw dai driver. + */ + for (k = 0; k < cpu_dai_num; k++) { + cpu_name = devm_kasprintf(dev, GFP_KERNEL, + "SDW%d Pin%d", cpu_dai_id[k], + j + SDW_INTEL_BIDIR_PDI_BASE); + if (!cpu_name) + return -ENOMEM; + + if (cpu_dai_index >= sdw_cpu_dai_num) { + dev_err(dev, "invalid cpu dai index %d", + cpu_dai_index); + return -EINVAL; + } + + cpus[cpu_dai_index++].dai_name = cpu_name; + } + + if (*be_index >= sdw_be_num) { + dev_err(dev, " invalid be dai index %d", *be_index); + return -EINVAL; + } + + if (*cpu_id >= sdw_cpu_dai_num) { + dev_err(dev, " invalid cpu dai index %d", *cpu_id); + return -EINVAL; + } + + playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); + capture = (stream == SNDRV_PCM_STREAM_CAPTURE); + init_dai_link(dai_links + *be_index, *be_index, name, + playback, capture, + cpus + *cpu_id, cpu_dai_num, + codecs, codec_num, + NULL, &sdw_ops); + + ret = set_codec_init_func(link, dai_links + (*be_index)++, + playback); + if (ret < 0) { + dev_err(dev, "failed to init codec %d", codec_index); + return ret; + } + + *cpu_id += cpu_dai_num; + j++; + } + + return 0; +} + +/* + * DAI link ID of SSP & DMIC & HDMI are based on last + * link ID used by sdw link. Since be_id may be changed + * in init func of sdw codec, it is not equal to be_id + */ +static inline int get_next_be_id(struct snd_soc_dai_link *links, + int be_id) +{ + return links[be_id - 1].id + 1; +} + +static int sof_card_dai_links_create(struct device *dev, + struct snd_soc_acpi_mach *mach, + struct snd_soc_card *card) +{ + int ssp_num, sdw_be_num = 0, hdmi_num = 0, dmic_num; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + struct snd_soc_dai_link_component *idisp_components; +#endif + struct snd_soc_dai_link_component *ssp_components; + struct snd_soc_acpi_mach_params *mach_params; + const struct snd_soc_acpi_link_adr *adr_link; + struct snd_soc_dai_link_component *cpus; + bool group_generated[SDW_MAX_GROUPS]; + int ssp_codec_index, ssp_mask; + struct snd_soc_dai_link *links; + int num_links, link_id = 0; + char *name, *cpu_name; + int total_cpu_dai_num; + int sdw_cpu_dai_num; + int i, j, be_id = 0; + int cpu_id = 0; + int comp_num; + int ret; + + /* reset amp_num to ensure amp_num++ starts from 0 in each probe */ + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + codec_info_list[i].amp_num = 0; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + hdmi_num = sof_sdw_quirk & SOF_SDW_TGL_HDMI ? + SOF_TGL_HDMI_COUNT : SOF_PRE_TGL_HDMI_COUNT; +#endif + + ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk); + /* + * on generic tgl platform, I2S or sdw mode is supported + * based on board rework. A ACPI device is registered in + * system only when I2S mode is supported, not sdw mode. + * Here check ACPI ID to confirm I2S is supported. + */ + ssp_codec_index = find_codec_info_acpi(mach->id); + ssp_num = ssp_codec_index >= 0 ? hweight_long(ssp_mask) : 0; + comp_num = hdmi_num + ssp_num; + + mach_params = &mach->mach_params; + ret = get_sdw_dailink_info(mach_params->links, + &sdw_be_num, &sdw_cpu_dai_num); + if (ret < 0) { + dev_err(dev, "failed to get sdw link info %d", ret); + return ret; + } + + /* enable dmic01 & dmic16k */ + dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC) ? 2 : 0; + comp_num += dmic_num; + + dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d", sdw_be_num, ssp_num, + dmic_num, hdmi_num); + + /* allocate BE dailinks */ + num_links = comp_num + sdw_be_num; + links = devm_kcalloc(dev, num_links, sizeof(*links), GFP_KERNEL); + + /* allocated CPU DAIs */ + total_cpu_dai_num = comp_num + sdw_cpu_dai_num; + cpus = devm_kcalloc(dev, total_cpu_dai_num, sizeof(*cpus), + GFP_KERNEL); + + if (!links || !cpus) + return -ENOMEM; + + /* SDW */ + if (!sdw_be_num) + goto SSP; + + adr_link = mach_params->links; + if (!adr_link) + return -EINVAL; + + /* + * SoundWire Slaves aggregated in the same group may be + * located on different hardware links. Clear array to indicate + * CPU DAIs for this group have not been generated. + */ + for (i = 0; i < SDW_MAX_GROUPS; i++) + group_generated[i] = false; + + /* generate DAI links by each sdw link */ + for (; adr_link->num_adr; adr_link++) { + const struct snd_soc_acpi_endpoint *endpoint; + + endpoint = adr_link->adr_d->endpoints; + if (endpoint->aggregated && !endpoint->group_id) { + dev_err(dev, "invalid group id on link %x", + adr_link->mask); + continue; + } + + /* this group has been generated */ + if (endpoint->aggregated && + group_generated[endpoint->group_id]) + continue; + + ret = create_sdw_dailink(dev, &be_id, links, sdw_be_num, + sdw_cpu_dai_num, cpus, adr_link, + &cpu_id, group_generated); + if (ret < 0) { + dev_err(dev, "failed to create dai link %d", be_id); + return -ENOMEM; + } + } + + /* non-sdw DAI follows sdw DAI */ + link_id = be_id; + + /* get BE ID for non-sdw DAI */ + be_id = get_next_be_id(links, be_id); + +SSP: + /* SSP */ + if (!ssp_num) + goto DMIC; + + for (i = 0, j = 0; ssp_mask; i++, ssp_mask >>= 1) { + struct sof_sdw_codec_info *info; + int playback, capture; + char *codec_name; + + if (!(ssp_mask & 0x1)) + continue; + + name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", i); + if (!name) + return -ENOMEM; + + cpu_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); + if (!cpu_name) + return -ENOMEM; + + ssp_components = devm_kzalloc(dev, sizeof(*ssp_components), + GFP_KERNEL); + if (!ssp_components) + return -ENOMEM; + + info = &codec_info_list[ssp_codec_index]; + codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", + info->acpi_id, j++); + if (!codec_name) + return -ENOMEM; + + ssp_components->name = codec_name; + ssp_components->dai_name = info->dai_name; + cpus[cpu_id].dai_name = cpu_name; + + playback = info->direction[SNDRV_PCM_STREAM_PLAYBACK]; + capture = info->direction[SNDRV_PCM_STREAM_CAPTURE]; + init_dai_link(links + link_id, be_id, name, + playback, capture, + cpus + cpu_id, 1, + ssp_components, 1, + NULL, info->ops); + + ret = info->init(NULL, links + link_id, info, 0); + if (ret < 0) + return ret; + + INC_ID(be_id, cpu_id, link_id); + } + +DMIC: + /* dmic */ + if (dmic_num > 0) { + cpus[cpu_id].dai_name = "DMIC01 Pin"; + init_dai_link(links + link_id, be_id, "dmic01", + 0, 1, // DMIC only supports capture + cpus + cpu_id, 1, + dmic_component, 1, + sof_sdw_dmic_init, NULL); + INC_ID(be_id, cpu_id, link_id); + + cpus[cpu_id].dai_name = "DMIC16k Pin"; + init_dai_link(links + link_id, be_id, "dmic16k", + 0, 1, // DMIC only supports capture + cpus + cpu_id, 1, + dmic_component, 1, + /* don't call sof_sdw_dmic_init() twice */ + NULL, NULL); + INC_ID(be_id, cpu_id, link_id); + } + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kcalloc(dev, hdmi_num, + sizeof(*idisp_components), + GFP_KERNEL); + if (!idisp_components) + return -ENOMEM; + } + + for (i = 0; i < hdmi_num; i++) { + name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i + 1); + if (!name) + return -ENOMEM; + + idisp_components[i].name = "ehdaudio0D2"; + idisp_components[i].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i + 1); + if (!idisp_components[i].dai_name) + return -ENOMEM; + + cpu_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i + 1); + if (!cpu_name) + return -ENOMEM; + + cpus[cpu_id].dai_name = cpu_name; + init_dai_link(links + link_id, be_id, name, + 1, 0, // HDMI only supports playback + cpus + cpu_id, 1, + idisp_components + i, 1, + sof_sdw_hdmi_init, NULL); + INC_ID(be_id, cpu_id, link_id); + } +#endif + + card->dai_link = links; + card->num_links = num_links; + + return 0; +} + +/* SoC card */ +static const char sdw_card_long_name[] = "Intel Soundwire SOF"; + +static struct snd_soc_card card_sof_sdw = { + .name = "soundwire", + .late_probe = sof_sdw_hdmi_card_late_probe, + .codec_conf = codec_conf, + .num_configs = ARRAY_SIZE(codec_conf), +}; + +static int mc_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &card_sof_sdw; + struct snd_soc_acpi_mach *mach; + struct mc_private *ctx; + int ret; + + dev_dbg(&pdev->dev, "Entry %s\n", __func__); + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + dmi_check_system(sof_sdw_quirk_table); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); +#endif + + card->dev = &pdev->dev; + + mach = pdev->dev.platform_data; + ret = sof_card_dai_links_create(&pdev->dev, mach, + card); + if (ret < 0) + return ret; + + ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; + + snd_soc_card_set_drvdata(card, ctx); + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "cfg-spk:%d", + (sof_sdw_quirk & SOF_SDW_FOUR_SPK) ? 4 : 2); + if (!card->components) + return -ENOMEM; + + card->long_name = sdw_card_long_name; + + /* Register the card */ + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) { + dev_err(card->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + + platform_set_drvdata(pdev, card); + + return ret; +} + +static struct platform_driver sof_sdw_driver = { + .driver = { + .name = "sof_sdw", + .pm = &snd_soc_pm_ops, + }, + .probe = mc_probe, +}; + +module_platform_driver(sof_sdw_driver); + +MODULE_DESCRIPTION("ASoC SoundWire Generic Machine driver"); +MODULE_AUTHOR("Bard Liao "); +MODULE_AUTHOR("Rander Wang "); +MODULE_AUTHOR("Pierre-Louis Bossart "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_sdw"); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h new file mode 100644 index 00000000000000..0d738e234bc5b1 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -0,0 +1,108 @@ +/* SPDX-License-Identifier: GPL-2.0 + * Copyright (c) 2020 Intel Corporation + */ + +/* + * sof_sdw_common.h - prototypes for common helpers + */ + +#ifndef SND_SOC_SOF_SDW_COMMON_H +#define SND_SOC_SOF_SDW_COMMON_H + +#include +#include + +#define MAX_NO_PROPS 2 +#define MAX_HDMI_NUM 4 +#define SDW_DMIC_DAI_ID 4 +#define SDW_MAX_CPU_DAIS 16 +#define SDW_INTEL_BIDIR_PDI_BASE 2 + +/* 8 combinations with 4 links + unused group 0 */ +#define SDW_MAX_GROUPS 9 + +enum { + SOF_RT711_JD_SRC_JD1 = 1, + SOF_RT711_JD_SRC_JD2 = 2, +}; + +enum { + SOF_PRE_TGL_HDMI_COUNT = 3, + SOF_TGL_HDMI_COUNT = 4, +}; + +enum { + SOF_I2S_SSP0 = BIT(0), + SOF_I2S_SSP1 = BIT(1), + SOF_I2S_SSP2 = BIT(2), + SOF_I2S_SSP3 = BIT(3), + SOF_I2S_SSP4 = BIT(4), + SOF_I2S_SSP5 = BIT(5), +}; + +#define SOF_RT711_JDSRC(quirk) ((quirk) & GENMASK(1, 0)) +#define SOF_SDW_FOUR_SPK BIT(2) +#define SOF_SDW_TGL_HDMI BIT(3) +#define SOF_SDW_PCH_DMIC BIT(4) +#define SOF_SSP_PORT(x) (((x) & GENMASK(5, 0)) << 5) +#define SOF_SSP_GET_PORT(quirk) (((quirk) >> 5) & GENMASK(5, 0)) +#define SOF_RT715_DAI_ID_FIX BIT(11) +#define SOF_SDW_NO_AGGREGATION BIT(12) + +struct sof_sdw_codec_info { + const int id; + int amp_num; + const u8 acpi_id[ACPI_ID_LEN]; + const bool direction[2]; // playback & capture support + const char *dai_name; + const struct snd_soc_ops *ops; + + int (*init)(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); +}; + +struct mc_private { + struct list_head hdmi_pcm_list; + bool common_hdmi_codec_drv; + struct snd_soc_jack sdw_headset; +}; + +extern unsigned long sof_sdw_quirk; + +/* generic HDMI support */ +int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); + +int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card); + +/* DMIC support */ +int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); + +/* RT711 support */ +int sof_sdw_rt711_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +/* RT700 support */ +int sof_sdw_rt700_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +/* RT1308 support */ +extern struct snd_soc_ops sof_sdw_rt1308_i2s_ops; + +int sof_sdw_rt1308_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +/* RT715 support */ +int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +#endif diff --git a/sound/soc/intel/boards/sof_sdw_dmic.c b/sound/soc/intel/boards/sof_sdw_dmic.c new file mode 100644 index 00000000000000..e92176bf0ad40f --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_dmic.c @@ -0,0 +1,42 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_dmic - Helpers to handle dmic from generic machine driver + */ + +#include +#include +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + diff --git a/sound/soc/intel/boards/sof_sdw_hdmi.c b/sound/soc/intel/boards/sof_sdw_hdmi.c new file mode 100644 index 00000000000000..c7b5612a39e6a1 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_hdmi.c @@ -0,0 +1,97 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_hdmi - Helpers to handle HDMI from generic machine driver + */ + +#include +#include +#include +#include +#include +#include +#include +#include "sof_sdw_common.h" +#include "../../codecs/hdac_hdmi.h" +#include "hda_dsp_common.h" + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) +static struct snd_soc_jack hdmi[MAX_HDMI_NUM]; + +struct hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +#define NAME_SIZE 32 +int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card) +{ + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm, + head); + component = pcm->codec_dai->component; + + if (ctx->common_hdmi_codec_drv) + return hda_dsp_hdmi_build_controls(card, component); + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &hdmi[i], + NULL, 0); + + if (err) + return err; + + err = snd_jack_add_new_kctl(hdmi[i].jack, + jack_name, SND_JACK_AVOUT); + if (err) + dev_warn(component->dev, "failed creating Jack kctl\n"); + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} +#else +int hdmi_card_late_probe(struct snd_soc_card *card) +{ + return 0; +} +#endif diff --git a/sound/soc/intel/boards/sof_sdw_rt1308.c b/sound/soc/intel/boards/sof_sdw_rt1308.c new file mode 100644 index 00000000000000..321768e54d08a0 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt1308.c @@ -0,0 +1,151 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt1308 - Helpers to handle RT1308 from generic machine driver + */ + +#include +#include +#include +#include +#include "sof_sdw_common.h" +#include "../../codecs/rt1308.h" + +static const struct snd_soc_dapm_widget rt1308_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +/* + * dapm routes for rt1308 will be registered dynamically according + * to the number of rt1308 used. The first two entries will be registered + * for one codec case, and the last two entries are also registered + * if two 1308s are used. + */ +static const struct snd_soc_dapm_route rt1308_map[] = { + { "Speaker", NULL, "rt1308-1 SPOL" }, + { "Speaker", NULL, "rt1308-1 SPOR" }, + { "Speaker", NULL, "rt1308-2 SPOL" }, + { "Speaker", NULL, "rt1308-2 SPOR" }, +}; + +static const struct snd_kcontrol_new rt1308_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static int first_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s spk:rt1308", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt1308_controls, + ARRAY_SIZE(rt1308_controls)); + if (ret) { + dev_err(card->dev, "rt1308 controls addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt1308_widgets, + ARRAY_SIZE(rt1308_widgets)); + if (ret) { + dev_err(card->dev, "rt1308 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1308_map, 2); + if (ret) + dev_err(rtd->dev, "failed to add first SPK map: %d\n", ret); + + return ret; +} + +static int second_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1308_map + 2, 2); + if (ret) + dev_err(rtd->dev, "failed to add second SPK map: %d\n", ret); + + return ret; +} + +static int all_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + + ret = first_spk_init(rtd); + if (ret) + return ret; + + return second_spk_init(rtd); +} + +static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int clk_id, clk_freq, pll_out; + int err; + + clk_id = RT1308_PLL_S_MCLK; + clk_freq = 38400000; + + pll_out = params_rate(params) * 512; + + /* Set rt1308 pll */ + err = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out); + if (err < 0) { + dev_err(card->dev, "Failed to set RT1308 PLL: %d\n", err); + return err; + } + + /* Set rt1308 sysclk */ + err = snd_soc_dai_set_sysclk(codec_dai, RT1308_FS_SYS_S_PLL, pll_out, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "Failed to set RT1308 SYSCLK: %d\n", err); + return err; + } + + return 0; +} + +/* machine stream operations */ +struct snd_soc_ops sof_sdw_rt1308_i2s_ops = { + .hw_params = rt1308_i2s_hw_params, +}; + +int sof_sdw_rt1308_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + info->amp_num++; + if (info->amp_num == 1) + dai_links->init = first_spk_init; + + if (info->amp_num == 2) { + /* + * if two 1308s are in one dai link, the init function + * in this dai link will be first set for the first speaker, + * and it should be reset to initialize all speakers when + * the second speaker is found. + */ + if (dai_links->init) + dai_links->init = all_spk_init; + else + dai_links->init = second_spk_init; + } + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c new file mode 100644 index 00000000000000..2ee4e6910d7f97 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -0,0 +1,125 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt700 - Helpers to handle RT700 from generic machine driver + */ + +#include +#include +#include +#include +#include +#include +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget rt700_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route rt700_map[] = { + /* Headphones */ + { "Headphones", NULL, "HP" }, + { "Speaker", NULL, "SPK" }, + { "MIC2", NULL, "AMIC" }, +}; + +static const struct snd_kcontrol_new rt700_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("AMIC"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static struct snd_soc_jack_pin rt700_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s hs:rt700", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt700_controls, + ARRAY_SIZE(rt700_controls)); + if (ret) { + dev_err(card->dev, "rt700 controls addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt700_widgets, + ARRAY_SIZE(rt700_widgets)); + if (ret) { + dev_err(card->dev, "rt700 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt700_map, + ARRAY_SIZE(rt700_map)); + + if (ret) { + dev_err(card->dev, "rt700 map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sdw_headset, + rt700_jack_pins, + ARRAY_SIZE(rt700_jack_pins)); + if (ret) { + dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n", + ret); + return ret; + } + + jack = &ctx->sdw_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, jack, NULL); + if (ret) + dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n", + ret); + + return ret; +} + +int sof_sdw_rt700_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* + * headset should be initialized once. + * Do it with dai link for playback. + */ + if (!playback) + return 0; + + dai_links->init = rt700_rtd_init; + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c new file mode 100644 index 00000000000000..2a4917e3d56148 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -0,0 +1,156 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt711 - Helpers to handle RT711 from generic machine driver + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "sof_sdw_common.h" + +/* + * Note this MUST be called before snd_soc_register_card(), so that the props + * are in place before the codec component driver's probe function parses them. + */ +static int rt711_add_codec_device_props(const char *sdw_dev_name) +{ + struct property_entry props[MAX_NO_PROPS] = {}; + struct device *sdw_dev; + int ret; + + sdw_dev = bus_find_device_by_name(&sdw_bus_type, NULL, sdw_dev_name); + if (!sdw_dev) + return -EPROBE_DEFER; + + if (SOF_RT711_JDSRC(sof_sdw_quirk)) { + props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", + SOF_RT711_JDSRC(sof_sdw_quirk)); + } + + ret = device_add_properties(sdw_dev, props); + put_device(sdw_dev); + + return ret; +} + +static const struct snd_soc_dapm_widget rt711_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route rt711_map[] = { + /* Headphones */ + { "Headphone", NULL, "rt711 HP" }, + { "rt711 MIC2", NULL, "Headset Mic" }, +}; + +static const struct snd_kcontrol_new rt711_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_jack_pin rt711_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s hs:rt711", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt711_controls, + ARRAY_SIZE(rt711_controls)); + if (ret) { + dev_err(card->dev, "rt711 controls addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt711_widgets, + ARRAY_SIZE(rt711_widgets)); + if (ret) { + dev_err(card->dev, "rt711 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt711_map, + ARRAY_SIZE(rt711_map)); + + if (ret) { + dev_err(card->dev, "rt711 map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sdw_headset, + rt711_jack_pins, + ARRAY_SIZE(rt711_jack_pins)); + if (ret) { + dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n", + ret); + return ret; + } + + jack = &ctx->sdw_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) + dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n", + ret); + + return ret; +} + +int sof_sdw_rt711_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + int ret; + + /* + * headset should be initialized once. + * Do it with dai link for playback. + */ + if (!playback) + return 0; + + ret = rt711_add_codec_device_props("sdw:0:25d:711:0"); + if (ret < 0) + return ret; + + dai_links->init = rt711_rtd_init; + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw_rt715.c b/sound/soc/intel/boards/sof_sdw_rt715.c new file mode 100644 index 00000000000000..321e1cbc03ed72 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt715.c @@ -0,0 +1,42 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt715 - Helpers to handle RT715 from generic machine driver + */ + +#include +#include +#include +#include +#include "sof_sdw_common.h" + +static int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s mic:rt715", + card->components); + if (!card->components) + return -ENOMEM; + + return 0; +} + +int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* + * DAI ID is fixed at SDW_DMIC_DAI_ID for 715 to + * keep sdw DMIC and HDMI setting static in UCM + */ + if (sof_sdw_quirk & SOF_RT715_DAI_ID_FIX) + dai_links->id = SDW_DMIC_DAI_ID; + + dai_links->init = rt715_rtd_init; + + return 0; +} From 095ee71907ea02702f5fd84184d2656572e2c81c Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Wed, 25 Mar 2020 17:07:45 -0500 Subject: [PATCH 405/415] ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver RT5682 is in SoundWire mode on link0. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Naveen Manohar Link: https://lore.kernel.org/r/20200325220746.29601-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-tgl-match.c | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index db360c9a8e5bfe..449d9d2286ae24 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -56,6 +56,14 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt5682_0_adr[] = { + { + .adr = 0x000021025D568200, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + static const struct snd_soc_acpi_link_adr tgl_i2s_rt1308[] = { { .mask = BIT(0), @@ -79,6 +87,15 @@ static const struct snd_soc_acpi_link_adr tgl_rvp[] = { {} }; +static const struct snd_soc_acpi_link_adr tgl_chromebook_base[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt5682_0_adr), + .adr_d = rt5682_0_adr, + }, + {} +}; + static struct snd_soc_acpi_codecs tgl_max98373_amp = { .num_codecs = 1, .codecs = {"MX98373"} @@ -122,6 +139,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg", }, + { + .link_mask = 0x1, /* this will only enable rt5682 for now */ + .links = tgl_chromebook_base, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_sdw_machines); From 798313f29b6b510a7df386cf7e8e4636afe61e81 Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Wed, 25 Mar 2020 17:07:46 -0500 Subject: [PATCH 406/415] ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Add support for Google Volteer device. As per new unified soundwire machine driver, add rt5682-sdw helper function, which configures codec to Link0. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Naveen Manohar Link: https://lore.kernel.org/r/20200325220746.29601-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/sof_sdw.c | 18 ++++ sound/soc/intel/boards/sof_sdw_common.h | 6 ++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 ++++++++++++++++++++++++ 5 files changed, 152 insertions(+) create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 4110ae5db65f22..556c3104e641d7 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -538,6 +538,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_RT1308_SDW select SND_SOC_RT1308 select SND_SOC_RT715_SDW + select SND_SOC_RT5682_SDW select SND_SOC_DMIC help Add support for Intel SoundWire-based platforms connected to diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index c4ff5166a04269..1ef6e60bc2a034 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -34,6 +34,7 @@ snd-soc-sof_da7219_max98373-objs := sof_da7219_max98373.o hda_dsp_common.o snd-soc-sof-sdw-objs += sof_sdw.o \ sof_sdw_rt711.o sof_sdw_rt700.o \ sof_sdw_rt1308.o sof_sdw_rt715.o \ + sof_sdw_rt5682.o \ sof_sdw_dmic.o sof_sdw_hdmi.o hda_dsp_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 8ed6d2079deecf..a64dc563b47e1b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -91,6 +91,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)SOF_SDW_PCH_DMIC, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Volteer"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC), + }, {} }; @@ -128,6 +136,10 @@ static struct snd_soc_codec_conf codec_conf[] = { .dlc = COMP_CODEC_CONF("sdw:3:25d:715:0"), .name_prefix = "rt715", }, + { + .dlc = COMP_CODEC_CONF("sdw:0:25d:5682:0"), + .name_prefix = "rt5682", + }, }; static struct snd_soc_dai_link_component dmic_component[] = { @@ -187,6 +199,12 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .init = sof_sdw_rt715_init, }, + { + .id = 0x5682, + .direction = {true, true}, + .dai_name = "rt5682-sdw", + .init = sof_sdw_rt5682_init, + }, }; static inline int find_codec_info_part(unsigned int part_id) diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 0d738e234bc5b1..dd593ff3575b54 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -105,4 +105,10 @@ int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link, struct sof_sdw_codec_info *info, bool playback); +/* RT5682 support */ +int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + #endif diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c new file mode 100644 index 00000000000000..5aa6211a1ed9ec --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -0,0 +1,126 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt5682 - Helpers to handle RT5682 from generic machine driver + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget rt5682_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route rt5682_map[] = { + /*Headphones*/ + { "Headphone", NULL, "rt5682 HPOL" }, + { "Headphone", NULL, "rt5682 HPOR" }, + { "rt5682 IN1P", NULL, "Headset Mic" }, +}; + +static const struct snd_kcontrol_new rt5682_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_jack_pin rt5682_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s hs:rt5682", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt5682_controls, + ARRAY_SIZE(rt5682_controls)); + if (ret) { + dev_err(card->dev, "rt5682 control addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt5682_widgets, + ARRAY_SIZE(rt5682_widgets)); + if (ret) { + dev_err(card->dev, "rt5682 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5682_map, + ARRAY_SIZE(rt5682_map)); + + if (ret) { + dev_err(card->dev, "rt5682 map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sdw_headset, + rt5682_jack_pins, + ARRAY_SIZE(rt5682_jack_pins)); + if (ret) { + dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n", + ret); + return ret; + } + + jack = &ctx->sdw_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) + dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n", + ret); + + return ret; +} + +int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* + * headset should be initialized once. + * Do it with dai link for playback. + */ + if (!playback) + return 0; + + dai_links->init = rt5682_rtd_init; + + return 0; +} From d4061518c3982010d8f07b9b327c8e00297b14a3 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 27 Mar 2020 11:24:32 -0500 Subject: [PATCH 407/415] ASoC: tlv320adcx140: Remove undocumented property Remove undocumented and unneeded ti,use-internal-reg from the example as it was an artifact from initial development. The code does not query for this property and as the document indicates if areg-supply is undefined then the internal regulator is used. Fixes: 302c0b7490cd ("dt-bindings: sound: Add TLV320ADCx140 dt bindings") Signed-off-by: Dan Murphy CC: Rob Herring Link: https://lore.kernel.org/r/20200327162432.17067-1-dmurphy@ti.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 1433ff62b14f02..ab2268c0ee67f8 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -76,7 +76,6 @@ examples: codec: codec@4c { compatible = "ti,tlv320adc5140"; reg = <0x4c>; - ti,use-internal-areg; ti,mic-bias-source = <6>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; }; From acd4946f5bf031fa38e64bfe2467be94a1b8c25d Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Fri, 27 Mar 2020 14:14:29 +0000 Subject: [PATCH 408/415] ASoC: amd: acp3x-pcm-dma: clean up two indentation issues There are a couple of statements that are not indented correctly, add in the missing tab and break the lines to address a checkpatch warning. Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20200327141429.269191-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index d62c0d90c41e34..e362f0bc9e4633 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -458,7 +458,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_ITER; frmt_val = mmACP_I2STDM_TXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { struct i2s_stream_instance *rtd = @@ -474,7 +475,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_IRER; frmt_val = mmACP_I2STDM_RXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->tdm_mode == TDM_ENABLE) { rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val); From 652bb5d8df4b3a79ed350db35cda12637e63efa7 Mon Sep 17 00:00:00 2001 From: Rouven Czerwinski Date: Sun, 29 Mar 2020 07:30:15 +0200 Subject: [PATCH 409/415] ALSA: hda: default enable CA0132 DSP support If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as well. Disabled DSP support leads to a hanging alsa system and no sound output on the card otherwise. Tested on: 06:00.0 Audio device: Creative Labs Sound Core3D [Sound Blaster Recon3D / Z-Series] (rev 01) Signed-off-by: Rouven Czerwinski Link: https://lore.kernel.org/r/20200329053710.4276-1-r.czerwinski@pengutronix.de Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bd48335d09d74f..e1d3082a4fe931 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 + default y select SND_HDA_DSP_LOADER select FW_LOADER help From 476c02e0b4fd9071d158f6a1a1dfea1d36ee0ffd Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Sun, 29 Mar 2020 16:20:18 +0800 Subject: [PATCH 410/415] ALSA: hda/realtek - a fake key event is triggered by running shutup On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 76f7dec08fd6 (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1) Cc: Kailang Yang Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20200329082018.20486-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1aff2..0af33f00617a47 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { From f128090491c3f5aacef91a863f8c52abf869c436 Mon Sep 17 00:00:00 2001 From: Thomas Hebb Date: Mon, 30 Mar 2020 12:09:37 -0400 Subject: [PATCH 411/415] ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 This codec (among others) has a hidden set of audio routes, apparently designed to allow PC Beep output without a mixer widget on the output path, which are controlled by an undocumented Realtek vendor register. The default configuration of these routes means that certain inputs aren't accessible, necessitating driver control of the register. However, Realtek has provided no documentation of the register, instead opting to fix issues by providing magic numbers, most of which have been at least somewhat erroneous. These magic numbers then get copied by others into model-specific fixups, leading to a fragmented and buggy set of configurations. To get out of this situation, I've reverse engineered the register by flipping bits and observing how the codec's behavior changes. This commit documents my findings. It does not change any code. Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb Link: https://lore.kernel.org/r/bd69dfdeaf40ff31c4b7b797c829bb320031739c.1585584498.git.tommyhebb@gmail.com Signed-off-by: Takashi Iwai --- Documentation/sound/hd-audio/index.rst | 1 + .../sound/hd-audio/realtek-pc-beep.rst | 129 ++++++++++++++++++ 2 files changed, 130 insertions(+) create mode 100644 Documentation/sound/hd-audio/realtek-pc-beep.rst diff --git a/Documentation/sound/hd-audio/index.rst b/Documentation/sound/hd-audio/index.rst index f8a72ffffe66f4..6e12de9fc34e29 100644 --- a/Documentation/sound/hd-audio/index.rst +++ b/Documentation/sound/hd-audio/index.rst @@ -8,3 +8,4 @@ HD-Audio models controls dp-mst + realtek-pc-beep diff --git a/Documentation/sound/hd-audio/realtek-pc-beep.rst b/Documentation/sound/hd-audio/realtek-pc-beep.rst new file mode 100644 index 00000000000000..be47c6f76a6e9f --- /dev/null +++ b/Documentation/sound/hd-audio/realtek-pc-beep.rst @@ -0,0 +1,129 @@ +=============================== +Realtek PC Beep Hidden Register +=============================== + +This file documents the "PC Beep Hidden Register", which is present in certain +Realtek HDA codecs and controls a muxer and pair of passthrough mixers that can +route audio between pins but aren't themselves exposed as HDA widgets. As far +as I can tell, these hidden routes are designed to allow flexible PC Beep output +for codecs that don't have mixer widgets in their output paths. Why it's easier +to hide a mixer behind an undocumented vendor register than to just expose it +as a widget, I have no idea. + +Register Description +==================== + +The register is accessed via processing coefficient 0x36 on NID 20h. Bits not +identified below have no discernible effect on my machine, a Dell XPS 13 9350:: + + MSB LSB + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + | |h|S|L| | B |R| | Known bits + +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ + |0|0|1|1| 0x7 |0|0x0|1| 0x7 | Reset value + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +1Ah input select (B): 2 bits + When zero, expose the PC Beep line (from the internal beep generator, when + enabled with the Set Beep Generation verb on NID 01h, or else from the + external PCBEEP pin) on the 1Ah pin node. When nonzero, expose the headphone + jack (or possibly Line In on some machines) input instead. If PC Beep is + selected, the 1Ah boost control has no effect. + +Amplify 1Ah loopback, left (L): 1 bit + Amplify the left channel of 1Ah before mixing it into outputs as specified + by h and S bits. Does not affect the level of 1Ah exposed to other widgets. + +Amplify 1Ah loopback, right (R): 1 bit + Amplify the right channel of 1Ah before mixing it into outputs as specified + by h and S bits. Does not affect the level of 1Ah exposed to other widgets. + +Loopback 1Ah to 21h [active low] (h): 1 bit + When zero, mix 1Ah (possibly with amplification, depending on L and R bits) + into 21h (headphone jack on my machine). Mixed signal respects the mute + setting on 21h. + +Loopback 1Ah to 14h (S): 1 bit + When one, mix 1Ah (possibly with amplification, depending on L and R bits) + into 14h (internal speaker on my machine). Mixed signal **ignores** the mute + setting on 14h and is present whenever 14h is configured as an output. + +Path diagrams +============= + +1Ah input selection (DIV is the PC Beep divider set on NID 01h):: + + + | | | + +--DIV--+--!DIV--+ {1Ah boost control} + | | + +--(b == 0)--+--(b != 0)--+ + | + >1Ah (Beep/Headphone Mic/Line In)< + +Loopback of 1Ah to 21h/14h:: + + <1Ah (Beep/Headphone Mic/Line In)> + | + {amplify if L/R} + | + +-----!h-----+-----S-----+ + | | + {21h mute control} | + | | + >21h (Headphone)< >14h (Internal Speaker)< + +Background +========== + +All Realtek HDA codecs have a vendor-defined widget with node ID 20h which +provides access to a bank of registers that control various codec functions. +Registers are read and written via the standard HDA processing coefficient +verbs (Set/Get Coefficient Index, Set/Get Processing Coefficient). The node is +named "Realtek Vendor Registers" in public datasheets' verb listings and, +apart from that, is entirely undocumented. + +This particular register, exposed at coefficient 0x36 and named in commits from +Realtek, is of note: unlike most registers, which seem to control detailed +amplifier parameters not in scope of the HDA specification, it controls audio +routing which could just as easily have been defined using standard HDA mixer +and selector widgets. + +Specifically, it selects between two sources for the input pin widget with Node +ID (NID) 1Ah: the widget's signal can come either from an audio jack (on my +laptop, a Dell XPS 13 9350, it's the headphone jack, but comments in Realtek +commits indicate that it might be a Line In on some machines) or from the PC +Beep line (which is itself multiplexed between the codec's internal beep +generator and external PCBEEP pin, depending on if the beep generator is +enabled via verbs on NID 01h). Additionally, it can mix (with optional +amplification) that signal onto the 21h and/or 14h output pins. + +The register's reset value is 0x3717, corresponding to PC Beep on 1Ah that is +then amplified and mixed into both the headphones and the speakers. Not only +does this violate the HDA specification, which says that "[a vendor defined +beep input pin] connection may be maintained *only* while the Link reset +(**RST#**) is asserted", it means that we cannot ignore the register if we care +about the input that 1Ah would otherwise expose or if the PCBEEP trace is +poorly shielded and picks up chassis noise (both of which are the case on my +machine). + +Unfortunately, there are lots of ways to get this register configuration wrong. +Linux, it seems, has gone through most of them. For one, the register resets +after S3 suspend: judging by existing code, this isn't the case for all vendor +registers, and it's led to some fixes that improve behavior on cold boot but +don't last after suspend. Other fixes have successfully switched the 1Ah input +away from PC Beep but have failed to disable both loopback paths. On my +machine, this means that the headphone input is amplified and looped back to +the headphone output, which uses the exact same pins! As you might expect, this +causes terrible headphone noise, the character of which is controlled by the +1Ah boost control. (If you've seen instructions online to fix XPS 13 headphone +noise by changing "Headphone Mic Boost" in ALSA, now you know why.) + +The information here has been obtained through black-box reverse engineering of +the ALC256 codec's behavior and is not guaranteed to be correct. It likely +also applies for the ALC255, ALC257, ALC235, and ALC236, since those codecs +seem to be close relatives of the ALC256. (They all share one initialization +function.) Additionally, other codecs like the ALC225 and ALC285 also have this +register, judging by existing fixups in ``patch_realtek.c``, but specific +data (e.g. node IDs, bit positions, pin mappings) for those codecs may differ +from what I've described here. From c44737449468a0bdc50e09ec75e530f208391561 Mon Sep 17 00:00:00 2001 From: Thomas Hebb Date: Mon, 30 Mar 2020 12:09:38 -0400 Subject: [PATCH 412/415] ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 The Realtek PC Beep Hidden Register[1] is currently set by patch_realtek.c in two different places: In alc_fill_eapd_coef(), it's set to the value 0x5757, corresponding to non-beep input on 1Ah and no 1Ah loopback to either headphones or speakers. (Although, curiously, the loopback amp is still enabled.) This write was added fairly recently by commit e3743f431143 ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236") and is a safe default. However, it happens in the wrong place: alc_fill_eapd_coef() runs on module load and cold boot but not on S3 resume, meaning the register loses its value after suspend. Conversely, in alc256_init(), the register is updated to unset bit 13 (disable speaker loopback) and set bit 5 (set non-beep input on 1Ah). Although this write does run on S3 resume, it's not quite enough to fix up the register's default value of 0x3717. What's missing is a set of bit 14 to disable headphone loopback. Without that, we end up with a feedback loop where the headphone jack is being driven by amplified samples of itself[2]. This change eliminates the update in alc256_init() and replaces it with the 0x5757 write from alc_fill_eapd_coef(). Kailang says that 0x5757 is supposed to be the codec's default value, so using it will make debugging easier for Realtek. Affects the ALC255, ALC256, ALC257, ALC235, and ALC236 codecs. [1] Newly documented in Documentation/sound/hd-audio/realtek-pc-beep.rst [2] Setting the "Headphone Mic Boost" control from userspace changes this feedback loop and has been a widely-shared workaround for headphone noise on laptops like the Dell XPS 13 9350. This commit eliminates the feedback loop and makes the workaround unnecessary. Fixes: e1e8c1fdce8b ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236") Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb Link: https://lore.kernel.org/r/bf22b417d1f2474b12011c2a39ed6cf8b06d3bf5.1585584498.git.tommyhebb@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0af33f00617a47..425521d03dc329 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -368,7 +368,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0233: case 0x10ec0235: + case 0x10ec0236: case 0x10ec0255: + case 0x10ec0256: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -380,11 +382,6 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0300: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x36, 0x5757); - alc_update_coef_idx(codec, 0x10, 1<<9, 0); - break; case 0x10ec0275: alc_update_coef_idx(codec, 0xe, 0, 1<<0); break; @@ -3371,7 +3368,13 @@ static void alc256_init(struct hda_codec *codec) alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */ alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 1 << 15); /* Clear bit */ alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 0 << 15); - alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/ + /* + * Expose headphone mic (or possibly Line In on some machines) instead + * of PC Beep on 1Ah, and disable 1Ah loopback for all outputs. See + * Documentation/sound/hd-audio/realtek-pc-beep.rst for details of + * this register. + */ + alc_write_coef_idx(codec, 0x36, 0x5757); } static void alc256_shutup(struct hda_codec *codec) From f36938aa7440f46a0a365f1cfde5f5985af2bef3 Mon Sep 17 00:00:00 2001 From: Thomas Hebb Date: Mon, 30 Mar 2020 12:09:39 -0400 Subject: [PATCH 413/415] ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups patch_realtek.c has historically failed to properly configure the PC Beep Hidden Register for the ALC256 codec (among others). Depending on your kernel version, symptoms of this misconfiguration can range from chassis noise, picked up by a poorly-shielded PCBEEP trace, getting amplified and played on your internal speaker and/or headphones to loud feedback, which responds to the "Headphone Mic Boost" ALSA control, getting played through your headphones. For details of the problem, see the patch in this series titled "ALSA: hda/realtek - Set principled PC Beep configuration for ALC256", which fixes the configuration. These symptoms have been most noticed on the Dell XPS 13 9350 and 9360, popular laptops that use the ALC256. As a result, several model-specific fixups have been introduced to try and fix the problem, the most egregious of which locks the "Headphone Mic Boost" control as a hack to minimize noise from a feedback loop that shouldn't have been there in the first place. Now that the underlying issue has been fixed, remove all these fixups. Remaining fixups needed by the XPS 13 are all picked up by existing pin quirks. This change should, for the XPS 13 9350/9360 - Significantly increase volume and audio quality on headphones - Eliminate headphone popping on suspend/resume - Allow "Headphone Mic Boost" to be set again, making the headphone jack fully usable as a microphone jack too. Fixes: 8c69729b4439 ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3") Fixes: 423cd785619a ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360") Fixes: e4c9fd10eb21 ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant") Fixes: 1099f48457d0 ("ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360") Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb Link: https://lore.kernel.org/r/b649a00edfde150cf6eebbb4390e15e0c2deb39a.1585584498.git.tommyhebb@gmail.com Signed-off-by: Takashi Iwai --- Documentation/sound/hd-audio/models.rst | 2 -- sound/pci/hda/patch_realtek.c | 34 ------------------------- 2 files changed, 36 deletions(-) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 11298f0ce44dbf..0ea967d3458387 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -216,8 +216,6 @@ alc298-dell-aio ALC298 fixups on Dell AIO machines alc275-dell-xps ALC275 fixups on Dell XPS models -alc256-dell-xps13 - ALC256 fixups on Dell XPS13 lenovo-spk-noise Workaround for speaker noise on Lenovo machines lenovo-hotkey diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 425521d03dc329..f66a48154a5751 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5491,17 +5491,6 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } -static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec, - const struct hda_fixup *fix, - int action) -{ - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; - - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1); - snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP); -} - static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5916,8 +5905,6 @@ enum { ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, - ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, - ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, @@ -6658,23 +6645,6 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, - [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* Disable pass-through path for FRONT 14h */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x36}, - {0x20, AC_VERB_SET_PROC_COEF, 0x1737}, - {} - }, - .chained = true, - .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE - }, - [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = { - .type = HDA_FIXUP_FUNC, - .v.func = alc256_fixup_dell_xps_13_headphone_noise2, - .chained = true, - .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE - }, [ALC293_FIXUP_LENOVO_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -7172,17 +7142,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), - SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), @@ -7536,7 +7503,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc298-dell1"}, {.id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, .name = "alc298-dell-aio"}, {.id = ALC275_FIXUP_DELL_XPS, .name = "alc275-dell-xps"}, - {.id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, .name = "alc256-dell-xps13"}, {.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"}, {.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"}, {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, From b6f69c795547f59ddf1db17cddbd2b9a15c656ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 31 Mar 2020 11:00:23 +0200 Subject: [PATCH 414/415] Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h" This reverts commit 645c08f17f477915f6d900b767e789852f150054 which was reported to break the build a program using this header. The original issue was addressed in the alsa-lib side recently, so we can make the header more self-contained again. Reported-by: Dmitry V. Levin Fixes: 645c08f17f47 ("ALSA: uapi: Drop asound.h inclusion from asoc.h") Cc: Link: https://lore.kernel.org/r/20200331090023.8112-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/asoc.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6048553c119d94..a74ca232f1fc1f 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -17,6 +17,7 @@ #define __LINUX_UAPI_SND_ASOC_H #include +#include /* * Maximum number of channels topology kcontrol can represent. From 5c6cd7021a05a02fcf37f360592d7c18d4d807fb Mon Sep 17 00:00:00 2001 From: Andreas Steinmetz Date: Tue, 31 Mar 2020 14:25:54 +0200 Subject: [PATCH 415/415] ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@domdv.de Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index be5c597ed40c84..047b90595d6581 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1826,6 +1826,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1863,11 +1885,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) {