From 135d1535f4619ce74e46b9268c4a7899bc531cb1 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 15 Aug 2011 00:22:50 +0200 Subject: [PATCH 01/45] ALSA: hdspm - Allow for 8192 period size on RME MADI and AES cards Older RME cards like MADI and AES support period sizes of 8192 samples. The original hdspm driver already featured this value, apparently, it was lost during the rewrite. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 493e3946756f9e..204e1ced16a778 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5673,7 +5673,7 @@ static int snd_hdspm_prepare(struct snd_pcm_substream *substream) } static unsigned int period_sizes_old[] = { - 64, 128, 256, 512, 1024, 2048, 4096 + 64, 128, 256, 512, 1024, 2048, 4096, 8192 }; static unsigned int period_sizes_new[] = { From 1b6fa108b33f4a3e3999563e830daff39d332f70 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 15 Aug 2011 00:22:51 +0200 Subject: [PATCH 02/45] ALSA: hdspm - Set period_bytes_min to 32 * 4 for new RME cards On newer RME cards like RayDAT and AIO, the lower bound is 32 samples per period in contrast to 64 samples as seen on older cards. We hence lower period_bytes_min to 32 * 4. Four bytes per sample. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 204e1ced16a778..8dc2a894f6f7e6 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5703,7 +5703,7 @@ static struct snd_pcm_hardware snd_hdspm_playback_subinfo = { .channels_max = HDSPM_MAX_CHANNELS, .buffer_bytes_max = HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS, - .period_bytes_min = (64 * 4), + .period_bytes_min = (32 * 4), .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS, .periods_min = 2, .periods_max = 512, @@ -5728,7 +5728,7 @@ static struct snd_pcm_hardware snd_hdspm_capture_subinfo = { .channels_max = HDSPM_MAX_CHANNELS, .buffer_bytes_max = HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS, - .period_bytes_min = (64 * 4), + .period_bytes_min = (32 * 4), .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS, .periods_min = 2, .periods_max = 512, From 1ad5972f71f94d8a8b5b683dd5f81a52a4ddf54c Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 15 Aug 2011 00:22:52 +0200 Subject: [PATCH 03/45] ALSA: hdspm - Reorder period sizes according to their bit representation On newer RME cards like RayDAT and AIO, the 8192 samples per period size are no longer supported. Instead, setting all three bits of HDSP_LatencyMask to one ({1,1,1}) now corresponds to 32 samples per period. To make this more obvious to future developers, let's reorder the array according to their bit representation, starting at 64 ({0,0,0}) up to 4096 ({1,1,0}) and finally 32 ({1,1,1}). Note that this patch doesn't change semantics. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8dc2a894f6f7e6..159133a14464b0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5677,7 +5677,7 @@ static unsigned int period_sizes_old[] = { }; static unsigned int period_sizes_new[] = { - 32, 64, 128, 256, 512, 1024, 2048, 4096 + 64, 128, 256, 512, 1024, 2048, 4096, 32 }; /* RayDAT and AIO always have a buffer of 16384 samples per channel */ From 7cb155ff3e4645188c42d707300e36cfce44e28a Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 15 Aug 2011 00:22:53 +0200 Subject: [PATCH 04/45] ALSA: hdspm - Introduce hdspm_get_latency() to harmonize latency calculation Currently, hdspm_decode_latency is called several times, violating the DRY principle. Given that we need to distinguish between old and new cards when decoding the latency bits in the control register, introduce hdspm_get_latency() to provide the required functionality. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 159133a14464b0..1a52a1ae1f4cc9 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1241,10 +1241,30 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) return rate; } +/* return latency in samples per period */ +static int hdspm_get_latency(struct hdspm *hdspm) +{ + int n; + + n = hdspm_decode_latency(hdspm->control_register); + + /* Special case for new RME cards with 32 samples period size. + * The three latency bits in the control register + * (HDSP_LatencyMask) encode latency values of 64 samples as + * 0, 128 samples as 1 ... 4096 samples as 6. For old cards, 7 + * denotes 8192 samples, but on new cards like RayDAT or AIO, + * it corresponds to 32 samples. + */ + if ((7 == n) && (RayDAT == hdspm->io_type || AIO == hdspm->io_type)) + n = -1; + + return 1 << (n + 6); +} + /* Latency function */ static inline void hdspm_compute_period_size(struct hdspm *hdspm) { - hdspm->period_bytes = 1 << ((hdspm_decode_latency(hdspm->control_register) + 8)); + hdspm->period_bytes = 4 * hdspm_get_latency(hdspm); } @@ -4801,8 +4821,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Settings ---\n"); - x = 1 << (6 + hdspm_decode_latency(hdspm->control_register & - HDSPM_LatencyMask)); + x = hdspm_get_latency(hdspm); snd_iprintf(buffer, "Size (Latency): %d samples (2 periods of %lu bytes)\n", @@ -4965,8 +4984,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Settings ---\n"); - x = 1 << (6 + hdspm_decode_latency(hdspm->control_register & - HDSPM_LatencyMask)); + x = hdspm_get_latency(hdspm); snd_iprintf(buffer, "Size (Latency): %d samples (2 periods of %lu bytes)\n", From 2e61027079ed70f54fec41ddb8fa8af37d79d8d8 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 15 Aug 2011 00:22:54 +0200 Subject: [PATCH 05/45] ALSA: hdspm - Enable 32 samples/period on RME RayDAT/AIO Newer RME cards like RayDAT and AIO support 32 samples per period. This value is encoded as {1,1,1} in the HDSP_LatencyMask bits in the control register. Since {1,1,1} is also the representation for 8192 samples/period on older RME cards, we have to special case 32 samples and 32768 bytes according to the actual card. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 25 ++++++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 1a52a1ae1f4cc9..92ac64ced29ae8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1323,12 +1323,27 @@ static int hdspm_set_interrupt_interval(struct hdspm *s, unsigned int frames) spin_lock_irq(&s->lock); - frames >>= 7; - n = 0; - while (frames) { - n++; - frames >>= 1; + if (32 == frames) { + /* Special case for new RME cards like RayDAT/AIO which + * support period sizes of 32 samples. Since latency is + * encoded in the three bits of HDSP_LatencyMask, we can only + * have values from 0 .. 7. While 0 still means 64 samples and + * 6 represents 4096 samples on all cards, 7 represents 8192 + * on older cards and 32 samples on new cards. + * + * In other words, period size in samples is calculated by + * 2^(n+6) with n ranging from 0 .. 7. + */ + n = 7; + } else { + frames >>= 7; + n = 0; + while (frames) { + n++; + frames >>= 1; + } } + s->control_register &= ~HDSPM_LatencyMask; s->control_register |= hdspm_encode_latency(n); From f39d5a88badb22139cca99b06fc4fe729450ba5c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 19 Jun 2011 12:10:43 +0200 Subject: [PATCH 06/45] ALSA: isight: remove superfluous field Remove a field that is not used at all. This remained from earlier tests, but the current driver has decided not to handle iris notifications. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 440030818db70c..cd094ecaca3bef 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -51,7 +51,6 @@ struct isight { struct fw_unit *unit; struct fw_device *device; u64 audio_base; - struct fw_address_handler iris_handler; struct snd_pcm_substream *pcm; struct mutex mutex; struct iso_packets_buffer buffer; From dc3fcd1655bf1ba01843c557d6646500b0759173 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sat, 18 Jun 2011 23:05:00 +0200 Subject: [PATCH 07/45] ALSA: virtuoso: fix Essence ST(X) S/PDIF input On the Xonar Essence ST/STX, the connector J14 has been confirmed to be a digital input, so enable it in the driver. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_pcm179x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 32d096c98f5bc2..8433aa7c3d7540 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -1074,6 +1074,7 @@ static const struct oxygen_model model_xonar_st = { .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF | AC97_FMIC_SWITCH, .dac_channels_pcm = 2, .dac_channels_mixer = 2, From 52e6fb48121a552d11ea0eb05540178fb3ac4e15 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Aug 2011 10:40:59 +0200 Subject: [PATCH 08/45] ALSA: hdspm - Correct max buffer size limit Some modesl can support up to 8192 frames per period. Tested-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 92ac64ced29ae8..c33f4a5c52419f 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5737,7 +5737,7 @@ static struct snd_pcm_hardware snd_hdspm_playback_subinfo = { .buffer_bytes_max = HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS, .period_bytes_min = (32 * 4), - .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS, + .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS, .periods_min = 2, .periods_max = 512, .fifo_size = 0 @@ -5762,7 +5762,7 @@ static struct snd_pcm_hardware snd_hdspm_capture_subinfo = { .buffer_bytes_max = HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS, .period_bytes_min = (32 * 4), - .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS, + .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS, .periods_min = 2, .periods_max = 512, .fifo_size = 0 From 3fa9e3d230911272eaf1c3856f5483b0af3903f3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Aug 2011 10:42:23 +0200 Subject: [PATCH 09/45] ALSA: hdspm - Add missing KNOT flag for AES32 rate restriction AES32 supports the non-standard 128kHZ, and this is enabled only when SNDRV_PCM_RATE_KNOT is set in hw.rates field. Tested-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index c33f4a5c52419f..4add485e6b1632 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6006,6 +6006,7 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) } if (AES32 == hdspm->io_type) { + runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdspm_hw_constraints_aes32_sample_rates); } else { @@ -6076,6 +6077,7 @@ static int snd_hdspm_capture_open(struct snd_pcm_substream *substream) } if (AES32 == hdspm->io_type) { + runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdspm_hw_constraints_aes32_sample_rates); } else { From d877681d2eab28ae2a7ff08bec9a6fe3b65973fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Aug 2011 10:45:42 +0200 Subject: [PATCH 10/45] ALSA: hdspm - Simplify with snd_pcm_hw_constraint_pow2() Refactoring the code using snd_pcm_hw_constraint_pow2() helper function. Tested-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 76 +++++++++++++-------------------------- 1 file changed, 25 insertions(+), 51 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 4add485e6b1632..214110d6a2bf3b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5705,19 +5705,6 @@ static int snd_hdspm_prepare(struct snd_pcm_substream *substream) return 0; } -static unsigned int period_sizes_old[] = { - 64, 128, 256, 512, 1024, 2048, 4096, 8192 -}; - -static unsigned int period_sizes_new[] = { - 64, 128, 256, 512, 1024, 2048, 4096, 32 -}; - -/* RayDAT and AIO always have a buffer of 16384 samples per channel */ -static unsigned int raydat_aio_buffer_sizes[] = { - 16384 -}; - static struct snd_pcm_hardware snd_hdspm_playback_subinfo = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -5768,24 +5755,6 @@ static struct snd_pcm_hardware snd_hdspm_capture_subinfo = { .fifo_size = 0 }; -static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes_old = { - .count = ARRAY_SIZE(period_sizes_old), - .list = period_sizes_old, - .mask = 0 -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes_new = { - .count = ARRAY_SIZE(period_sizes_new), - .list = period_sizes_new, - .mask = 0 -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_raydat_io_buffer = { - .count = ARRAY_SIZE(raydat_aio_buffer_sizes), - .list = raydat_aio_buffer_sizes, - .mask = 0 -}; - static int snd_hdspm_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -5986,23 +5955,25 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) spin_unlock_irq(&hdspm->lock); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); switch (hdspm->io_type) { case AIO: case RayDAT: - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - &hw_constraints_period_sizes_new); - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - &hw_constraints_raydat_io_buffer); - + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 32, 4096); + /* RayDAT & AIO have a fixed buffer of 16384 samples per channel */ + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 16384, 16384); break; default: - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - &hw_constraints_period_sizes_old); + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 64, 8192); + break; } if (AES32 == hdspm->io_type) { @@ -6059,21 +6030,24 @@ static int snd_hdspm_capture_open(struct snd_pcm_substream *substream) spin_unlock_irq(&hdspm->lock); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + switch (hdspm->io_type) { case AIO: case RayDAT: - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - &hw_constraints_period_sizes_new); - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - &hw_constraints_raydat_io_buffer); - break; + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 32, 4096); + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 16384, 16384); + break; default: - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - &hw_constraints_period_sizes_old); + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 64, 8192); + break; } if (AES32 == hdspm->io_type) { From 9fcd0ab130579d9742538340edda3225f2b49a3e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Aug 2011 08:30:53 +0200 Subject: [PATCH 11/45] ALSA: usb-audio - Check the dB-range validity in the later read, too When the initial check of dB-range failed due to the read error, try to check again at the later read, too. When an invalid dB range is found, remove TLV flags and notify the mixer info change. Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index cdd19d7fe500b2..78a5abda6793dc 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -881,8 +881,17 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ uinfo->value.integer.min = 0; uinfo->value.integer.max = 1; } else { - if (! cval->initialized) - get_min_max(cval, 0); + if (!cval->initialized) { + get_min_max(cval, 0); + if (cval->initialized && cval->dBmin >= cval->dBmax) { + kcontrol->vd[0].access &= + ~(SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); + snd_ctl_notify(cval->mixer->chip->card, + SNDRV_CTL_EVENT_MASK_INFO, + &kcontrol->id); + } + } uinfo->value.integer.min = 0; uinfo->value.integer.max = (cval->max - cval->min + cval->res - 1) / cval->res; From 983929cafc4225d61f50b3e35cf892606a15bc69 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 24 Aug 2011 11:12:34 +0800 Subject: [PATCH 12/45] ALSA: core: trivial code style fix remove trailing tab on the line. Signed-off-by: Lu Guanqun Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index f8c5be46451058..7f2b3a7eabb2bb 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1072,7 +1072,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, long private_size; struct user_element *ue; int idx, err; - + if (card->user_ctl_count >= MAX_USER_CONTROLS) return -ENOMEM; if (info->count < 1) From 08ede038a738f22c1b3425051175e1d627d8dd43 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 24 Aug 2011 14:45:10 +0800 Subject: [PATCH 13/45] ALSA: core: release the constraint check for replace ops Suppose the ALSA card already has a number of MAX_USER_CONTROLS controls, and the user wants to replace one, it should not fail at this condition check. Signed-off-by: Lu Guanqun Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 7f2b3a7eabb2bb..dc2a44048c850e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1073,7 +1073,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, struct user_element *ue; int idx, err; - if (card->user_ctl_count >= MAX_USER_CONTROLS) + if (!replace && card->user_ctl_count >= MAX_USER_CONTROLS) return -ENOMEM; if (info->count < 1) return -EINVAL; From 391e69143d0a05f960e3ab39a8c26b7b230bb8a9 Mon Sep 17 00:00:00 2001 From: Maarten Lankhorst Date: Wed, 24 Aug 2011 00:48:59 +0200 Subject: [PATCH 14/45] ALSA: ctxfi: Bump playback substreams to 256 There are references in the code to 256 sources, so I tested it with 256 aplays, of which the first and last with real data and the rest playing /dev/zero . Also increase amount of page tables, so the default aplay size works. Signed-off-by: Maarten Lankhorst Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctpcm.c | 2 +- sound/pci/ctxfi/ctsrc.c | 2 +- sound/pci/ctxfi/ctvmem.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 457d21189b0db1..2c8622617c8c4c 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -404,7 +404,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, int err; int playback_count, capture_count; - playback_count = (IEC958 == device) ? 1 : 8; + playback_count = (IEC958 == device) ? 1 : 256; capture_count = (FRONT == device) ? 1 : 0; err = snd_pcm_new(atc->card, "ctxfi", device, playback_count, capture_count, &pcm); diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index c749fa72088996..e134b3a5780da7 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -20,7 +20,7 @@ #include "cthardware.h" #include -#define SRC_RESOURCE_NUM 64 +#define SRC_RESOURCE_NUM 256 #define SRCIMP_RESOURCE_NUM 256 static unsigned int conj_mask; diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index b23adfca4de6eb..e6da60eb19ceb7 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -18,7 +18,7 @@ #ifndef CTVMEM_H #define CTVMEM_H -#define CT_PTP_NUM 1 /* num of device page table pages */ +#define CT_PTP_NUM 4 /* num of device page table pages */ #include #include From 1ef0e0a05345b7411bdabbfca27f58bd33dcc7c8 Mon Sep 17 00:00:00 2001 From: Kristian Amlie Date: Fri, 26 Aug 2011 13:19:49 +0200 Subject: [PATCH 15/45] ALSA: usb-audio: add Starr Labs USB MIDI support Add support for Starr Labs USB MIDI devices such as the Z7S, which are based on an FTDI serial UART chip. Based on a patch by Daniel Mack. Signed-off-by: Kristian Amlie Acked-by: Daniel Mack Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 27 +++++++++++++++++++++++++++ sound/usb/quirks-table.h | 11 +++++++++++ sound/usb/quirks.c | 1 + sound/usb/usbaudio.h | 1 + 4 files changed, 40 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index f9289102886ad8..e21f026d957750 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -816,6 +816,22 @@ static struct usb_protocol_ops snd_usbmidi_raw_ops = { .output = snd_usbmidi_raw_output, }; +/* + * FTDI protocol: raw MIDI bytes, but input packets have two modem status bytes. + */ + +static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint* ep, + uint8_t* buffer, int buffer_length) +{ + if (buffer_length > 2) + snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2); +} + +static struct usb_protocol_ops snd_usbmidi_ftdi_ops = { + .input = snd_usbmidi_ftdi_input, + .output = snd_usbmidi_raw_output, +}; + static void snd_usbmidi_us122l_input(struct snd_usb_midi_in_endpoint *ep, uint8_t *buffer, int buffer_length) { @@ -2163,6 +2179,17 @@ int snd_usbmidi_create(struct snd_card *card, /* endpoint 1 is input-only */ endpoints[1].out_cables = 0; break; + case QUIRK_MIDI_FTDI: + umidi->usb_protocol_ops = &snd_usbmidi_ftdi_ops; + + /* set baud rate to 31250 (48 MHz / 16 / 96) */ + err = usb_control_msg(umidi->dev, usb_sndctrlpipe(umidi->dev, 0), + 3, 0x40, 0x60, 0, NULL, 0, 1000); + if (err < 0) + break; + + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index a42e3ef3832d70..da898229bb1192 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -39,6 +39,17 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +/* FTDI devices */ +{ + USB_DEVICE(0x0403, 0xb8d8), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "STARR LABS", */ + /* .product_name = "Starr Labs MIDI USB device", */ + .ifnum = 0, + .type = QUIRK_MIDI_FTDI + } +}, + /* Creative/Toshiba Multimedia Center SB-0500 */ { USB_DEVICE(0x041e, 0x3048), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 81e07d84258177..cf61b03400263d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -306,6 +306,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_MIDI_AKAI] = create_any_midi_quirk, + [QUIRK_MIDI_FTDI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1e79986b577749..3e2b035779362d 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -80,6 +80,7 @@ enum quirk_type { QUIRK_MIDI_CME, QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, + QUIRK_MIDI_FTDI, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UAXX, From 294c4fb8ab01728358836f478bcc1174ba7fb9d8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 6 Sep 2011 19:15:34 -0500 Subject: [PATCH 16/45] ALSA: usb: refine delay information with USB frame counter Existing code only updates the audio delay when URBs were submitted/retired. This can introduce an uncertainty of 8ms on the number of samples played out with the default settings, and a lot more when URBs convey more packets to reduce the interrupt rate and power consumption. This patch relies on the USB frame counter to reduce the uncertainty to less than 2ms worst-case. The delay information essentially becomes independent of the URB size and number of packets. This should help applications like PulseAudio which require accurate audio timing. Clemens Ladisch reported a decrease of mplayer's A-V difference from nrpacks down to at most 1ms. Thanks to Clemens for also pointing out that the implementation of frame counters varies between different HCDs. Only the 8 lowest-bits are used to estimate the delay. Signed-off-by: Pierre-Louis Bossart [clemens: changed debug code] Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/card.h | 2 ++ sound/usb/pcm.c | 28 ++++++++++++++++++++++++++++ sound/usb/pcm.h | 3 +++ sound/usb/urb.c | 30 +++++++++++++++++++++++++++--- 4 files changed, 60 insertions(+), 3 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index ae4251d5abf7bc..a39edcc32a93f0 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -94,6 +94,8 @@ struct snd_usb_substream { spinlock_t lock; struct snd_urb_ops ops; /* callbacks (must be filled at init) */ + int last_frame_number; /* stored frame number */ + int last_delay; /* stored delay */ }; struct snd_usb_stream { diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b8dcbf407bbbbe..0b699ca1957ef8 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -34,6 +34,30 @@ #include "clock.h" #include "power.h" +/* return the estimated delay based on USB frame counters */ +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate) +{ + int current_frame_number; + int frame_diff; + int est_delay; + + current_frame_number = usb_get_current_frame_number(subs->dev); + /* + * HCD implementations use different widths, use lower 8 bits. + * The delay will be managed up to 256ms, which is more than + * enough + */ + frame_diff = (current_frame_number - subs->last_frame_number) & 0xff; + + /* Approximation based on number of samples per USB frame (ms), + some truncation for 44.1 but the estimate is good enough */ + est_delay = subs->last_delay - (frame_diff * rate / 1000); + if (est_delay < 0) + est_delay = 0; + return est_delay; +} + /* * return the current pcm pointer. just based on the hwptr_done value. */ @@ -45,6 +69,8 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; + substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); } @@ -417,6 +443,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->hwptr_done = 0; subs->transfer_done = 0; subs->phase = 0; + subs->last_delay = 0; + subs->last_frame_number = 0; runtime->delay = 0; return snd_usb_substream_prepare(subs, runtime); diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index ed3e283f618d11..df7a003682ad64 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -1,6 +1,9 @@ #ifndef __USBAUDIO_PCM_H #define __USBAUDIO_PCM_H +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate); + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, diff --git a/sound/usb/urb.c b/sound/usb/urb.c index e184349aee83f1..b4dcccc237dc9a 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -718,7 +718,16 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + + /* update delay with exact number of samples queued */ + runtime->delay = subs->last_delay; runtime->delay += frames; + subs->last_delay = runtime->delay; + + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; if (period_elapsed) @@ -737,12 +746,27 @@ static int retire_playback_urb(struct snd_usb_substream *subs, unsigned long flags; int stride = runtime->frame_bits >> 3; int processed = urb->transfer_buffer_length / stride; + int est_delay; spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; + + est_delay = snd_usb_pcm_delay(subs, runtime->rate); + /* update delay with exact number of samples played */ + if (processed > subs->last_delay) + subs->last_delay = 0; else - runtime->delay -= processed; + subs->last_delay -= processed; + runtime->delay = subs->last_delay; + + /* + * Report when delay estimate is off by more than 2ms. + * The error should be lower than 2ms since the estimate relies + * on two reads of a counter updated every ms. + */ + if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) + snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", + est_delay, subs->last_delay); + spin_unlock_irqrestore(&subs->lock, flags); return 0; } From 89f3325a6e3002f33bc5e0412d35fc097e219dbd Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Fri, 9 Sep 2011 19:15:01 +0800 Subject: [PATCH 17/45] ALSA: ymfpci: add "Playback" to FM Legacy Volume control YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth. Signed-off-by: Raymond Yau Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/ymfpci/ymfpci_main.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index f3260e658b8add..ebfbb28c35cc60 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1615,7 +1615,7 @@ YMFPCI_DOUBLE("ADC Playback Volume", 0, YDSXGR_PRIADCOUTVOL), YMFPCI_DOUBLE("ADC Capture Volume", 0, YDSXGR_PRIADCLOOPVOL), YMFPCI_DOUBLE("ADC Playback Volume", 1, YDSXGR_SECADCOUTVOL), YMFPCI_DOUBLE("ADC Capture Volume", 1, YDSXGR_SECADCLOOPVOL), -YMFPCI_DOUBLE("FM Legacy Volume", 0, YDSXGR_LEGACYOUTVOL), +YMFPCI_DOUBLE("FM Legacy Playback Volume", 0, YDSXGR_LEGACYOUTVOL), YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ", PLAYBACK,VOLUME), 0, YDSXGR_ZVOUTVOL), YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("", CAPTURE,VOLUME), 0, YDSXGR_ZVLOOPVOL), YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ",PLAYBACK,VOLUME), 1, YDSXGR_SPDIFOUTVOL), From 5758960353d179f0541226764a1e47eff01666ff Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 9 Sep 2011 19:04:45 +0800 Subject: [PATCH 18/45] ALSA: aoa: Remove obsolete cleanup for clientdata The i2c core will clear the clientdata pointer automatically. We don't have to set the `data' field to NULL in remove() or if probe() failed anymore. Also remove a unneeded NULL checking for kfree. Signed-off-by: Axel Lin Reviewed-by: Wolfram Sang Signed-off-by: Takashi Iwai --- sound/aoa/codecs/onyx.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 3687a6cc9881ed..762af68c8996fb 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1067,7 +1067,6 @@ static int onyx_i2c_probe(struct i2c_client *client, printk(KERN_DEBUG PFX "created and attached onyx instance\n"); return 0; fail: - i2c_set_clientdata(client, NULL); kfree(onyx); return -ENODEV; } @@ -1112,8 +1111,7 @@ static int onyx_i2c_remove(struct i2c_client *client) aoa_codec_unregister(&onyx->codec); of_node_put(onyx->codec.node); - if (onyx->codec_info) - kfree(onyx->codec_info); + kfree(onyx->codec_info); kfree(onyx); return 0; } From 47124373b59e43fd07cbf7b44d9288f19c1d5a93 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 9 Sep 2011 17:50:52 +0800 Subject: [PATCH 19/45] ALSA: keywest: Remove obsolete cleanup for clientdata The i2c core will clear the clientdata pointer automatically. We don't have to set the `data' field to NULL in remove() or if probe() failed anymore. Signed-off-by: Axel Lin Reviewed-by: Wolfram Sang Signed-off-by: Takashi Iwai --- sound/ppc/keywest.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 8f064c7ce74569..4080becf4cef0f 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -82,7 +82,6 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) static int keywest_remove(struct i2c_client *client) { - i2c_set_clientdata(client, NULL); if (! keywest_ctx) return 0; if (client == keywest_ctx->client) From dba8b46992c55946d3b092934f581a343403118f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 Sep 2011 11:24:41 +0200 Subject: [PATCH 20/45] ALSA: mpu401: clean up interrupt specification The semantics of snd_mpu401_uart_new()'s interrupt parameters are somewhat counterintuitive: To prevent the function from allocating its own interrupt, either the irq number must be invalid, or the irq_flags parameter must be zero. At the same time, the irq parameter being invalid specifies that the mpu401 code has to work without an interrupt allocated by the caller. This implies that, if there is an interrupt and it is allocated by the caller, the irq parameter must be set to a valid-looking number which then isn't actually used. With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value, which forces us to handle the parameters differently. This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the device interrupt is handled by the caller, and makes the allocation of the interrupt to depend only on the irq parameter. As suggested by Takashi, the irq_flags parameter was dropped because, when used, it had the constant value IRQF_DISABLED. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- .../DocBook/writing-an-alsa-driver.tmpl | 36 +++++++++++-------- include/sound/mpu401.h | 7 ++-- sound/drivers/mpu401/mpu401.c | 3 +- sound/drivers/mpu401/mpu401_uart.c | 20 +++++------ sound/isa/ad1816a/ad1816a.c | 2 +- sound/isa/als100.c | 1 - sound/isa/azt2320.c | 3 +- sound/isa/cmi8330.c | 2 +- sound/isa/cs423x/cs4231.c | 1 - sound/isa/cs423x/cs4236.c | 3 +- sound/isa/es1688/es1688.c | 2 +- sound/isa/es18xx.c | 4 +-- sound/isa/galaxy/galaxy.c | 3 +- sound/isa/gus/gusextreme.c | 3 +- sound/isa/msnd/msnd_pinnacle.c | 2 +- sound/isa/opl3sa2.c | 5 +-- sound/isa/opti9xx/miro.c | 3 +- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- sound/isa/sb/jazz16.c | 1 - sound/isa/sb/sb16.c | 5 +-- sound/isa/sc6000.c | 3 +- sound/isa/sscape.c | 3 +- sound/isa/wavefront/wavefront.c | 3 +- sound/pci/als4000.c | 5 +-- sound/pci/au88x0/au88x0_mpu401.c | 6 ++-- sound/pci/azt3328.c | 5 +-- sound/pci/cmipci.c | 5 +-- sound/pci/es1938.c | 5 +-- sound/pci/es1968.c | 5 +-- sound/pci/fm801.c | 5 +-- sound/pci/ice1712/ice1712.c | 10 +++--- sound/pci/maestro3.c | 4 +-- sound/pci/oxygen/oxygen_lib.c | 6 ++-- sound/pci/riptide/riptide.c | 2 +- sound/pci/sonicvibes.c | 7 ++-- sound/pci/trident/trident.c | 5 +-- sound/pci/via82xx.c | 5 +-- sound/pci/ymfpci/ymfpci.c | 5 +-- 38 files changed, 103 insertions(+), 94 deletions(-) diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index 598c22f3b3ac8f..5de23c00707828 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -4288,7 +4288,7 @@ struct _snd_pcm_runtime { @@ -4343,6 +4343,13 @@ struct _snd_pcm_runtime { by itself to start processing the output stream in the irq handler. + + If the MPU-401 interface shares its interrupt with the other logical + devices on the card, set MPU401_INFO_IRQ_HOOK + (see + below). + + Usually, the port address corresponds to the command port and port + 1 corresponds to the data port. If not, you may change @@ -4375,14 +4382,12 @@ struct _snd_pcm_runtime { - The 6th argument specifies the irq number for UART. If the irq - is already allocated, pass 0 to the 7th argument - (irq_flags). Otherwise, pass the flags - for irq allocation - (SA_XXX bits) to it, and the irq will be - reserved by the mpu401-uart layer. If the card doesn't generate - UART interrupts, pass -1 as the irq number. Then a timer - interrupt will be invoked for polling. + The 6th argument specifies the ISA irq number that will be + allocated. If no interrupt is to be allocated (because your + code is already allocating a shared interrupt, or because the + device does not use interrupts), pass -1 instead. + For a MPU-401 device without an interrupt, a polling timer + will be used instead. @@ -4390,12 +4395,13 @@ struct _snd_pcm_runtime { Interrupt Handler When the interrupt is allocated in - snd_mpu401_uart_new(), the private - interrupt handler is used, hence you don't have anything else to do - than creating the mpu401 stuff. Otherwise, you have to call - snd_mpu401_uart_interrupt() explicitly when - a UART interrupt is invoked and checked in your own interrupt - handler. + snd_mpu401_uart_new(), an exclusive ISA + interrupt handler is automatically used, hence you don't have + anything else to do than creating the mpu401 stuff. Otherwise, you + have to set MPU401_INFO_IRQ_HOOK, and call + snd_mpu401_uart_interrupt() explicitly from your + own interrupt handler when it has determined that a UART interrupt + has occurred. diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h index 1f1d53f8830b55..20230db00ef170 100644 --- a/include/sound/mpu401.h +++ b/include/sound/mpu401.h @@ -50,7 +50,10 @@ #define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */ #define MPU401_INFO_MMIO (1 << 3) /* MMIO access */ #define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */ +#define MPU401_INFO_IRQ_HOOK (1 << 5) /* mpu401 irq handler is called + from driver irq handler */ #define MPU401_INFO_NO_ACK (1 << 6) /* No ACK cmd needed */ +#define MPU401_INFO_USE_TIMER (1 << 15) /* internal */ #define MPU401_MODE_BIT_INPUT 0 #define MPU401_MODE_BIT_OUTPUT 1 @@ -73,8 +76,7 @@ struct snd_mpu401 { unsigned long port; /* base port of MPU-401 chip */ unsigned long cport; /* port + 1 (usually) */ struct resource *res; /* port resource */ - int irq; /* IRQ number of MPU-401 chip (-1 = poll) */ - int irq_flags; + int irq; /* IRQ number of MPU-401 chip */ unsigned long mode; /* MPU401_MODE_XXXX */ int timer_invoked; @@ -131,7 +133,6 @@ int snd_mpu401_uart_new(struct snd_card *card, unsigned long port, unsigned int info_flags, int irq, - int irq_flags, struct snd_rawmidi ** rrawmidi); #endif /* __SOUND_MPU401_H */ diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 149d05a8202d58..1c02852aceea77 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -86,8 +86,7 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard) } err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0, - irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0, - NULL); + irq[dev], NULL); if (err < 0) { printk(KERN_ERR "MPU401 not detected at 0x%lx\n", port[dev]); goto _err; diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 2af09996a3d01a..9d01c181fecaef 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -3,7 +3,7 @@ * Routines for control of MPU-401 in UART mode * * MPU-401 supports UART mode which is not capable generate transmit - * interrupts thus output is done via polling. Also, if irq < 0, then + * interrupts thus output is done via polling. Without interrupt, * input is done also via polling. Do not expect good performance. * * @@ -374,7 +374,7 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up) /* first time - flush FIFO */ while (max-- > 0) mpu->read(mpu, MPU401D(mpu)); - if (mpu->irq < 0) + if (mpu->info_flags & MPU401_INFO_USE_TIMER) snd_mpu401_uart_add_timer(mpu, 1); } @@ -383,7 +383,7 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up) snd_mpu401_uart_input_read(mpu); spin_unlock_irqrestore(&mpu->input_lock, flags); } else { - if (mpu->irq < 0) + if (mpu->info_flags & MPU401_INFO_USE_TIMER) snd_mpu401_uart_remove_timer(mpu, 1); clear_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode); } @@ -496,7 +496,7 @@ static struct snd_rawmidi_ops snd_mpu401_uart_input = static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) { struct snd_mpu401 *mpu = rmidi->private_data; - if (mpu->irq_flags && mpu->irq >= 0) + if (mpu->irq >= 0) free_irq(mpu->irq, (void *) mpu); release_and_free_resource(mpu->res); kfree(mpu); @@ -509,8 +509,7 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) * @hardware: the hardware type, MPU401_HW_XXXX * @port: the base address of MPU401 port * @info_flags: bitflags MPU401_INFO_XXX - * @irq: the irq number, -1 if no interrupt for mpu - * @irq_flags: the irq request flags (SA_XXX), 0 if irq was already reserved. + * @irq: the ISA irq number, -1 if not to be allocated * @rrawmidi: the pointer to store the new rawmidi instance * * Creates a new MPU-401 instance. @@ -525,7 +524,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, unsigned short hardware, unsigned long port, unsigned int info_flags, - int irq, int irq_flags, + int irq, struct snd_rawmidi ** rrawmidi) { struct snd_mpu401 *mpu; @@ -577,8 +576,8 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, mpu->cport = port + 2; else mpu->cport = port + 1; - if (irq >= 0 && irq_flags) { - if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags, + if (irq >= 0) { + if (request_irq(irq, snd_mpu401_uart_interrupt, IRQF_DISABLED, "MPU401 UART", (void *) mpu)) { snd_printk(KERN_ERR "mpu401_uart: " "unable to grab IRQ %d\n", irq); @@ -586,9 +585,10 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, return -EBUSY; } } + if (irq < 0 && !(info_flags & MPU401_INFO_IRQ_HOOK)) + info_flags |= MPU401_INFO_USE_TIMER; mpu->info_flags = info_flags; mpu->irq = irq; - mpu->irq_flags = irq_flags; if (card->shortname[0]) snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI", card->shortname); diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 3cb75bc9769927..a87a2b566e19e4 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -204,7 +204,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard if (mpu_port[dev] > 0) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED, + mpu_port[dev], 0, mpu_irq[dev], NULL) < 0) printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpu_port[dev]); } diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 20becc89f6f6f4..706effd6b3cd8d 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -256,7 +256,6 @@ static int __devinit snd_card_als100_probe(int dev, mpu_type, mpu_port[dev], 0, mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index aac8dc15c2fe6e..b7bdbf30774025 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -234,8 +234,7 @@ static int __devinit snd_card_azt2320_probe(int dev, if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_AZT2320, mpu_port[dev], 0, - mpu_irq[dev], IRQF_DISABLED, - NULL) < 0) + mpu_irq[dev], NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index fe79a169acb52e..dca69f80305fb8 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -597,7 +597,7 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) if (mpuport[dev] != SNDRV_AUTO_PORT) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpuport[dev], 0, mpuirq[dev], - IRQF_DISABLED, NULL) < 0) + NULL) < 0) printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpuport[dev]); } diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index cb9153e75b8228..409fa0ad7843b1 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -131,7 +131,6 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n) mpu_irq[n] = -1; if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232, mpu_port[n], 0, mpu_irq[n], - mpu_irq[n] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) dev_warn(dev, "MPU401 not detected\n"); } diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 999dc1e0fdbd5f..0dbde461e6c1b0 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -449,8 +449,7 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) mpu_irq[dev] = -1; if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232, mpu_port[dev], 0, - mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) + mpu_irq[dev], NULL) < 0) printk(KERN_WARNING IDENT ": MPU401 not detected\n"); } diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 0cde8131a57544..5493e9e4bcd5dc 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -174,7 +174,7 @@ static int __devinit snd_es1688_probe(struct snd_card *card, unsigned int n) chip->mpu_port > 0) { error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688, chip->mpu_port, 0, - mpu_irq[n], IRQF_DISABLED, NULL); + mpu_irq[n], NULL); if (error < 0) return error; } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index fb4d6b34bbca5c..aeee8f8bf5e99b 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2160,8 +2160,8 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, - mpu_port[dev], 0, - irq[dev], 0, &chip->rmidi); + mpu_port[dev], MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); if (err < 0) return err; } diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index ee54df082b9c07..e51d3244742af2 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -585,8 +585,7 @@ static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n) if (mpu_port[n] >= 0) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port[n], 0, mpu_irq[n], - IRQF_DISABLED, NULL); + mpu_port[n], 0, mpu_irq[n], NULL); if (err < 0) goto error; } diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 008e8e5bfa37ae..c4733c08b60b2d 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -317,8 +317,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) if (es1688->mpu_port >= 0x300) { error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688, - es1688->mpu_port, 0, - mpu_irq[n], IRQF_DISABLED, NULL); + es1688->mpu_port, 0, mpu_irq[n], NULL); if (error < 0) goto out; } diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 91d6023a63e57c..0961e2cf20caa8 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -600,7 +600,7 @@ static int __devinit snd_msnd_attach(struct snd_card *card) mpu_io[0], MPU401_MODE_INPUT | MPU401_MODE_OUTPUT, - mpu_irq[0], IRQF_DISABLED, + mpu_irq[0], &chip->rmidi); if (err < 0) { printk(KERN_ERR LOGNAME diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 9b915e27b5bd7d..de99f47770bfb8 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -707,8 +707,9 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev) } if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x340) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_OPL3SA2, - midi_port[dev], 0, - xirq, 0, &chip->rmidi)) < 0) + midi_port[dev], + MPU401_INFO_IRQ_HOOK, -1, + &chip->rmidi)) < 0) return err; } sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d", diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 8c24102d0d9389..d94d0f35cb765c 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1377,8 +1377,7 @@ static int __devinit snd_miro_probe(struct snd_card *card) rmidi = NULL; else { error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, - &rmidi); + mpu_port, 0, miro->mpu_irq, &rmidi); if (error < 0) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", mpu_port); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index c35dc68930dc52..346e12baa98e98 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -914,7 +914,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) rmidi = NULL; else { error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi); + mpu_port, 0, mpu_irq, &rmidi); if (error) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", mpu_port); diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 8ccbcddf08e19d..54e3c2c1806075 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -322,7 +322,6 @@ static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev) MPU401_HW_MPU401, mpu_port[dev], 0, mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", mpu_port[dev]); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 4d1c5a300ff84d..237f8bd7fbe422 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -394,8 +394,9 @@ static int __devinit snd_sb16_probe(struct snd_card *card, int dev) if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SB, - chip->mpu_port, 0, - xirq, 0, &chip->rmidi)) < 0) + chip->mpu_port, + MPU401_INFO_IRQ_HOOK, -1, + &chip->rmidi)) < 0) return err; chip->rmidi_callback = snd_mpu401_uart_interrupt; } diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 9a8bbf6dd62aea..207c161f100c5b 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -658,8 +658,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port[dev], 0, - mpu_irq[dev], IRQF_DISABLED, - NULL) < 0) + mpu_irq[dev], NULL) < 0) snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n", mpu_port[dev]); } diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index e2d5d2d3ed9647..f2379e102b63e6 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -825,8 +825,7 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, int err; err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, - MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, - &rawmidi); + MPU401_INFO_INTEGRATED, irq, &rawmidi); if (err == 0) { struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 711670e4a4251e..83f291d89a95ce 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -449,8 +449,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) if (cs4232_mpu_port[dev] > 0 && cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) { err = snd_mpu401_uart_new(card, midi_dev, MPU401_HW_CS4232, cs4232_mpu_port[dev], 0, - cs4232_mpu_irq[dev], IRQF_DISABLED, - NULL); + cs4232_mpu_irq[dev], NULL); if (err < 0) { snd_printk (KERN_ERR "can't allocate CS4232 MPU-401 device\n"); return err; diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index a9c1af33f27662..04628696eb082a 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -931,8 +931,9 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000, iobase + ALS4K_IOB_30_MIDI_DATA, - MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi)) < 0) { printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n", iobase + ALS4K_IOB_30_MIDI_DATA); goto out_err; diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 0dc8d259d1ed09..e6c6a0febb752a 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -84,7 +84,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) #ifdef VORTEX_MPU401_LEGACY if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_MPU401, 0x330, - 0, 0, 0, &rmidi)) != 0) { + MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) { hwwrite(vortex->mmio, VORTEX_CTRL, (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN); @@ -94,8 +94,8 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA); if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port, - MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO, - 0, 0, &rmidi)) != 0) { + MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO | + MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) { hwwrite(vortex->mmio, VORTEX_CTRL, (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 579fc0dce12850..d24fe425e87f45 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2652,8 +2652,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) since our hardware ought to be similar, thus use same ID. */ err = snd_mpu401_uart_new( card, 0, - MPU401_HW_AZT2320, chip->mpu_io, MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi + MPU401_HW_AZT2320, chip->mpu_io, + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi ); if (err < 0) { snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 9cf99fb7eb9c5b..da9c73211eca7c 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3228,8 +3228,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, iomidi, (integrated_midi ? - MPU401_INFO_INTEGRATED : 0), - cm->irq, 0, &cm->rmidi)) < 0) { + MPU401_INFO_INTEGRATED : 0) | + MPU401_INFO_IRQ_HOOK, + -1, &cm->rmidi)) < 0) { printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi); } } diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 26a5a2f25d4bc8..718a2643474e5b 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1854,8 +1854,9 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci, } } if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi) < 0) { + chip->mpu_port, + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi) < 0) { printk(KERN_ERR "es1938: unable to initialize MPU-401\n"); } else { // this line is vital for MIDI interrupt handling on ess-solo1 diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 99ea9320c6b559..407e4abc43568e 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2843,8 +2843,9 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, if (enable_mpu[dev]) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, chip->io_port + ESM_MPU401_PORT, - MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi)) < 0) { printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n"); } } diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index f9123f09e83e79..c55b1b319b747e 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1306,8 +1306,9 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801, FM801_REG(chip, MPU401_DATA), - MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0ccc0eb75775dd..8531b983f3aff9 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2748,8 +2748,9 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, if (!c->no_mpu401) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, ICEREG(ice, MPU1_CTRL), - (c->mpu401_1_info_flags | MPU401_INFO_INTEGRATED), - ice->irq, 0, &ice->rmidi[0]); + c->mpu401_1_info_flags | + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &ice->rmidi[0]); if (err < 0) { snd_card_free(card); return err; @@ -2764,8 +2765,9 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, /* 2nd port used */ err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, ICEREG(ice, MPU2_CTRL), - (c->mpu401_2_info_flags | MPU401_INFO_INTEGRATED), - ice->irq, 0, &ice->rmidi[1]); + c->mpu401_2_info_flags | + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &ice->rmidi[1]); if (err < 0) { snd_card_free(card); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 0378126e6272b2..2fd4bf2d6653db 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2820,8 +2820,8 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) /* TODO enable MIDI IRQ and I/O */ err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401, chip->iobase + MPU401_DATA_PORT, - MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi); + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); if (err < 0) printk(KERN_WARNING "maestro3: no MIDI support.\n"); #endif diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 82311fcb86f6d3..53e5508abcbf58 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -678,15 +678,15 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, goto err_card; if (chip->model.device_config & (MIDI_OUTPUT | MIDI_INPUT)) { - unsigned int info_flags = MPU401_INFO_INTEGRATED; + unsigned int info_flags = + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK; if (chip->model.device_config & MIDI_OUTPUT) info_flags |= MPU401_INFO_OUTPUT; if (chip->model.device_config & MIDI_INPUT) info_flags |= MPU401_INFO_INPUT; err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, chip->addr + OXYGEN_MPU401, - info_flags, 0, 0, - &chip->midi); + info_flags, -1, &chip->midi); if (err < 0) goto err_card; } diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e34ae14908b312..88cc776aa38b1b 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -2109,7 +2109,7 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) val = mpu_port[dev]; pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, val); err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE, - val, 0, chip->irq, 0, + val, MPU401_INFO_IRQ_HOOK, -1, &chip->rmidi); if (err < 0) snd_printk(KERN_WARNING diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 2571a67b389a14..c5008166cf1f0d 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1493,9 +1493,10 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, return err; } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES, - sonic->midi_port, MPU401_INFO_INTEGRATED, - sonic->irq, 0, - &midi_uart)) < 0) { + sonic->midi_port, + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &midi_uart)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index d8a128f6fc0269..5e707effdc7cac 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -148,8 +148,9 @@ static int __devinit snd_trident_probe(struct pci_dev *pci, if (trident->device != TRIDENT_DEVICE_ID_SI7018 && (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE, trident->midi_port, - MPU401_INFO_INTEGRATED, - trident->irq, 0, &trident->rmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &trident->rmidi)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index f03fd620a2a076..35d5f4313d99ba 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2068,8 +2068,9 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip) pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, legacy_cfg); if (chip->mpu_res) { if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A, - mpu_port, MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi) < 0) { + mpu_port, MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, -1, + &chip->rmidi) < 0) { printk(KERN_WARNING "unable to initialize MPU-401" " at 0x%lx, skipping\n", mpu_port); legacy &= ~VIA_FUNC_ENABLE_MIDI; diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 511d5765312495..3253b04da18449 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -305,8 +305,9 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, if (chip->mpu_res) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI, mpu_port[dev], - MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rawmidi)) < 0) { + MPU401_INFO_INTEGRATED | + MPU401_INFO_IRQ_HOOK, + -1, &chip->rawmidi)) < 0) { printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]); legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */ pci_write_config_word(pci, PCIR_DSXG_LEGACY, legacy_ctrl); From 00137425fe5892e6e531ffee6bf5f108d823b70f Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 12 Sep 2011 18:54:10 +0200 Subject: [PATCH 21/45] USB: Add endpoint usage definitions to ch9.h The endpoint usage field is described in the USB 2.0 specification, chapter 9.6.6. Also, move the sync type fields block down by some lines to reflect the fact that these are also stuffed in bmAttributes. Signed-off-by: Daniel Mack Acked-by: Clemens Ladisch Acked-by: Greg Kroah-Hartman Signed-off-by: Takashi Iwai --- include/linux/usb/ch9.h | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/include/linux/usb/ch9.h b/include/linux/usb/ch9.h index 0fd3fbdd8283fe..f302535995015d 100644 --- a/include/linux/usb/ch9.h +++ b/include/linux/usb/ch9.h @@ -377,12 +377,6 @@ struct usb_endpoint_descriptor { #define USB_ENDPOINT_NUMBER_MASK 0x0f /* in bEndpointAddress */ #define USB_ENDPOINT_DIR_MASK 0x80 -#define USB_ENDPOINT_SYNCTYPE 0x0c -#define USB_ENDPOINT_SYNC_NONE (0 << 2) -#define USB_ENDPOINT_SYNC_ASYNC (1 << 2) -#define USB_ENDPOINT_SYNC_ADAPTIVE (2 << 2) -#define USB_ENDPOINT_SYNC_SYNC (3 << 2) - #define USB_ENDPOINT_XFERTYPE_MASK 0x03 /* in bmAttributes */ #define USB_ENDPOINT_XFER_CONTROL 0 #define USB_ENDPOINT_XFER_ISOC 1 @@ -390,6 +384,17 @@ struct usb_endpoint_descriptor { #define USB_ENDPOINT_XFER_INT 3 #define USB_ENDPOINT_MAX_ADJUSTABLE 0x80 +#define USB_ENDPOINT_SYNCTYPE 0x0c +#define USB_ENDPOINT_SYNC_NONE (0 << 2) +#define USB_ENDPOINT_SYNC_ASYNC (1 << 2) +#define USB_ENDPOINT_SYNC_ADAPTIVE (2 << 2) +#define USB_ENDPOINT_SYNC_SYNC (3 << 2) + +#define USB_ENDPOINT_USAGE_MASK 0x30 +#define USB_ENDPOINT_USAGE_DATA 0x00 +#define USB_ENDPOINT_USAGE_FEEDBACK 0x10 +#define USB_ENDPOINT_USAGE_IMPLICIT_FB 0x20 /* Implicit feedback Data endpoint */ + /*-------------------------------------------------------------------------*/ /** From 358e2bd4a97780f5522e1666c8188a3a60a0d03c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 12 Sep 2011 18:54:11 +0200 Subject: [PATCH 22/45] ALSA: snd-usb: re-order the Makefile Sort its entries in alphabetical order. Signed-off-by: Daniel Mack Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/Makefile | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/usb/Makefile b/sound/usb/Makefile index cf9ed66445fad6..083501e78f3418 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -3,16 +3,16 @@ # snd-usb-audio-objs := card.o \ + clock.o \ + endpoint.o \ + format.o \ + helper.o \ mixer.o \ mixer_quirks.o \ + pcm.o \ proc.o \ quirks.o \ - format.o \ - endpoint.o \ - urb.o \ - pcm.o \ - helper.o \ - clock.o + urb.o snd-usbmidi-lib-objs := midi.o From e8e8babf561c9560f37b9bd80407ebaf90ad2ca4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 12 Sep 2011 18:54:12 +0200 Subject: [PATCH 23/45] ALSA: snd-usb: re-order code Move code from endpoint.c into a new file called stream.c and rename functions so that their names actually reflect what they're doing. This way, endpoint.c will be available to functions that hold all the endpoint logic. Signed-off-by: Daniel Mack Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/Makefile | 1 + sound/usb/card.c | 3 +- sound/usb/endpoint.c | 433 ----------------------------------------- sound/usb/endpoint.h | 7 - sound/usb/quirks.c | 7 +- sound/usb/stream.c | 453 +++++++++++++++++++++++++++++++++++++++++++ sound/usb/stream.h | 12 ++ 7 files changed, 472 insertions(+), 444 deletions(-) create mode 100644 sound/usb/stream.c create mode 100644 sound/usb/stream.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 083501e78f3418..5390db00e09859 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -12,6 +12,7 @@ snd-usb-audio-objs := card.o \ pcm.o \ proc.o \ quirks.o \ + stream.o \ urb.o snd-usbmidi-lib-objs := midi.o diff --git a/sound/usb/card.c b/sound/usb/card.c index 781d9e61adfbe6..a3136afb2198b6 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -68,6 +68,7 @@ #include "urb.h" #include "format.h" #include "power.h" +#include "stream.h" MODULE_AUTHOR("Takashi Iwai "); MODULE_DESCRIPTION("USB Audio"); @@ -185,7 +186,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int return -EINVAL; } - if (! snd_usb_parse_audio_endpoints(chip, interface)) { + if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); return -EINVAL; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7d46e482375d1d..b3ee7cf243dfde 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -15,436 +15,3 @@ * */ -#include -#include -#include -#include -#include - -#include -#include - -#include "usbaudio.h" -#include "card.h" -#include "proc.h" -#include "quirks.h" -#include "endpoint.h" -#include "urb.h" -#include "pcm.h" -#include "helper.h" -#include "format.h" -#include "clock.h" - -/* - * free a substream - */ -static void free_substream(struct snd_usb_substream *subs) -{ - struct list_head *p, *n; - - if (!subs->num_formats) - return; /* not initialized */ - list_for_each_safe(p, n, &subs->fmt_list) { - struct audioformat *fp = list_entry(p, struct audioformat, list); - kfree(fp->rate_table); - kfree(fp); - } - kfree(subs->rate_list.list); -} - - -/* - * free a usb stream instance - */ -static void snd_usb_audio_stream_free(struct snd_usb_stream *stream) -{ - free_substream(&stream->substream[0]); - free_substream(&stream->substream[1]); - list_del(&stream->list); - kfree(stream); -} - -static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) -{ - struct snd_usb_stream *stream = pcm->private_data; - if (stream) { - stream->pcm = NULL; - snd_usb_audio_stream_free(stream); - } -} - - -/* - * add this endpoint to the chip instance. - * if a stream with the same endpoint already exists, append to it. - * if not, create a new pcm stream. - */ -int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp) -{ - struct list_head *p; - struct snd_usb_stream *as; - struct snd_usb_substream *subs; - struct snd_pcm *pcm; - int err; - - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); - if (as->fmt_type != fp->fmt_type) - continue; - subs = &as->substream[stream]; - if (!subs->endpoint) - continue; - if (subs->endpoint == fp->endpoint) { - list_add_tail(&fp->list, &subs->fmt_list); - subs->num_formats++; - subs->formats |= fp->formats; - return 0; - } - } - /* look for an empty stream */ - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); - if (as->fmt_type != fp->fmt_type) - continue; - subs = &as->substream[stream]; - if (subs->endpoint) - continue; - err = snd_pcm_new_stream(as->pcm, stream, 1); - if (err < 0) - return err; - snd_usb_init_substream(as, stream, fp); - return 0; - } - - /* create a new pcm */ - as = kzalloc(sizeof(*as), GFP_KERNEL); - if (!as) - return -ENOMEM; - as->pcm_index = chip->pcm_devs; - as->chip = chip; - as->fmt_type = fp->fmt_type; - err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1, - &pcm); - if (err < 0) { - kfree(as); - return err; - } - as->pcm = pcm; - pcm->private_data = as; - pcm->private_free = snd_usb_audio_pcm_free; - pcm->info_flags = 0; - if (chip->pcm_devs > 0) - sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs); - else - strcpy(pcm->name, "USB Audio"); - - snd_usb_init_substream(as, stream, fp); - - list_add(&as->list, &chip->pcm_list); - chip->pcm_devs++; - - snd_usb_proc_pcm_format_add(as); - - return 0; -} - -static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, - struct usb_host_interface *alts, - int protocol, int iface_no) -{ - /* parsed with a v1 header here. that's ok as we only look at the - * header first which is the same for both versions */ - struct uac_iso_endpoint_descriptor *csep; - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - int attributes = 0; - - csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT); - - /* Creamware Noah has this descriptor after the 2nd endpoint */ - if (!csep && altsd->bNumEndpoints >= 2) - csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); - - if (!csep || csep->bLength < 7 || - csep->bDescriptorSubtype != UAC_EP_GENERAL) { - snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" - " class specific endpoint descriptor\n", - chip->dev->devnum, iface_no, - altsd->bAlternateSetting); - return 0; - } - - if (protocol == UAC_VERSION_1) { - attributes = csep->bmAttributes; - } else { - struct uac2_iso_endpoint_descriptor *csep2 = - (struct uac2_iso_endpoint_descriptor *) csep; - - attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX; - - /* emulate the endpoint attributes of a v1 device */ - if (csep2->bmControls & UAC2_CONTROL_PITCH) - attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL; - } - - return attributes; -} - -static struct uac2_input_terminal_descriptor * - snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) -{ - struct uac2_input_terminal_descriptor *term = NULL; - - while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - term, UAC_INPUT_TERMINAL))) { - if (term->bTerminalID == terminal_id) - return term; - } - - return NULL; -} - -static struct uac2_output_terminal_descriptor * - snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) -{ - struct uac2_output_terminal_descriptor *term = NULL; - - while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, - ctrl_iface->extralen, - term, UAC_OUTPUT_TERMINAL))) { - if (term->bTerminalID == terminal_id) - return term; - } - - return NULL; -} - -int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) -{ - struct usb_device *dev; - struct usb_interface *iface; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - int i, altno, err, stream; - int format = 0, num_channels = 0; - struct audioformat *fp = NULL; - int num, protocol, clock = 0; - struct uac_format_type_i_continuous_descriptor *fmt; - - dev = chip->dev; - - /* parse the interface's altsettings */ - iface = usb_ifnum_to_if(dev, iface_no); - - num = iface->num_altsetting; - - /* - * Dallas DS4201 workaround: It presents 5 altsettings, but the last - * one misses syncpipe, and does not produce any sound. - */ - if (chip->usb_id == USB_ID(0x04fa, 0x4201)) - num = 4; - - for (i = 0; i < num; i++) { - alts = &iface->altsetting[i]; - altsd = get_iface_desc(alts); - protocol = altsd->bInterfaceProtocol; - /* skip invalid one */ - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && - altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && - altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || - altsd->bNumEndpoints < 1 || - le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) - continue; - /* must be isochronous */ - if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != - USB_ENDPOINT_XFER_ISOC) - continue; - /* check direction */ - stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? - SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - altno = altsd->bAlternateSetting; - - if (snd_usb_apply_interface_quirk(chip, iface_no, altno)) - continue; - - /* get audio formats */ - switch (protocol) { - default: - snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", - dev->devnum, iface_no, altno, protocol); - protocol = UAC_VERSION_1; - /* fall through */ - - case UAC_VERSION_1: { - struct uac1_as_header_descriptor *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); - - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; - } - - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; - } - - format = le16_to_cpu(as->wFormatTag); /* remember the format value */ - break; - } - - case UAC_VERSION_2: { - struct uac2_input_terminal_descriptor *input_term; - struct uac2_output_terminal_descriptor *output_term; - struct uac2_as_header_descriptor *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); - - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; - } - - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; - } - - num_channels = as->bNrChannels; - format = le32_to_cpu(as->bmFormats); - - /* lookup the terminal associated to this interface - * to extract the clock */ - input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); - if (input_term) { - clock = input_term->bCSourceID; - break; - } - - output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); - if (output_term) { - clock = output_term->bCSourceID; - break; - } - - snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n", - dev->devnum, iface_no, altno, as->bTerminalLink); - continue; - } - } - - /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); - if (!fmt) { - snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", - dev->devnum, iface_no, altno); - continue; - } - if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || - ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", - dev->devnum, iface_no, altno); - continue; - } - - /* - * Blue Microphones workaround: The last altsetting is identical - * with the previous one, except for a larger packet size, but - * is actually a mislabeled two-channel setting; ignore it. - */ - if (fmt->bNrChannels == 1 && - fmt->bSubframeSize == 2 && - altno == 2 && num == 3 && - fp && fp->altsetting == 1 && fp->channels == 1 && - fp->formats == SNDRV_PCM_FMTBIT_S16_LE && - protocol == UAC_VERSION_1 && - le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == - fp->maxpacksize * 2) - continue; - - fp = kzalloc(sizeof(*fp), GFP_KERNEL); - if (! fp) { - snd_printk(KERN_ERR "cannot malloc\n"); - return -ENOMEM; - } - - fp->iface = iface_no; - fp->altsetting = altno; - fp->altset_idx = i; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = snd_usb_parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - /* num_channels is only set for v2 interfaces */ - fp->channels = num_channels; - if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) - fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) - * (fp->maxpacksize & 0x7ff); - fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); - fp->clock = clock; - - /* some quirks for attributes here */ - - switch (chip->usb_id) { - case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */ - /* Optoplay sets the sample rate attribute although - * it seems not supporting it in fact. - */ - fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; - break; - case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ - case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - /* doesn't set the sample rate attribute, but supports it */ - fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; - break; - case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ - case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ - case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ - case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is - an older model 77d:223) */ - /* - * plantronics headset and Griffin iMic have set adaptive-in - * although it's really not... - */ - fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; - else - fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; - break; - } - - /* ok, let's parse further... */ - if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { - kfree(fp->rate_table); - kfree(fp); - fp = NULL; - continue; - } - - snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); - err = snd_usb_add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp->rate_table); - kfree(fp); - return err; - } - /* try to set the interface... */ - usb_set_interface(chip->dev, iface_no, altno); - snd_usb_init_pitch(chip, iface_no, alts, fp); - snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max); - } - return 0; -} - diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 64dd0db023b2be..e5d8a6adf38fde 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -1,11 +1,4 @@ #ifndef __USBAUDIO_ENDPOINT_H #define __USBAUDIO_ENDPOINT_H -int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, - int iface_no); - -int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, - int stream, - struct audioformat *fp); - #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index cf61b03400263d..556edea28b9047 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -34,6 +34,7 @@ #include "endpoint.h" #include "pcm.h" #include "clock.h" +#include "stream.h" /* * handle the quirks for the contained interfaces @@ -106,7 +107,7 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, alts = &iface->altsetting[0]; altsd = get_iface_desc(alts); - err = snd_usb_parse_audio_endpoints(chip, altsd->bInterfaceNumber); + err = snd_usb_parse_audio_interface(chip, altsd->bInterfaceNumber); if (err < 0) { snd_printk(KERN_ERR "cannot setup if %d: error %d\n", altsd->bInterfaceNumber, err); @@ -147,7 +148,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, stream = (fp->endpoint & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = snd_usb_add_audio_endpoint(chip, stream, fp); + err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { kfree(fp); kfree(rate_table); @@ -254,7 +255,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, stream = (fp->endpoint & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = snd_usb_add_audio_endpoint(chip, stream, fp); + err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { kfree(fp); return err; diff --git a/sound/usb/stream.c b/sound/usb/stream.c new file mode 100644 index 00000000000000..4688d4c6208b93 --- /dev/null +++ b/sound/usb/stream.c @@ -0,0 +1,453 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +#include +#include +#include +#include +#include + +#include +#include + +#include "usbaudio.h" +#include "card.h" +#include "proc.h" +#include "quirks.h" +#include "endpoint.h" +#include "urb.h" +#include "pcm.h" +#include "helper.h" +#include "format.h" +#include "clock.h" +#include "stream.h" + +/* + * free a substream + */ +static void free_substream(struct snd_usb_substream *subs) +{ + struct list_head *p, *n; + + if (!subs->num_formats) + return; /* not initialized */ + list_for_each_safe(p, n, &subs->fmt_list) { + struct audioformat *fp = list_entry(p, struct audioformat, list); + kfree(fp->rate_table); + kfree(fp); + } + kfree(subs->rate_list.list); +} + + +/* + * free a usb stream instance + */ +static void snd_usb_audio_stream_free(struct snd_usb_stream *stream) +{ + free_substream(&stream->substream[0]); + free_substream(&stream->substream[1]); + list_del(&stream->list); + kfree(stream); +} + +static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) +{ + struct snd_usb_stream *stream = pcm->private_data; + if (stream) { + stream->pcm = NULL; + snd_usb_audio_stream_free(stream); + } +} + + +/* + * add this endpoint to the chip instance. + * if a stream with the same endpoint already exists, append to it. + * if not, create a new pcm stream. + */ +int snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp) +{ + struct list_head *p; + struct snd_usb_stream *as; + struct snd_usb_substream *subs; + struct snd_pcm *pcm; + int err; + + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + if (as->fmt_type != fp->fmt_type) + continue; + subs = &as->substream[stream]; + if (!subs->endpoint) + continue; + if (subs->endpoint == fp->endpoint) { + list_add_tail(&fp->list, &subs->fmt_list); + subs->num_formats++; + subs->formats |= fp->formats; + return 0; + } + } + /* look for an empty stream */ + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + if (as->fmt_type != fp->fmt_type) + continue; + subs = &as->substream[stream]; + if (subs->endpoint) + continue; + err = snd_pcm_new_stream(as->pcm, stream, 1); + if (err < 0) + return err; + snd_usb_init_substream(as, stream, fp); + return 0; + } + + /* create a new pcm */ + as = kzalloc(sizeof(*as), GFP_KERNEL); + if (!as) + return -ENOMEM; + as->pcm_index = chip->pcm_devs; + as->chip = chip; + as->fmt_type = fp->fmt_type; + err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, + stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0, + stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1, + &pcm); + if (err < 0) { + kfree(as); + return err; + } + as->pcm = pcm; + pcm->private_data = as; + pcm->private_free = snd_usb_audio_pcm_free; + pcm->info_flags = 0; + if (chip->pcm_devs > 0) + sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs); + else + strcpy(pcm->name, "USB Audio"); + + snd_usb_init_substream(as, stream, fp); + + list_add(&as->list, &chip->pcm_list); + chip->pcm_devs++; + + snd_usb_proc_pcm_format_add(as); + + return 0; +} + +static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int protocol, int iface_no) +{ + /* parsed with a v1 header here. that's ok as we only look at the + * header first which is the same for both versions */ + struct uac_iso_endpoint_descriptor *csep; + struct usb_interface_descriptor *altsd = get_iface_desc(alts); + int attributes = 0; + + csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT); + + /* Creamware Noah has this descriptor after the 2nd endpoint */ + if (!csep && altsd->bNumEndpoints >= 2) + csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); + + if (!csep || csep->bLength < 7 || + csep->bDescriptorSubtype != UAC_EP_GENERAL) { + snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" + " class specific endpoint descriptor\n", + chip->dev->devnum, iface_no, + altsd->bAlternateSetting); + return 0; + } + + if (protocol == UAC_VERSION_1) { + attributes = csep->bmAttributes; + } else { + struct uac2_iso_endpoint_descriptor *csep2 = + (struct uac2_iso_endpoint_descriptor *) csep; + + attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX; + + /* emulate the endpoint attributes of a v1 device */ + if (csep2->bmControls & UAC2_CONTROL_PITCH) + attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL; + } + + return attributes; +} + +static struct uac2_input_terminal_descriptor * + snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_input_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_INPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + +static struct uac2_output_terminal_descriptor * + snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_output_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_OUTPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +{ + struct usb_device *dev; + struct usb_interface *iface; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + int i, altno, err, stream; + int format = 0, num_channels = 0; + struct audioformat *fp = NULL; + int num, protocol, clock = 0; + struct uac_format_type_i_continuous_descriptor *fmt; + + dev = chip->dev; + + /* parse the interface's altsettings */ + iface = usb_ifnum_to_if(dev, iface_no); + + num = iface->num_altsetting; + + /* + * Dallas DS4201 workaround: It presents 5 altsettings, but the last + * one misses syncpipe, and does not produce any sound. + */ + if (chip->usb_id == USB_ID(0x04fa, 0x4201)) + num = 4; + + for (i = 0; i < num; i++) { + alts = &iface->altsetting[i]; + altsd = get_iface_desc(alts); + protocol = altsd->bInterfaceProtocol; + /* skip invalid one */ + if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && + altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || + altsd->bNumEndpoints < 1 || + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) + continue; + /* must be isochronous */ + if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != + USB_ENDPOINT_XFER_ISOC) + continue; + /* check direction */ + stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + altno = altsd->bAlternateSetting; + + if (snd_usb_apply_interface_quirk(chip, iface_no, altno)) + continue; + + /* get audio formats */ + switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", + dev->devnum, iface_no, altno, protocol); + protocol = UAC_VERSION_1; + /* fall through */ + + case UAC_VERSION_1: { + struct uac1_as_header_descriptor *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + break; + } + + case UAC_VERSION_2: { + struct uac2_input_terminal_descriptor *input_term; + struct uac2_output_terminal_descriptor *output_term; + struct uac2_as_header_descriptor *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + num_channels = as->bNrChannels; + format = le32_to_cpu(as->bmFormats); + + /* lookup the terminal associated to this interface + * to extract the clock */ + input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (input_term) { + clock = input_term->bCSourceID; + break; + } + + output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (output_term) { + clock = output_term->bCSourceID; + break; + } + + snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n", + dev->devnum, iface_no, altno, as->bTerminalLink); + continue; + } + } + + /* get format type */ + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); + if (!fmt) { + snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", + dev->devnum, iface_no, altno); + continue; + } + if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || + ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", + dev->devnum, iface_no, altno); + continue; + } + + /* + * Blue Microphones workaround: The last altsetting is identical + * with the previous one, except for a larger packet size, but + * is actually a mislabeled two-channel setting; ignore it. + */ + if (fmt->bNrChannels == 1 && + fmt->bSubframeSize == 2 && + altno == 2 && num == 3 && + fp && fp->altsetting == 1 && fp->channels == 1 && + fp->formats == SNDRV_PCM_FMTBIT_S16_LE && + protocol == UAC_VERSION_1 && + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == + fp->maxpacksize * 2) + continue; + + fp = kzalloc(sizeof(*fp), GFP_KERNEL); + if (! fp) { + snd_printk(KERN_ERR "cannot malloc\n"); + return -ENOMEM; + } + + fp->iface = iface_no; + fp->altsetting = altno; + fp->altset_idx = i; + fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; + fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = snd_usb_parse_datainterval(chip, alts); + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + /* num_channels is only set for v2 interfaces */ + fp->channels = num_channels; + if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) + fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) + * (fp->maxpacksize & 0x7ff); + fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); + fp->clock = clock; + + /* some quirks for attributes here */ + + switch (chip->usb_id) { + case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */ + /* Optoplay sets the sample rate attribute although + * it seems not supporting it in fact. + */ + fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; + break; + case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ + case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + /* doesn't set the sample rate attribute, but supports it */ + fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; + break; + case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ + case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ + case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ + case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is + an older model 77d:223) */ + /* + * plantronics headset and Griffin iMic have set adaptive-in + * although it's really not... + */ + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; + else + fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; + break; + } + + /* ok, let's parse further... */ + if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { + kfree(fp->rate_table); + kfree(fp); + fp = NULL; + continue; + } + + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); + err = snd_usb_add_audio_stream(chip, stream, fp); + if (err < 0) { + kfree(fp->rate_table); + kfree(fp); + return err; + } + /* try to set the interface... */ + usb_set_interface(chip->dev, iface_no, altno); + snd_usb_init_pitch(chip, iface_no, alts, fp); + snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max); + } + return 0; +} + diff --git a/sound/usb/stream.h b/sound/usb/stream.h new file mode 100644 index 00000000000000..c97f679fc84f7c --- /dev/null +++ b/sound/usb/stream.h @@ -0,0 +1,12 @@ +#ifndef __USBAUDIO_STREAM_H +#define __USBAUDIO_STREAM_H + +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, + int iface_no); + +int snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp); + +#endif /* __USBAUDIO_STREAM_H */ + From c731bc96ad641a5fa3d50a87b474652505507282 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 14 Sep 2011 12:46:57 +0200 Subject: [PATCH 24/45] ALSA: snd-usb: move code from urb.c to endpoint.c No code altered at this point, simply preparing for upcoming refactorizations. Signed-off-by: Daniel Mack Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/Makefile | 3 +- sound/usb/card.c | 1 - sound/usb/endpoint.c | 948 ++++++++++++++++++++++++++++++++++++++++++ sound/usb/endpoint.h | 17 + sound/usb/pcm.c | 2 +- sound/usb/stream.c | 1 - sound/usb/urb.c | 965 ------------------------------------------- sound/usb/urb.h | 21 - 8 files changed, 967 insertions(+), 991 deletions(-) delete mode 100644 sound/usb/urb.c delete mode 100644 sound/usb/urb.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 5390db00e09859..ac256dc4c6bed1 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -12,8 +12,7 @@ snd-usb-audio-objs := card.o \ pcm.o \ proc.o \ quirks.o \ - stream.o \ - urb.o + stream.o snd-usbmidi-lib-objs := midi.o diff --git a/sound/usb/card.c b/sound/usb/card.c index a3136afb2198b6..d2a79d166e0439 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -65,7 +65,6 @@ #include "helper.h" #include "debug.h" #include "pcm.h" -#include "urb.h" #include "format.h" #include "power.h" #include "stream.h" diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index b3ee7cf243dfde..81c6edecd86235 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -15,3 +15,951 @@ * */ +#include +#include +#include +#include + +#include +#include + +#include "usbaudio.h" +#include "helper.h" +#include "card.h" +#include "endpoint.h" +#include "pcm.h" + +/* + * convert a sampling rate into our full speed format (fs/1000 in Q16.16) + * this will overflow at approx 524 kHz + */ +static inline unsigned get_usb_full_speed_rate(unsigned int rate) +{ + return ((rate << 13) + 62) / 125; +} + +/* + * convert a sampling rate into USB high speed format (fs/8000 in Q16.16) + * this will overflow at approx 4 MHz + */ +static inline unsigned get_usb_high_speed_rate(unsigned int rate) +{ + return ((rate << 10) + 62) / 125; +} + +/* + * unlink active urbs. + */ +static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) +{ + struct snd_usb_audio *chip = subs->stream->chip; + unsigned int i; + int async; + + subs->running = 0; + + if (!force && subs->stream->chip->shutdown) /* to be sure... */ + return -EBADFD; + + async = !can_sleep && chip->async_unlink; + + if (!async && in_interrupt()) + return 0; + + for (i = 0; i < subs->nurbs; i++) { + if (test_bit(i, &subs->active_mask)) { + if (!test_and_set_bit(i, &subs->unlink_mask)) { + struct urb *u = subs->dataurb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } + } + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (test_bit(i+16, &subs->active_mask)) { + if (!test_and_set_bit(i+16, &subs->unlink_mask)) { + struct urb *u = subs->syncurb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } + } + } + } + return 0; +} + + +/* + * release a urb data + */ +static void release_urb_ctx(struct snd_urb_ctx *u) +{ + if (u->urb) { + if (u->buffer_size) + usb_free_coherent(u->subs->dev, u->buffer_size, + u->urb->transfer_buffer, + u->urb->transfer_dma); + usb_free_urb(u->urb); + u->urb = NULL; + } +} + +/* + * wait until all urbs are processed. + */ +static int wait_clear_urbs(struct snd_usb_substream *subs) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(1000); + unsigned int i; + int alive; + + do { + alive = 0; + for (i = 0; i < subs->nurbs; i++) { + if (test_bit(i, &subs->active_mask)) + alive++; + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (test_bit(i + 16, &subs->active_mask)) + alive++; + } + } + if (! alive) + break; + schedule_timeout_uninterruptible(1); + } while (time_before(jiffies, end_time)); + if (alive) + snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); + return 0; +} + +/* + * release a substream + */ +void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) +{ + int i; + + /* stop urbs (to be sure) */ + deactivate_urbs(subs, force, 1); + wait_clear_urbs(subs); + + for (i = 0; i < MAX_URBS; i++) + release_urb_ctx(&subs->dataurb[i]); + for (i = 0; i < SYNC_URBS; i++) + release_urb_ctx(&subs->syncurb[i]); + usb_free_coherent(subs->dev, SYNC_URBS * 4, + subs->syncbuf, subs->sync_dma); + subs->syncbuf = NULL; + subs->nurbs = 0; +} + +/* + * complete callback from data urb + */ +static void snd_complete_urb(struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_substream *subs = ctx->subs; + struct snd_pcm_substream *substream = ctx->subs->pcm_substream; + int err = 0; + + if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || + !subs->running || /* can be stopped during retire callback */ + (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || + (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { + clear_bit(ctx->index, &subs->active_mask); + if (err < 0) { + snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + } + } +} + + +/* + * complete callback from sync urb + */ +static void snd_complete_sync_urb(struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_substream *subs = ctx->subs; + struct snd_pcm_substream *substream = ctx->subs->pcm_substream; + int err = 0; + + if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || + !subs->running || /* can be stopped during retire callback */ + (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || + (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { + clear_bit(ctx->index + 16, &subs->active_mask); + if (err < 0) { + snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + } + } +} + + +/* + * initialize a substream for plaback/capture + */ +int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, + unsigned int period_bytes, + unsigned int rate, + unsigned int frame_bits) +{ + unsigned int maxsize, i; + int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int urb_packs, total_packs, packs_per_ms; + struct snd_usb_audio *chip = subs->stream->chip; + + /* calculate the frequency in 16.16 format */ + if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) + subs->freqn = get_usb_full_speed_rate(rate); + else + subs->freqn = get_usb_high_speed_rate(rate); + subs->freqm = subs->freqn; + subs->freqshift = INT_MIN; + /* calculate max. frequency */ + if (subs->maxpacksize) { + /* whatever fits into a max. size packet */ + maxsize = subs->maxpacksize; + subs->freqmax = (maxsize / (frame_bits >> 3)) + << (16 - subs->datainterval); + } else { + /* no max. packet size: just take 25% higher than nominal */ + subs->freqmax = subs->freqn + (subs->freqn >> 2); + maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - subs->datainterval); + } + subs->phase = 0; + + if (subs->fill_max) + subs->curpacksize = subs->maxpacksize; + else + subs->curpacksize = maxsize; + + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) + packs_per_ms = 8 >> subs->datainterval; + else + packs_per_ms = 1; + + if (is_playback) { + urb_packs = max(chip->nrpacks, 1); + urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); + } else + urb_packs = 1; + urb_packs *= packs_per_ms; + if (subs->syncpipe) + urb_packs = min(urb_packs, 1U << subs->syncinterval); + + /* decide how many packets to be used */ + if (is_playback) { + unsigned int minsize, maxpacks; + /* determine how small a packet can be */ + minsize = (subs->freqn >> (16 - subs->datainterval)) + * (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (subs->syncpipe) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + total_packs = (period_bytes + minsize - 1) / minsize; + /* we need at least two URBs for queueing */ + if (total_packs < 2) { + total_packs = 2; + } else { + /* and we don't want too long a queue either */ + maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); + total_packs = min(total_packs, maxpacks); + } + } else { + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + total_packs = MAX_URBS * urb_packs; + } + subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; + if (subs->nurbs > MAX_URBS) { + /* too much... */ + subs->nurbs = MAX_URBS; + total_packs = MAX_URBS * urb_packs; + } else if (subs->nurbs < 2) { + /* too little - we need at least two packets + * to ensure contiguous playback/capture + */ + subs->nurbs = 2; + } + + /* allocate and initialize data urbs */ + for (i = 0; i < subs->nurbs; i++) { + struct snd_urb_ctx *u = &subs->dataurb[i]; + u->index = i; + u->subs = subs; + u->packets = (i + 1) * total_packs / subs->nurbs + - i * total_packs / subs->nurbs; + u->buffer_size = maxsize * u->packets; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) + u->packets++; /* for transfer delimiter */ + u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = + usb_alloc_coherent(subs->dev, u->buffer_size, + GFP_KERNEL, &u->urb->transfer_dma); + if (!u->urb->transfer_buffer) + goto out_of_memory; + u->urb->pipe = subs->datapipe; + u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; + u->urb->interval = 1 << subs->datainterval; + u->urb->context = u; + u->urb->complete = snd_complete_urb; + } + + if (subs->syncpipe) { + /* allocate and initialize sync urbs */ + subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4, + GFP_KERNEL, &subs->sync_dma); + if (!subs->syncbuf) + goto out_of_memory; + for (i = 0; i < SYNC_URBS; i++) { + struct snd_urb_ctx *u = &subs->syncurb[i]; + u->index = i; + u->subs = subs; + u->packets = 1; + u->urb = usb_alloc_urb(1, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = subs->syncbuf + i * 4; + u->urb->transfer_dma = subs->sync_dma + i * 4; + u->urb->transfer_buffer_length = 4; + u->urb->pipe = subs->syncpipe; + u->urb->transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + u->urb->number_of_packets = 1; + u->urb->interval = 1 << subs->syncinterval; + u->urb->context = u; + u->urb->complete = snd_complete_sync_urb; + } + } + return 0; + +out_of_memory: + snd_usb_release_substream_urbs(subs, 0); + return -ENOMEM; +} + +/* + * prepare urb for full speed capture sync pipe + * + * fill the length and offset of each urb descriptor. + * the fixed 10.14 frequency is passed through the pipe. + */ +static int prepare_capture_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned char *cp = urb->transfer_buffer; + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].offset = 0; + cp[0] = subs->freqn >> 2; + cp[1] = subs->freqn >> 10; + cp[2] = subs->freqn >> 18; + return 0; +} + +/* + * prepare urb for high speed capture sync pipe + * + * fill the length and offset of each urb descriptor. + * the fixed 12.13 frequency is passed as 16.16 through the pipe. + */ +static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned char *cp = urb->transfer_buffer; + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = 4; + urb->iso_frame_desc[0].offset = 0; + cp[0] = subs->freqn; + cp[1] = subs->freqn >> 8; + cp[2] = subs->freqn >> 16; + cp[3] = subs->freqn >> 24; + return 0; +} + +/* + * process after capture sync complete + * - nothing to do + */ +static int retire_capture_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + return 0; +} + +/* + * prepare urb for capture data pipe + * + * fill the offset and length of each descriptor. + * + * we use a temporary buffer to write the captured data. + * since the length of written data is determined by host, we cannot + * write onto the pcm buffer directly... the data is thus copied + * later at complete callback to the global buffer. + */ +static int prepare_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + int i, offs; + struct snd_urb_ctx *ctx = urb->context; + + offs = 0; + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + for (i = 0; i < ctx->packets; i++) { + urb->iso_frame_desc[i].offset = offs; + urb->iso_frame_desc[i].length = subs->curpacksize; + offs += subs->curpacksize; + } + urb->transfer_buffer_length = offs; + urb->number_of_packets = ctx->packets; + return 0; +} + +/* + * process after capture complete + * + * copy the data from each desctiptor to the pcm buffer, and + * update the current position. + */ +static int retire_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned long flags; + unsigned char *cp; + int i; + unsigned int stride, frames, bytes, oldptr; + int period_elapsed = 0; + + stride = runtime->frame_bits >> 3; + + for (i = 0; i < urb->number_of_packets; i++) { + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; + if (urb->iso_frame_desc[i].status) { + snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); + // continue; + } + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } + /* update the current pointer */ + spin_lock_irqsave(&subs->lock, flags); + oldptr = subs->hwptr_done; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; + subs->transfer_done += frames; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + } + spin_unlock_irqrestore(&subs->lock, flags); + /* copy a data chunk */ + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); + } else { + memcpy(runtime->dma_area + oldptr, cp, bytes); + } + } + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); + return 0; +} + +/* + * Process after capture complete when paused. Nothing to do. + */ +static int retire_paused_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + return 0; +} + + +/* + * prepare urb for playback sync pipe + * + * set up the offset and length to receive the current frequency. + */ +static int prepare_playback_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); + urb->iso_frame_desc[0].offset = 0; + return 0; +} + +/* + * process after playback sync complete + * + * Full speed devices report feedback values in 10.14 format as samples per + * frame, high speed devices in 16.16 format as samples per microframe. + * Because the Audio Class 1 spec was written before USB 2.0, many high speed + * devices use a wrong interpretation, some others use an entirely different + * format. Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. + */ +static int retire_playback_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int f; + int shift; + unsigned long flags; + + if (urb->iso_frame_desc[0].status != 0 || + urb->iso_frame_desc[0].actual_length < 3) + return 0; + + f = le32_to_cpup(urb->transfer_buffer); + if (urb->iso_frame_desc[0].actual_length == 3) + f &= 0x00ffffff; + else + f &= 0x0fffffff; + if (f == 0) + return 0; + + if (unlikely(subs->freqshift == INT_MIN)) { + /* + * The first time we see a feedback value, determine its format + * by shifting it left or right until it matches the nominal + * frequency value. This assumes that the feedback does not + * differ from the nominal value more than +50% or -25%. + */ + shift = 0; + while (f < subs->freqn - subs->freqn / 4) { + f <<= 1; + shift++; + } + while (f > subs->freqn + subs->freqn / 2) { + f >>= 1; + shift--; + } + subs->freqshift = shift; + } + else if (subs->freqshift >= 0) + f <<= subs->freqshift; + else + f >>= -subs->freqshift; + + if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { + /* + * If the frequency looks valid, set it. + * This value is referred to in prepare_playback_urb(). + */ + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } else { + /* + * Out of range; maybe the shift value is wrong. + * Reset it so that we autodetect again the next time. + */ + subs->freqshift = INT_MIN; + } + + return 0; +} + +/* determine the number of frames in the next packet */ +static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) +{ + if (subs->fill_max) + return subs->maxframesize; + else { + subs->phase = (subs->phase & 0xffff) + + (subs->freqm << subs->datainterval); + return min(subs->phase >> 16, subs->maxframesize); + } +} + +/* + * Prepare urb for streaming before playback starts or when paused. + * + * We don't have any data, so we send silence. + */ +static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int i, offs, counts; + struct snd_urb_ctx *ctx = urb->context; + int stride = runtime->frame_bits >> 3; + + offs = 0; + urb->dev = ctx->subs->dev; + for (i = 0; i < ctx->packets; ++i) { + counts = snd_usb_audio_next_packet_size(subs); + urb->iso_frame_desc[i].offset = offs * stride; + urb->iso_frame_desc[i].length = counts * stride; + offs += counts; + } + urb->number_of_packets = ctx->packets; + urb->transfer_buffer_length = offs * stride; + memset(urb->transfer_buffer, + runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, + offs * stride); + return 0; +} + +/* + * prepare urb for playback data pipe + * + * Since a URB can handle only a single linear buffer, we must use double + * buffering when the data to be transferred overflows the buffer boundary. + * To avoid inconsistencies when updating hwptr_done, we use double buffering + * for all URBs. + */ +static int prepare_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + int i, stride; + unsigned int counts, frames, bytes; + unsigned long flags; + int period_elapsed = 0; + struct snd_urb_ctx *ctx = urb->context; + + stride = runtime->frame_bits >> 3; + + frames = 0; + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->number_of_packets = 0; + spin_lock_irqsave(&subs->lock, flags); + for (i = 0; i < ctx->packets; i++) { + counts = snd_usb_audio_next_packet_size(subs); + /* set up descriptor */ + urb->iso_frame_desc[i].offset = frames * stride; + urb->iso_frame_desc[i].length = counts * stride; + frames += counts; + urb->number_of_packets++; + subs->transfer_done += counts; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { + if (subs->transfer_done > 0) { + /* FIXME: fill-max mode is not + * supported yet */ + frames -= subs->transfer_done; + counts -= subs->transfer_done; + urb->iso_frame_desc[i].length = + counts * stride; + subs->transfer_done = 0; + } + i++; + if (i < ctx->packets) { + /* add a transfer delimiter */ + urb->iso_frame_desc[i].offset = + frames * stride; + urb->iso_frame_desc[i].length = 0; + urb->number_of_packets++; + } + break; + } + } + if (period_elapsed) /* finish at the period boundary */ + break; + } + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { + /* err, the transferred area goes over buffer boundary. */ + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); + } else { + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes); + } + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + + /* update delay with exact number of samples queued */ + runtime->delay = subs->last_delay; + runtime->delay += frames; + subs->last_delay = runtime->delay; + + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + + spin_unlock_irqrestore(&subs->lock, flags); + urb->transfer_buffer_length = bytes; + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); + return 0; +} + +/* + * process after playback data complete + * - decrease the delay count again + */ +static int retire_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned long flags; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + int est_delay; + + spin_lock_irqsave(&subs->lock, flags); + + est_delay = snd_usb_pcm_delay(subs, runtime->rate); + /* update delay with exact number of samples played */ + if (processed > subs->last_delay) + subs->last_delay = 0; + else + subs->last_delay -= processed; + runtime->delay = subs->last_delay; + + /* + * Report when delay estimate is off by more than 2ms. + * The error should be lower than 2ms since the estimate relies + * on two reads of a counter updated every ms. + */ + if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) + snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", + est_delay, subs->last_delay); + + spin_unlock_irqrestore(&subs->lock, flags); + return 0; +} + +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} + +/* + * set up and start data/sync urbs + */ +static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) +{ + unsigned int i; + int err; + + if (subs->stream->chip->shutdown) + return -EBADFD; + + for (i = 0; i < subs->nurbs; i++) { + if (snd_BUG_ON(!subs->dataurb[i].urb)) + return -EINVAL; + if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { + snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); + goto __error; + } + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (snd_BUG_ON(!subs->syncurb[i].urb)) + return -EINVAL; + if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { + snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); + goto __error; + } + } + } + + subs->active_mask = 0; + subs->unlink_mask = 0; + subs->running = 1; + for (i = 0; i < subs->nurbs; i++) { + err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit datapipe " + "for urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; + } + set_bit(i, &subs->active_mask); + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit syncpipe " + "for urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; + } + set_bit(i + 16, &subs->active_mask); + } + } + return 0; + + __error: + // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); + deactivate_urbs(subs, 0, 0); + return -EPIPE; +} + + +/* + */ +static struct snd_urb_ops audio_urb_ops[2] = { + { + .prepare = prepare_nodata_playback_urb, + .retire = retire_playback_urb, + .prepare_sync = prepare_playback_sync_urb, + .retire_sync = retire_playback_sync_urb, + }, + { + .prepare = prepare_capture_urb, + .retire = retire_capture_urb, + .prepare_sync = prepare_capture_sync_urb, + .retire_sync = retire_capture_sync_urb, + }, +}; + +/* + * initialize the substream instance. + */ + +void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; + + INIT_LIST_HEAD(&subs->fmt_list); + spin_lock_init(&subs->lock); + + subs->stream = as; + subs->direction = stream; + subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; + subs->ops = audio_urb_ops[stream]; + if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) + subs->ops.prepare_sync = prepare_capture_sync_urb_hs; + + snd_usb_set_pcm_ops(as->pcm, stream); + + list_add_tail(&fp->list, &subs->fmt_list); + subs->formats |= fp->formats; + subs->endpoint = fp->endpoint; + subs->num_formats++; + subs->fmt_type = fp->fmt_type; +} + +int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->ops.prepare = prepare_playback_urb; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + return deactivate_urbs(subs, 0, 0); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->ops.prepare = prepare_nodata_playback_urb; + return 0; + } + + return -EINVAL; +} + +int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + subs->ops.retire = retire_capture_urb; + return start_urbs(subs, substream->runtime); + case SNDRV_PCM_TRIGGER_STOP: + return deactivate_urbs(subs, 0, 0); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->ops.retire = retire_paused_capture_urb; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->ops.retire = retire_capture_urb; + return 0; + } + + return -EINVAL; +} + +int snd_usb_substream_prepare(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime) +{ + /* clear urbs (to be sure) */ + deactivate_urbs(subs, 0, 1); + wait_clear_urbs(subs); + + /* for playback, submit the URBs now; otherwise, the first hwptr_done + * updates for all URBs would happen at the same time when starting */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + subs->ops.prepare = prepare_nodata_playback_urb; + return start_urbs(subs, runtime); + } + + return 0; +} + diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index e5d8a6adf38fde..88eb63a636eb28 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -1,4 +1,21 @@ #ifndef __USBAUDIO_ENDPOINT_H #define __USBAUDIO_ENDPOINT_H +void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, + struct audioformat *fp); + +int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, + unsigned int period_bytes, + unsigned int rate, + unsigned int frame_bits); + +void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); + +int snd_usb_substream_prepare(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime); + +int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); +int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); + #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0b699ca1957ef8..b5bc870878dbd7 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -28,7 +28,7 @@ #include "card.h" #include "quirks.h" #include "debug.h" -#include "urb.h" +#include "endpoint.h" #include "helper.h" #include "pcm.h" #include "clock.h" diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 4688d4c6208b93..5ff8010b2d6f59 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -29,7 +29,6 @@ #include "proc.h" #include "quirks.h" #include "endpoint.h" -#include "urb.h" #include "pcm.h" #include "helper.h" #include "format.h" diff --git a/sound/usb/urb.c b/sound/usb/urb.c deleted file mode 100644 index b4dcccc237dc9a..00000000000000 --- a/sound/usb/urb.c +++ /dev/null @@ -1,965 +0,0 @@ -/* - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include - -#include -#include - -#include "usbaudio.h" -#include "helper.h" -#include "card.h" -#include "urb.h" -#include "pcm.h" - -/* - * convert a sampling rate into our full speed format (fs/1000 in Q16.16) - * this will overflow at approx 524 kHz - */ -static inline unsigned get_usb_full_speed_rate(unsigned int rate) -{ - return ((rate << 13) + 62) / 125; -} - -/* - * convert a sampling rate into USB high speed format (fs/8000 in Q16.16) - * this will overflow at approx 4 MHz - */ -static inline unsigned get_usb_high_speed_rate(unsigned int rate) -{ - return ((rate << 10) + 62) / 125; -} - -/* - * unlink active urbs. - */ -static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) -{ - struct snd_usb_audio *chip = subs->stream->chip; - unsigned int i; - int async; - - subs->running = 0; - - if (!force && subs->stream->chip->shutdown) /* to be sure... */ - return -EBADFD; - - async = !can_sleep && chip->async_unlink; - - if (!async && in_interrupt()) - return 0; - - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) { - if (!test_and_set_bit(i, &subs->unlink_mask)) { - struct urb *u = subs->dataurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i+16, &subs->active_mask)) { - if (!test_and_set_bit(i+16, &subs->unlink_mask)) { - struct urb *u = subs->syncurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } - } - } - return 0; -} - - -/* - * release a urb data - */ -static void release_urb_ctx(struct snd_urb_ctx *u) -{ - if (u->urb) { - if (u->buffer_size) - usb_free_coherent(u->subs->dev, u->buffer_size, - u->urb->transfer_buffer, - u->urb->transfer_dma); - usb_free_urb(u->urb); - u->urb = NULL; - } -} - -/* - * wait until all urbs are processed. - */ -static int wait_clear_urbs(struct snd_usb_substream *subs) -{ - unsigned long end_time = jiffies + msecs_to_jiffies(1000); - unsigned int i; - int alive; - - do { - alive = 0; - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) - alive++; - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i + 16, &subs->active_mask)) - alive++; - } - } - if (! alive) - break; - schedule_timeout_uninterruptible(1); - } while (time_before(jiffies, end_time)); - if (alive) - snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); - return 0; -} - -/* - * release a substream - */ -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) -{ - int i; - - /* stop urbs (to be sure) */ - deactivate_urbs(subs, force, 1); - wait_clear_urbs(subs); - - for (i = 0; i < MAX_URBS; i++) - release_urb_ctx(&subs->dataurb[i]); - for (i = 0; i < SYNC_URBS; i++) - release_urb_ctx(&subs->syncurb[i]); - usb_free_coherent(subs->dev, SYNC_URBS * 4, - subs->syncbuf, subs->sync_dma); - subs->syncbuf = NULL; - subs->nurbs = 0; -} - -/* - * complete callback from data urb - */ -static void snd_complete_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} - - -/* - * complete callback from sync urb - */ -static void snd_complete_sync_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index + 16, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} - - -/* - * initialize a substream for plaback/capture - */ -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits) -{ - unsigned int maxsize, i; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int urb_packs, total_packs, packs_per_ms; - struct snd_usb_audio *chip = subs->stream->chip; - - /* calculate the frequency in 16.16 format */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->freqn = get_usb_full_speed_rate(rate); - else - subs->freqn = get_usb_high_speed_rate(rate); - subs->freqm = subs->freqn; - subs->freqshift = INT_MIN; - /* calculate max. frequency */ - if (subs->maxpacksize) { - /* whatever fits into a max. size packet */ - maxsize = subs->maxpacksize; - subs->freqmax = (maxsize / (frame_bits >> 3)) - << (16 - subs->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - subs->freqmax = subs->freqn + (subs->freqn >> 2); - maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - subs->datainterval); - } - subs->phase = 0; - - if (subs->fill_max) - subs->curpacksize = subs->maxpacksize; - else - subs->curpacksize = maxsize; - - if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) - packs_per_ms = 8 >> subs->datainterval; - else - packs_per_ms = 1; - - if (is_playback) { - urb_packs = max(chip->nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); - } else - urb_packs = 1; - urb_packs *= packs_per_ms; - if (subs->syncpipe) - urb_packs = min(urb_packs, 1U << subs->syncinterval); - - /* decide how many packets to be used */ - if (is_playback) { - unsigned int minsize, maxpacks; - /* determine how small a packet can be */ - minsize = (subs->freqn >> (16 - subs->datainterval)) - * (frame_bits >> 3); - /* with sync from device, assume it can be 12% lower */ - if (subs->syncpipe) - minsize -= minsize >> 3; - minsize = max(minsize, 1u); - total_packs = (period_bytes + minsize - 1) / minsize; - /* we need at least two URBs for queueing */ - if (total_packs < 2) { - total_packs = 2; - } else { - /* and we don't want too long a queue either */ - maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); - total_packs = min(total_packs, maxpacks); - } - } else { - while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) - urb_packs >>= 1; - total_packs = MAX_URBS * urb_packs; - } - subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (subs->nurbs > MAX_URBS) { - /* too much... */ - subs->nurbs = MAX_URBS; - total_packs = MAX_URBS * urb_packs; - } else if (subs->nurbs < 2) { - /* too little - we need at least two packets - * to ensure contiguous playback/capture - */ - subs->nurbs = 2; - } - - /* allocate and initialize data urbs */ - for (i = 0; i < subs->nurbs; i++) { - struct snd_urb_ctx *u = &subs->dataurb[i]; - u->index = i; - u->subs = subs; - u->packets = (i + 1) * total_packs / subs->nurbs - - i * total_packs / subs->nurbs; - u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) - u->packets++; /* for transfer delimiter */ - u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = - usb_alloc_coherent(subs->dev, u->buffer_size, - GFP_KERNEL, &u->urb->transfer_dma); - if (!u->urb->transfer_buffer) - goto out_of_memory; - u->urb->pipe = subs->datapipe; - u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; - u->urb->interval = 1 << subs->datainterval; - u->urb->context = u; - u->urb->complete = snd_complete_urb; - } - - if (subs->syncpipe) { - /* allocate and initialize sync urbs */ - subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4, - GFP_KERNEL, &subs->sync_dma); - if (!subs->syncbuf) - goto out_of_memory; - for (i = 0; i < SYNC_URBS; i++) { - struct snd_urb_ctx *u = &subs->syncurb[i]; - u->index = i; - u->subs = subs; - u->packets = 1; - u->urb = usb_alloc_urb(1, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = subs->syncbuf + i * 4; - u->urb->transfer_dma = subs->sync_dma + i * 4; - u->urb->transfer_buffer_length = 4; - u->urb->pipe = subs->syncpipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; - u->urb->number_of_packets = 1; - u->urb->interval = 1 << subs->syncinterval; - u->urb->context = u; - u->urb->complete = snd_complete_sync_urb; - } - } - return 0; - -out_of_memory: - snd_usb_release_substream_urbs(subs, 0); - return -ENOMEM; -} - -/* - * prepare urb for full speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 10.14 frequency is passed through the pipe. - */ -static int prepare_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn >> 2; - cp[1] = subs->freqn >> 10; - cp[2] = subs->freqn >> 18; - return 0; -} - -/* - * prepare urb for high speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 12.13 frequency is passed as 16.16 through the pipe. - */ -static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn; - cp[1] = subs->freqn >> 8; - cp[2] = subs->freqn >> 16; - cp[3] = subs->freqn >> 24; - return 0; -} - -/* - * process after capture sync complete - * - nothing to do - */ -static int retire_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; -} - -/* - * prepare urb for capture data pipe - * - * fill the offset and length of each descriptor. - * - * we use a temporary buffer to write the captured data. - * since the length of written data is determined by host, we cannot - * write onto the pcm buffer directly... the data is thus copied - * later at complete callback to the global buffer. - */ -static int prepare_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, offs; - struct snd_urb_ctx *ctx = urb->context; - - offs = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - for (i = 0; i < ctx->packets; i++) { - urb->iso_frame_desc[i].offset = offs; - urb->iso_frame_desc[i].length = subs->curpacksize; - offs += subs->curpacksize; - } - urb->transfer_buffer_length = offs; - urb->number_of_packets = ctx->packets; - return 0; -} - -/* - * process after capture complete - * - * copy the data from each desctiptor to the pcm buffer, and - * update the current position. - */ -static int retire_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - unsigned char *cp; - int i; - unsigned int stride, frames, bytes, oldptr; - int period_elapsed = 0; - - stride = runtime->frame_bits >> 3; - - for (i = 0; i < urb->number_of_packets; i++) { - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; - if (urb->iso_frame_desc[i].status) { - snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); - // continue; - } - bytes = urb->iso_frame_desc[i].actual_length; - frames = bytes / stride; - if (!subs->txfr_quirk) - bytes = frames * stride; - if (bytes % (runtime->sample_bits >> 3) != 0) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - int oldbytes = bytes; -#endif - bytes = frames * stride; - snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", - oldbytes, bytes); - } - /* update the current pointer */ - spin_lock_irqsave(&subs->lock, flags); - oldptr = subs->hwptr_done; - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - frames = (bytes + (oldptr % stride)) / stride; - subs->transfer_done += frames; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - } - spin_unlock_irqrestore(&subs->lock, flags); - /* copy a data chunk */ - if (oldptr + bytes > runtime->buffer_size * stride) { - unsigned int bytes1 = - runtime->buffer_size * stride - oldptr; - memcpy(runtime->dma_area + oldptr, cp, bytes1); - memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); - } else { - memcpy(runtime->dma_area + oldptr, cp, bytes); - } - } - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * Process after capture complete when paused. Nothing to do. - */ -static int retire_paused_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; -} - - -/* - * prepare urb for playback sync pipe - * - * set up the offset and length to receive the current frequency. - */ -static int prepare_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); - urb->iso_frame_desc[0].offset = 0; - return 0; -} - -/* - * process after playback sync complete - * - * Full speed devices report feedback values in 10.14 format as samples per - * frame, high speed devices in 16.16 format as samples per microframe. - * Because the Audio Class 1 spec was written before USB 2.0, many high speed - * devices use a wrong interpretation, some others use an entirely different - * format. Therefore, we cannot predict what format any particular device uses - * and must detect it automatically. - */ -static int retire_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - int shift; - unsigned long flags; - - if (urb->iso_frame_desc[0].status != 0 || - urb->iso_frame_desc[0].actual_length < 3) - return 0; - - f = le32_to_cpup(urb->transfer_buffer); - if (urb->iso_frame_desc[0].actual_length == 3) - f &= 0x00ffffff; - else - f &= 0x0fffffff; - if (f == 0) - return 0; - - if (unlikely(subs->freqshift == INT_MIN)) { - /* - * The first time we see a feedback value, determine its format - * by shifting it left or right until it matches the nominal - * frequency value. This assumes that the feedback does not - * differ from the nominal value more than +50% or -25%. - */ - shift = 0; - while (f < subs->freqn - subs->freqn / 4) { - f <<= 1; - shift++; - } - while (f > subs->freqn + subs->freqn / 2) { - f >>= 1; - shift--; - } - subs->freqshift = shift; - } - else if (subs->freqshift >= 0) - f <<= subs->freqshift; - else - f >>= -subs->freqshift; - - if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { - /* - * If the frequency looks valid, set it. - * This value is referred to in prepare_playback_urb(). - */ - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } else { - /* - * Out of range; maybe the shift value is wrong. - * Reset it so that we autodetect again the next time. - */ - subs->freqshift = INT_MIN; - } - - return 0; -} - -/* determine the number of frames in the next packet */ -static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) -{ - if (subs->fill_max) - return subs->maxframesize; - else { - subs->phase = (subs->phase & 0xffff) - + (subs->freqm << subs->datainterval); - return min(subs->phase >> 16, subs->maxframesize); - } -} - -/* - * Prepare urb for streaming before playback starts or when paused. - * - * We don't have any data, so we send silence. - */ -static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int i, offs, counts; - struct snd_urb_ctx *ctx = urb->context; - int stride = runtime->frame_bits >> 3; - - offs = 0; - urb->dev = ctx->subs->dev; - for (i = 0; i < ctx->packets; ++i) { - counts = snd_usb_audio_next_packet_size(subs); - urb->iso_frame_desc[i].offset = offs * stride; - urb->iso_frame_desc[i].length = counts * stride; - offs += counts; - } - urb->number_of_packets = ctx->packets; - urb->transfer_buffer_length = offs * stride; - memset(urb->transfer_buffer, - runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, - offs * stride); - return 0; -} - -/* - * prepare urb for playback data pipe - * - * Since a URB can handle only a single linear buffer, we must use double - * buffering when the data to be transferred overflows the buffer boundary. - * To avoid inconsistencies when updating hwptr_done, we use double buffering - * for all URBs. - */ -static int prepare_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, stride; - unsigned int counts, frames, bytes; - unsigned long flags; - int period_elapsed = 0; - struct snd_urb_ctx *ctx = urb->context; - - stride = runtime->frame_bits >> 3; - - frames = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->number_of_packets = 0; - spin_lock_irqsave(&subs->lock, flags); - for (i = 0; i < ctx->packets; i++) { - counts = snd_usb_audio_next_packet_size(subs); - /* set up descriptor */ - urb->iso_frame_desc[i].offset = frames * stride; - urb->iso_frame_desc[i].length = counts * stride; - frames += counts; - urb->number_of_packets++; - subs->transfer_done += counts; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) { - if (subs->transfer_done > 0) { - /* FIXME: fill-max mode is not - * supported yet */ - frames -= subs->transfer_done; - counts -= subs->transfer_done; - urb->iso_frame_desc[i].length = - counts * stride; - subs->transfer_done = 0; - } - i++; - if (i < ctx->packets) { - /* add a transfer delimiter */ - urb->iso_frame_desc[i].offset = - frames * stride; - urb->iso_frame_desc[i].length = 0; - urb->number_of_packets++; - } - break; - } - } - if (period_elapsed) /* finish at the period boundary */ - break; - } - bytes = frames * stride; - if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { - /* err, the transferred area goes over buffer boundary. */ - unsigned int bytes1 = - runtime->buffer_size * stride - subs->hwptr_done; - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes1); - memcpy(urb->transfer_buffer + bytes1, - runtime->dma_area, bytes - bytes1); - } else { - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes); - } - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - - /* update delay with exact number of samples queued */ - runtime->delay = subs->last_delay; - runtime->delay += frames; - subs->last_delay = runtime->delay; - - /* realign last_frame_number */ - subs->last_frame_number = usb_get_current_frame_number(subs->dev); - subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ - - spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = bytes; - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * process after playback data complete - * - decrease the delay count again - */ -static int retire_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - int stride = runtime->frame_bits >> 3; - int processed = urb->transfer_buffer_length / stride; - int est_delay; - - spin_lock_irqsave(&subs->lock, flags); - - est_delay = snd_usb_pcm_delay(subs, runtime->rate); - /* update delay with exact number of samples played */ - if (processed > subs->last_delay) - subs->last_delay = 0; - else - subs->last_delay -= processed; - runtime->delay = subs->last_delay; - - /* - * Report when delay estimate is off by more than 2ms. - * The error should be lower than 2ms since the estimate relies - * on two reads of a counter updated every ms. - */ - if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) - snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", - est_delay, subs->last_delay); - - spin_unlock_irqrestore(&subs->lock, flags); - return 0; -} - -static const char *usb_error_string(int err) -{ - switch (err) { - case -ENODEV: - return "no device"; - case -ENOENT: - return "endpoint not enabled"; - case -EPIPE: - return "endpoint stalled"; - case -ENOSPC: - return "not enough bandwidth"; - case -ESHUTDOWN: - return "device disabled"; - case -EHOSTUNREACH: - return "device suspended"; - case -EINVAL: - case -EAGAIN: - case -EFBIG: - case -EMSGSIZE: - return "internal error"; - default: - return "unknown error"; - } -} - -/* - * set up and start data/sync urbs - */ -static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) -{ - unsigned int i; - int err; - - if (subs->stream->chip->shutdown) - return -EBADFD; - - for (i = 0; i < subs->nurbs; i++) { - if (snd_BUG_ON(!subs->dataurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); - goto __error; - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (snd_BUG_ON(!subs->syncurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); - goto __error; - } - } - } - - subs->active_mask = 0; - subs->unlink_mask = 0; - subs->running = 1; - for (i = 0; i < subs->nurbs; i++) { - err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit datapipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i, &subs->active_mask); - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit syncpipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i + 16, &subs->active_mask); - } - } - return 0; - - __error: - // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - deactivate_urbs(subs, 0, 0); - return -EPIPE; -} - - -/* - */ -static struct snd_urb_ops audio_urb_ops[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb, - .retire_sync = retire_playback_sync_urb, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb, - .retire_sync = retire_capture_sync_urb, - }, -}; - -/* - * initialize the substream instance. - */ - -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, struct audioformat *fp) -{ - struct snd_usb_substream *subs = &as->substream[stream]; - - INIT_LIST_HEAD(&subs->fmt_list); - spin_lock_init(&subs->lock); - - subs->stream = as; - subs->direction = stream; - subs->dev = as->chip->dev; - subs->txfr_quirk = as->chip->txfr_quirk; - subs->ops = audio_urb_ops[stream]; - if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) - subs->ops.prepare_sync = prepare_capture_sync_urb_hs; - - snd_usb_set_pcm_ops(as->pcm, stream); - - list_add_tail(&fp->list, &subs->fmt_list); - subs->formats |= fp->formats; - subs->endpoint = fp->endpoint; - subs->num_formats++; - subs->fmt_type = fp->fmt_type; -} - -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.prepare = prepare_playback_urb; - return 0; - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.prepare = prepare_nodata_playback_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - subs->ops.retire = retire_capture_urb; - return start_urbs(subs, substream->runtime); - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.retire = retire_paused_capture_urb; - return 0; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.retire = retire_capture_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime) -{ - /* clear urbs (to be sure) */ - deactivate_urbs(subs, 0, 1); - wait_clear_urbs(subs); - - /* for playback, submit the URBs now; otherwise, the first hwptr_done - * updates for all URBs would happen at the same time when starting */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { - subs->ops.prepare = prepare_nodata_playback_urb; - return start_urbs(subs, runtime); - } - - return 0; -} - diff --git a/sound/usb/urb.h b/sound/usb/urb.h deleted file mode 100644 index 888da38079cf3f..00000000000000 --- a/sound/usb/urb.h +++ /dev/null @@ -1,21 +0,0 @@ -#ifndef __USBAUDIO_URB_H -#define __USBAUDIO_URB_H - -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp); - -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits); - -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); - -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime); - -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); - -#endif /* __USBAUDIO_URB_H */ From 14515a08294644b0ca557b440b8ddde9e7d65ede Mon Sep 17 00:00:00 2001 From: Daniele Guerrieri Date: Fri, 16 Sep 2011 08:31:45 +0200 Subject: [PATCH 25/45] ALSA: usb-audio: Added support for Roland UM-ONE midi-usb interface Roland UM-ONE midi usb interface differs from Roland UM-1. Signed-off-by: Daniele Guerrieri Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index da898229bb1192..b61945f3af9e59 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1688,6 +1688,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Added support for Roland UM-ONE which differs from UM-1 */ + USB_DEVICE(0x0582, 0x012a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "ROLAND", */ + /* .product_name = "UM-ONE", */ + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0003 + } + } +}, { USB_DEVICE(0x0582, 0x011e), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { From 84f9df159df6311f33ab16637772788cf3729ede Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 16 Sep 2011 22:52:48 +0200 Subject: [PATCH 26/45] ALSA: ymfpci: fix PCM open error handling The installation of the minimum period size constraint in the PCM open callbacks was not checked for errors. Add this check, and move the call to the beginning of the function to avoid having to do any cleanups in the error case. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/ymfpci/ymfpci_main.c | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index ebfbb28c35cc60..88c5c5c28d02a6 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -897,6 +897,15 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream) struct snd_ymfpci *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ymfpci_pcm *ypcm; + int err; + + runtime->hw = snd_ymfpci_playback; + /* FIXME? True value is 256/48 = 5.33333 ms */ + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 5334, UINT_MAX); + if (err < 0) + return err; ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL); if (ypcm == NULL) @@ -904,11 +913,8 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream) ypcm->chip = chip; ypcm->type = PLAYBACK_VOICE; ypcm->substream = substream; - runtime->hw = snd_ymfpci_playback; runtime->private_data = ypcm; runtime->private_free = snd_ymfpci_pcm_free_substream; - /* FIXME? True value is 256/48 = 5.33333 ms */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX); return 0; } @@ -1013,6 +1019,15 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream, struct snd_ymfpci *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ymfpci_pcm *ypcm; + int err; + + runtime->hw = snd_ymfpci_capture; + /* FIXME? True value is 256/48 = 5.33333 ms */ + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 5334, UINT_MAX); + if (err < 0) + return err; ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL); if (ypcm == NULL) @@ -1022,9 +1037,6 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream, ypcm->substream = substream; ypcm->capture_bank_number = capture_bank_number; chip->capture_substream[capture_bank_number] = substream; - runtime->hw = snd_ymfpci_capture; - /* FIXME? True value is 256/48 = 5.33333 ms */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX); runtime->private_data = ypcm; runtime->private_free = snd_ymfpci_pcm_free_substream; snd_ymfpci_hw_start(chip); From d5b702a64b4c273c8eed7e4e721364493d01fdc9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 16 Sep 2011 23:03:02 +0200 Subject: [PATCH 27/45] ALSA: pcm: add snd_pcm_hw_rule_noresample() Add a helper function to allow drivers to disable hardware resampling when the application has specified the SNDRV_PCM_HW_PARAMS_NORESAMPLE flag. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 ++ sound/core/pcm_lib.c | 26 ++++++++++++++++++++++++++ 2 files changed, 28 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 57e71fa33f7ce8..dc36f756fe8d55 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -825,6 +825,8 @@ int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var); +int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, + unsigned int base_rate); int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, int var, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 62e90b862a0ddd..95d1e789715f13 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1399,6 +1399,32 @@ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2); +static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + unsigned int base_rate = (unsigned int)(uintptr_t)rule->private; + struct snd_interval *rate; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + return snd_interval_list(rate, 1, &base_rate, 0); +} + +/** + * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling + * @runtime: PCM runtime instance + * @base_rate: the rate at which the hardware does not resample + */ +int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, + unsigned int base_rate) +{ + return snd_pcm_hw_rule_add(runtime, SNDRV_PCM_HW_PARAMS_NORESAMPLE, + SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_noresample_func, + (void *)(uintptr_t)base_rate, + SNDRV_PCM_HW_PARAM_RATE, -1); +} +EXPORT_SYMBOL(snd_pcm_hw_rule_noresample); + static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { From 5b0416a3c2f301e67d307ffc26ba43dff2d0d435 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 16 Sep 2011 23:08:28 +0200 Subject: [PATCH 28/45] ALSA: ymfpci: allow to disable the SRC Add the PCM rules to allow disabling the PCM playback and capture SRCs. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/ymfpci/ymfpci_main.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 88c5c5c28d02a6..66ea71b2a70d09 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -906,6 +906,9 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream) 5334, UINT_MAX); if (err < 0) return err; + err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (err < 0) + return err; ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL); if (ypcm == NULL) @@ -1028,6 +1031,9 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream, 5334, UINT_MAX); if (err < 0) return err; + err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (err < 0) + return err; ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL); if (ypcm == NULL) From 57e5c63007955838043e34c732d224b2cbbb128f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 16 Sep 2011 23:13:38 +0200 Subject: [PATCH 29/45] ALSA: emu10k1: allow to disable the SRC Add the PCM rule to allow disabling the PCM playback SRC. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emupcm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 622bace148e3c4..e22b8e2bbd8840 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1146,6 +1146,11 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream) kfree(epcm); return err; } + err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (err < 0) { + kfree(epcm); + return err; + } mix = &emu->pcm_mixer[substream->number]; for (i = 0; i < 4; i++) mix->send_routing[0][i] = mix->send_routing[1][i] = mix->send_routing[2][i] = i; From 5495ffbd7b56d8bffebc5e30f03ea374590f1bb4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 16 Sep 2011 23:16:05 +0200 Subject: [PATCH 30/45] ALSA: via82xx: allow to disable the SRC Add the PCM rule to allow disabling the PCM playback SRC. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 35d5f4313d99ba..c3656fffdb50d3 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1175,6 +1175,7 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, struct snd_pcm_runtime *runtime = substream->runtime; int err; struct via_rate_lock *ratep; + bool use_src = false; runtime->hw = snd_via82xx_hw; @@ -1196,6 +1197,7 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, SNDRV_PCM_RATE_8000_48000); runtime->hw.rate_min = 8000; runtime->hw.rate_max = 48000; + use_src = true; } else if (! ratep->rate) { int idx = viadev->direction ? AC97_RATES_ADC : AC97_RATES_FRONT_DAC; runtime->hw.rates = chip->ac97->rates[idx]; @@ -1212,6 +1214,12 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; + if (use_src) { + err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (err < 0) + return err; + } + runtime->private_data = viadev; viadev->substream = substream; From 8e699d2cc286506c00ce8ecc67c3d7d6cca9e814 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Sep 2011 16:54:23 +0200 Subject: [PATCH 31/45] ALSA: fm801 - Clean up redundant reference to snd_fm801_tea575x_gpios[] Use macro to improve readability. Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 76465f5d9f58c6..136f7232bb7cf0 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -729,11 +729,14 @@ static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = { { .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" }, }; +#define get_tea575x_gpio(chip) \ + (&snd_fm801_tea575x_gpios[((chip)->tea575x_tuner & TUNER_TYPE_MASK) - 1]) + static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct fm801 *chip = tea->private_data; unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip); reg &= ~(FM801_GPIO_GP(gpio.data) | FM801_GPIO_GP(gpio.clk) | @@ -751,7 +754,7 @@ static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea) { struct fm801 *chip = tea->private_data; unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip); return (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 | (reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0; @@ -761,7 +764,7 @@ static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output { struct fm801 *chip = tea->private_data; unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip); /* use GPIO lines and set write enable bit */ reg |= FM801_GPIO_GS(gpio.data) | @@ -1246,7 +1249,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, chip->tea575x_tuner = tea575x_tuner; if (!snd_tea575x_init(&chip->tea)) { snd_printk(KERN_INFO "detected TEA575x radio type %s\n", - snd_fm801_tea575x_gpios[tea575x_tuner - 1].name); + get_tea575x_gpio(chip)->name); break; } } @@ -1256,9 +1259,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, } } if (!(chip->tea575x_tuner & TUNER_DISABLED)) { - strlcpy(chip->tea.card, - snd_fm801_tea575x_gpios[(tea575x_tuner & - TUNER_TYPE_MASK) - 1].name, + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, sizeof(chip->tea.card)); } #endif From 643d6bbb9637a9b4bb47ec1a1ae3adf3ff9d75a1 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 23 Sep 2011 09:24:21 +0300 Subject: [PATCH 32/45] ALSA: hdspm - potential info leak in snd_hdspm_hwdep_ioctl() Smatch has a new check for Rosenberg type information leaks where structs are copied to the user with uninitialized stack data in them. The status struct has a hole in it, and on some paths not all the members were initialized. struct hdspm_status { unsigned char card_type; /* 0 1 */ /* XXX 3 bytes hole, try to pack */ enum hdspm_syncsource autosync_source; /* 4 4 */ long long unsigned int card_clock; /* 8 8 */ The hdspm_version struct had holes in it as well. struct hdspm_version { unsigned char card_type; /* 0 1 */ char cardname[20]; /* 1 20 */ /* XXX 3 bytes hole, try to pack */ unsigned int serial; /* 24 4 */ short unsigned int firmware_rev; /* 28 2 */ /* XXX 2 bytes hole, try to pack */ int addons; /* 32 4 */ Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 214110d6a2bf3b..bf438d121afe60 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6227,6 +6227,8 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, break; case SNDRV_HDSPM_IOCTL_GET_STATUS: + memset(&status, 0, sizeof(status)); + status.card_type = hdspm->io_type; status.autosync_source = hdspm_autosync_ref(hdspm); @@ -6266,6 +6268,8 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, break; case SNDRV_HDSPM_IOCTL_GET_VERSION: + memset(&hdspm_version, 0, sizeof(hdspm_version)); + hdspm_version.card_type = hdspm->io_type; strncpy(hdspm_version.cardname, hdspm->card_name, sizeof(hdspm_version.cardname)); From 2ca595ab7a557f6cee21bf073fe2a242004cd19e Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 23 Sep 2011 09:25:05 +0300 Subject: [PATCH 33/45] ALSA: hdspm - cleanup __user tags in ioctl() This makes the code cleaner and silences a Sparse complaint: sound/pci/rme9652/hdspm.c:6341:23: warning: incorrect type in assignment (incompatible argument 4 (different address spaces)) sound/pci/rme9652/hdspm.c:6341:23: expected int ( *ioctl )( ... ) sound/pci/rme9652/hdspm.c:6341:23: got int ( static [toplevel] * )( ... ) sound/pci/rme9652/hdspm.c:6102:44: warning: dereference of noderef expression sound/pci/rme9652/hdspm.c:6225:50: warning: dereference of noderef expression sound/pci/rme9652/hdspm.c:6264:50: warning: dereference of noderef expression sound/pci/rme9652/hdspm.c:6283:50: warning: dereference of noderef expression sound/pci/rme9652/hdspm.c:6289:59: warning: dereference of noderef expression Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index bf438d121afe60..6e2f7ef7ddb155 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6097,7 +6097,7 @@ static inline int copy_u32_le(void __user *dest, void __iomem *src) } static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, - unsigned int cmd, unsigned long __user arg) + unsigned int cmd, unsigned long arg) { void __user *argp = (void __user *)arg; struct hdspm *hdspm = hw->private_data; @@ -6222,7 +6222,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, info.line_out = hdspm_line_out(hdspm); info.passthru = 0; spin_unlock_irq(&hdspm->lock); - if (copy_to_user((void __user *) arg, &info, sizeof(info))) + if (copy_to_user(argp, &info, sizeof(info))) return -EFAULT; break; @@ -6261,7 +6261,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, break; } - if (copy_to_user((void __user *) arg, &status, sizeof(status))) + if (copy_to_user(argp, &status, sizeof(status))) return -EFAULT; @@ -6280,13 +6280,13 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, if (hdspm->tco) hdspm_version.addons |= HDSPM_ADDON_TCO; - if (copy_to_user((void __user *) arg, &hdspm_version, + if (copy_to_user(argp, &hdspm_version, sizeof(hdspm_version))) return -EFAULT; break; case SNDRV_HDSPM_IOCTL_GET_MIXER: - if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer))) + if (copy_from_user(&mixer, argp, sizeof(mixer))) return -EFAULT; if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer, sizeof(struct hdspm_mixer))) From 49957f39665d50343e04effc65c78919364228ce Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Fri, 23 Sep 2011 14:32:11 +0300 Subject: [PATCH 34/45] ALSA: 6fire: don't use custom hex_to_bin() Signed-off-by: Andy Shevchenko Signed-off-by: Takashi Iwai --- sound/usb/6fire/firmware.c | 25 ++++++++++++------------- 1 file changed, 12 insertions(+), 13 deletions(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 1e3ae3327dd3a6..07bcfe4d18a7a9 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -16,6 +16,7 @@ #include #include +#include #include "firmware.h" #include "chip.h" @@ -59,21 +60,19 @@ struct ihex_record { unsigned int txt_offset; /* current position in txt_data */ }; -static u8 usb6fire_fw_ihex_nibble(const u8 n) -{ - if (n >= '0' && n <= '9') - return n - '0'; - else if (n >= 'A' && n <= 'F') - return n - ('A' - 10); - else if (n >= 'a' && n <= 'f') - return n - ('a' - 10); - return 0; -} - static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc) { - u8 val = (usb6fire_fw_ihex_nibble(data[0]) << 4) | - usb6fire_fw_ihex_nibble(data[1]); + u8 val = 0; + int hval; + + hval = hex_to_bin(data[0]); + if (hval >= 0) + val |= (hval << 4); + + hval = hex_to_bin(data[1]); + if (hval >= 0) + val |= hval; + *crc += val; return val; } From 6b69a0e520a0dc6579901098d0810bcd2e1ea60b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 24 Sep 2011 12:16:29 +0200 Subject: [PATCH 35/45] ALSA: aloop - Use vmalloc buffer snd-aloop driver is virtual and has no need for allocating contiguous pages. It'll be more system-friendly to use vmalloc buffers. Tested-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a0da7755fceaad..4067f1548949b8 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -575,7 +575,8 @@ static void loopback_runtime_free(struct snd_pcm_runtime *runtime) static int loopback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(params)); } static int loopback_hw_free(struct snd_pcm_substream *substream) @@ -587,7 +588,7 @@ static int loopback_hw_free(struct snd_pcm_substream *substream) mutex_lock(&dpcm->loopback->cable_lock); cable->valid &= ~(1 << substream->stream); mutex_unlock(&dpcm->loopback->cable_lock); - return snd_pcm_lib_free_pages(substream); + return snd_pcm_lib_free_vmalloc_buffer(substream); } static unsigned int get_cable_index(struct snd_pcm_substream *substream) @@ -740,6 +741,8 @@ static struct snd_pcm_ops loopback_playback_ops = { .prepare = loopback_prepare, .trigger = loopback_trigger, .pointer = loopback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops loopback_capture_ops = { @@ -751,6 +754,8 @@ static struct snd_pcm_ops loopback_capture_ops = { .prepare = loopback_prepare, .trigger = loopback_trigger, .pointer = loopback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static int __devinit loopback_pcm_new(struct loopback *loopback, @@ -771,10 +776,6 @@ static int __devinit loopback_pcm_new(struct loopback *loopback, strcpy(pcm->name, "Loopback PCM"); loopback->pcm[device] = pcm; - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - 0, 2 * 1024 * 1024); return 0; } From 17d900c4a1b50bc191b3ca58cbd78acc04a1c5b3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 26 Sep 2011 21:15:27 +0200 Subject: [PATCH 36/45] ALSA: usb-audio: increase control transfer timeout There are certain devices that are reportedly so slow that they need more than 100 ms to handle control transfers. Therefore, increase the timeout in mixer(_quirks).c to 1000 ms. The timeout parameter of snd_usb_ctl_msg() is now constant, so we can drop it. Reported-by: Felipe Balbi Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 12 ++++++------ sound/usb/format.c | 4 ++-- sound/usb/helper.c | 4 ++-- sound/usb/helper.h | 2 +- sound/usb/mixer.c | 6 +++--- sound/usb/mixer_quirks.c | 10 +++++----- sound/usb/pcm.c | 4 ++-- sound/usb/quirks.c | 8 ++++---- 8 files changed, 25 insertions(+), 25 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 075195e8661a0f..379baad3d5ad54 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -91,7 +91,7 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, UAC2_CX_CLOCK_SELECTOR << 8, snd_usb_ctrl_intf(chip) | (selector_id << 8), - &buf, sizeof(buf), 1000); + &buf, sizeof(buf)); if (ret < 0) return ret; @@ -118,7 +118,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) 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), 1000); + &data, sizeof(data)); if (err < 0) { snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n", @@ -222,7 +222,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; @@ -231,7 +231,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ @@ -273,7 +273,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, UAC2_CS_CONTROL_SAM_FREQ << 8, snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", dev->devnum, iface, fmt->altsetting, rate); return err; @@ -283,7 +283,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", dev->devnum, iface, fmt->altsetting); return err; diff --git a/sound/usb/format.c b/sound/usb/format.c index 8d042dce0d16a0..89421d17657073 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -286,7 +286,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, snd_usb_ctrl_intf(chip) | (clock << 8), - tmp, sizeof(tmp), 1000); + tmp, sizeof(tmp)); if (ret < 0) { snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n", @@ -307,7 +307,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, snd_usb_ctrl_intf(chip) | (clock << 8), - data, data_size, 1000); + data, data_size); if (ret < 0) { snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n", diff --git a/sound/usb/helper.c b/sound/usb/helper.c index f280c1903c25bd..9eed8f40b179dd 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -81,7 +81,7 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype */ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, - __u16 size, int timeout) + __u16 size) { int err; void *buf = NULL; @@ -92,7 +92,7 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, return -ENOMEM; } err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, timeout); + value, index, buf, size, 1000); if (size > 0) { memcpy(data, buf, size); kfree(buf); diff --git a/sound/usb/helper.h b/sound/usb/helper.h index 09bd943c43bf73..805c300dd004df 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -8,7 +8,7 @@ void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsub int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, - void *data, __u16 size, int timeout); + void *data, __u16 size); unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, struct usb_host_interface *alts); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 78a5abda6793dc..b13b7ac5bad919 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -296,7 +296,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, val_len, 100) >= val_len) { + buf, val_len) >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); snd_usb_autosuspend(cval->mixer->chip); return 0; @@ -333,7 +333,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, size, 1000); + buf, size); snd_usb_autosuspend(chip); if (ret < 0) { @@ -445,7 +445,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, val_len, 100) >= 0) { + buf, val_len) >= 0) { snd_usb_autosuspend(chip); return 0; } diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 3d0f4873112b17..ab125ee0b0f0e2 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -190,18 +190,18 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - !value, 0, NULL, 0, 100); + !value, 0, NULL, 0); /* USB X-Fi S51 Pro */ if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - !value, 0, NULL, 0, 100); + !value, 0, NULL, 0); else err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - value, index + 2, NULL, 0, 100); + value, index + 2, NULL, 0); if (err < 0) return err; mixer->audigy2nx_leds[index] = value; @@ -299,7 +299,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, usb_rcvctrlpipe(mixer->chip->dev, 0), UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE, 0, - jacks[i].unitid << 8, buf, 3, 100); + jacks[i].unitid << 8, buf, 3); if (err == 3 && (buf[0] == 3 || buf[0] == 6)) snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); else @@ -332,7 +332,7 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 50, 0, &new_status, 1, 100); + 50, 0, &new_status, 1); if (err < 0) return err; mixer->xonar_u1_status = new_status; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b5bc870878dbd7..0220b0f335b9ad 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -152,7 +152,7 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n", dev->devnum, iface, ep); return err; @@ -176,7 +176,7 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, UAC2_EP_CS_PITCH << 8, 0, - data, sizeof(data), 1000)) < 0) { + data, sizeof(data))) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH (v2)\n", dev->devnum, iface, fmt->altsetting); return err; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 556edea28b9047..2e5bc73440262a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -340,7 +340,7 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac snd_printdd("sending Extigy boot sequence...\n"); /* Send message to force it to reconnect with full interface. */ err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0), - 0x10, 0x43, 0x0001, 0x000a, NULL, 0, 1000); + 0x10, 0x43, 0x0001, 0x000a, NULL, 0); if (err < 0) snd_printdd("error sending boot message: %d\n", err); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); @@ -361,11 +361,11 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a, USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 0, 0, &buf, 1, 1000); + 0, 0, &buf, 1); if (buf == 0) { snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x29, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 1, 2000, NULL, 0, 1000); + 1, 2000, NULL, 0); return -ENODEV; } return 0; @@ -408,7 +408,7 @@ static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 valu buf[3] = reg; return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, - 0, 0, &buf, 4, 1000); + 0, 0, &buf, 4); } static int snd_usb_cm106_boot_quirk(struct usb_device *dev) From bb690c9e2702e49af3be2fb6f03d9b60e0afcab7 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 29 Sep 2011 09:10:48 +0300 Subject: [PATCH 37/45] sound: oss: use strlcpy() in sound_timer_init() sound_timer.info.name is a 32 character buffer. This function only has one caller (in sound/oss/ad1848.c) and it passes as 128 character buffer as "name". I don't know if this is a problem in real life, and I doubt we're going to add more OSS drivers so it's unlikely to become an issue. But we may as well take care of it. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sound_timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index 48cda6c4c257b3..8021c85f076d09 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -320,7 +320,7 @@ void sound_timer_init(struct sound_lowlev_timer *t, char *name) n = sound_alloc_timerdev(); if (n == -1) n = 0; /* Overwrite the system timer */ - strcpy(sound_timer.info.name, name); + strlcpy(sound_timer.info.name, name, sizeof(sound_timer.info.name)); sound_timer_devs[n] = &sound_timer; } EXPORT_SYMBOL(sound_timer_init); From a0978e8039f1b1bfb9fbc68f682b14313bb4f9ad Mon Sep 17 00:00:00 2001 From: Stefan Richter Date: Sat, 27 Aug 2011 16:45:28 +0200 Subject: [PATCH 38/45] ALSA: firewire-speakers: fix locking There is a lock inversion between fwspk->mutex and pcm->open_mutex reported by lockdep when fwspk_hw_free is called. Fixed by copying the fix from the same former issue in the isight sound driver (commit f3f7c1837f6bcae3601fc535b339426868bf1549 "ALSA: isight: fix locking"). Signed-off-by: Stefan Richter Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/speakers.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 3fc257da180ce0..cbe6bb9e53b6fa 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -778,9 +778,10 @@ static int __devexit fwspk_remove(struct device *dev) { struct fwspk *fwspk = dev_get_drvdata(dev); - mutex_lock(&fwspk->mutex); amdtp_out_stream_pcm_abort(&fwspk->stream); snd_card_disconnect(fwspk->card); + + mutex_lock(&fwspk->mutex); fwspk_stop_stream(fwspk); mutex_unlock(&fwspk->mutex); @@ -796,8 +797,8 @@ static void fwspk_bus_reset(struct fw_unit *unit) fcp_bus_reset(fwspk->unit); if (cmp_connection_update(&fwspk->connection) < 0) { - mutex_lock(&fwspk->mutex); amdtp_out_stream_pcm_abort(&fwspk->stream); + mutex_lock(&fwspk->mutex); fwspk_stop_stream(fwspk); mutex_unlock(&fwspk->mutex); return; From f92766bc8948f978a838a5607bea95804c8dfdfe Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 4 Oct 2011 09:29:39 +0300 Subject: [PATCH 39/45] ALSA: oss-mixer - use strlcpy() instead strcpy() This is mostly a static checker fix more than anything else. We're copying from a 64 char buffer into a 44 char buffer. The 64 character buffer is str[] in snd_mixer_oss_build_test_all(). The call tree is: snd_mixer_oss_build_test_all() -> snd_mixer_oss_build_test() -> snd_mixer_oss_build_test(). We never actually do fill str[] buffer all the way to 64 characters. The longest string is: sprintf(str, "%s Playback Switch", ptr->name); ptr->name is a 32 character buffer so 32 plus 16 characters for " Playback Switch" still puts us over the 44 limit from "id.name". Most likely ptr->name never gets filled to the limit, but we can't really change the size of that buffer so lets just use strlcpy() here and be safe. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index d8359cfeca15a4..1b5e0c49a0adde 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -499,7 +499,7 @@ static struct snd_kcontrol *snd_mixer_oss_test_id(struct snd_mixer_oss *mixer, c memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, name); + strlcpy(id.name, name, sizeof(id.name)); id.index = index; return snd_ctl_find_id(card, &id); } From 8d448162bda5ae3b5ecb26fe50c8fbbeae99faa4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 7 Oct 2011 22:38:59 +0200 Subject: [PATCH 40/45] ALSA: control: add support for ENUMERATED user space controls Handling of user control elements was implemented for all types except ENUMERATED. This type will be needed for the device-specific mixers of upcoming FireWire drivers. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/asound.h | 4 +- sound/core/control.c | 80 +++++++++++++++++++++++++++++++++++-- sound/core/control_compat.c | 4 ++ 3 files changed, 83 insertions(+), 5 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 5d6074faa279ad..a2e4ff5ba9e9f1 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -706,7 +706,7 @@ struct snd_timer_tread { * * ****************************************************************************/ -#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6) +#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 7) struct snd_ctl_card_info { int card; /* card number */ @@ -803,6 +803,8 @@ struct snd_ctl_elem_info { unsigned int items; /* R: number of items */ unsigned int item; /* W: item number */ char name[64]; /* R: value name */ + __u64 names_ptr; /* W: names list (ELEM_ADD only) */ + unsigned int names_length; } enumerated; unsigned char reserved[128]; } value; diff --git a/sound/core/control.c b/sound/core/control.c index dc2a44048c850e..978fe1a8e9f087 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -989,7 +989,6 @@ struct user_element { void *tlv_data; /* TLV data */ unsigned long tlv_data_size; /* TLV data size */ void *priv_data; /* private data (like strings for enumerated type) */ - unsigned long priv_data_size; /* size of private data in bytes */ }; static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol, @@ -1001,6 +1000,28 @@ static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol, return 0; } +static int snd_ctl_elem_user_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct user_element *ue = kcontrol->private_data; + const char *names; + unsigned int item; + + item = uinfo->value.enumerated.item; + + *uinfo = ue->info; + + item = min(item, uinfo->value.enumerated.items - 1); + uinfo->value.enumerated.item = item; + + names = ue->priv_data; + for (; item > 0; --item) + names += strlen(names) + 1; + strcpy(uinfo->value.enumerated.name, names); + + return 0; +} + static int snd_ctl_elem_user_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1055,11 +1076,46 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol, return change; } +static int snd_ctl_elem_init_enum_names(struct user_element *ue) +{ + char *names, *p; + size_t buf_len, name_len; + unsigned int i; + + if (ue->info.value.enumerated.names_length > 64 * 1024) + return -EINVAL; + + names = memdup_user( + (const void __user *)ue->info.value.enumerated.names_ptr, + ue->info.value.enumerated.names_length); + if (IS_ERR(names)) + return PTR_ERR(names); + + /* check that there are enough valid names */ + buf_len = ue->info.value.enumerated.names_length; + p = names; + for (i = 0; i < ue->info.value.enumerated.items; ++i) { + name_len = strnlen(p, buf_len); + if (name_len == 0 || name_len >= 64 || name_len == buf_len) { + kfree(names); + return -EINVAL; + } + p += name_len + 1; + buf_len -= name_len + 1; + } + + ue->priv_data = names; + ue->info.value.enumerated.names_ptr = 0; + + return 0; +} + static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol) { struct user_element *ue = kcontrol->private_data; - if (ue->tlv_data) - kfree(ue->tlv_data); + + kfree(ue->tlv_data); + kfree(ue->priv_data); kfree(ue); } @@ -1101,7 +1157,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, memcpy(&kctl.id, &info->id, sizeof(info->id)); kctl.count = info->owner ? info->owner : 1; access |= SNDRV_CTL_ELEM_ACCESS_USER; - kctl.info = snd_ctl_elem_user_info; + if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) + kctl.info = snd_ctl_elem_user_enum_info; + else + kctl.info = snd_ctl_elem_user_info; if (access & SNDRV_CTL_ELEM_ACCESS_READ) kctl.get = snd_ctl_elem_user_get; if (access & SNDRV_CTL_ELEM_ACCESS_WRITE) @@ -1122,6 +1181,11 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count > 64) return -EINVAL; break; + case SNDRV_CTL_ELEM_TYPE_ENUMERATED: + private_size = sizeof(unsigned int); + if (info->count > 128 || info->value.enumerated.items == 0) + return -EINVAL; + break; case SNDRV_CTL_ELEM_TYPE_BYTES: private_size = sizeof(unsigned char); if (info->count > 512) @@ -1143,9 +1207,17 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, ue->info.access = 0; ue->elem_data = (char *)ue + sizeof(*ue); ue->elem_data_size = private_size; + if (ue->info.type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) { + err = snd_ctl_elem_init_enum_names(ue); + if (err < 0) { + kfree(ue); + return err; + } + } kctl.private_free = snd_ctl_elem_user_free; _kctl = snd_ctl_new(&kctl, access); if (_kctl == NULL) { + kfree(ue->priv_data); kfree(ue); return -ENOMEM; } diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 426874429a5e08..2bb95a7a8809fe 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -83,6 +83,8 @@ struct snd_ctl_elem_info32 { u32 items; u32 item; char name[64]; + u64 names_ptr; + u32 names_length; } enumerated; unsigned char reserved[128]; } value; @@ -372,6 +374,8 @@ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, &data32->value.enumerated, sizeof(data->value.enumerated))) goto error; + data->value.enumerated.names_ptr = + (uintptr_t)compat_ptr(data->value.enumerated.names_ptr); break; default: break; From ffd3d5c6c7a20fb718daf98a6c8a476d228f3995 Mon Sep 17 00:00:00 2001 From: Feng Tang Date: Mon, 10 Oct 2011 10:31:48 +0800 Subject: [PATCH 41/45] ALSA: pcm - remove the dead code from snd_pcm_open_file() The rpcm_file parameter is never used in current ALSA code, so remove it to make it cleaner. Signed-off-by: Feng Tang Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1c6be91dfb9887..2d3af5df88d644 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2058,16 +2058,12 @@ EXPORT_SYMBOL(snd_pcm_open_substream); static int snd_pcm_open_file(struct file *file, struct snd_pcm *pcm, - int stream, - struct snd_pcm_file **rpcm_file) + int stream) { struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; int err; - if (rpcm_file) - *rpcm_file = NULL; - err = snd_pcm_open_substream(pcm, stream, file, &substream); if (err < 0) return err; @@ -2083,8 +2079,7 @@ static int snd_pcm_open_file(struct file *file, substream->pcm_release = pcm_release_private; } file->private_data = pcm_file; - if (rpcm_file) - *rpcm_file = pcm_file; + return 0; } @@ -2113,7 +2108,6 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) { int err; - struct snd_pcm_file *pcm_file; wait_queue_t wait; if (pcm == NULL) { @@ -2131,7 +2125,7 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) add_wait_queue(&pcm->open_wait, &wait); mutex_lock(&pcm->open_mutex); while (1) { - err = snd_pcm_open_file(file, pcm, stream, &pcm_file); + err = snd_pcm_open_file(file, pcm, stream); if (err >= 0) break; if (err == -EAGAIN) { From 3d37fbe44112b06279efa04ad91a0e4b7a0c600c Mon Sep 17 00:00:00 2001 From: William Light Date: Mon, 10 Oct 2011 15:54:22 +0000 Subject: [PATCH 42/45] ALSA: snd-usb-caiaq: Fix NULL dereference in input.c There was a case where a newly-registered input device could be opened before a necessary variable in the device structure was set. When code tried to use the variable in the URB reply callback, it would cause an Oops. This fix sets the aforementioned variable before calling input_register_device. Signed-off-by: William Light Signed-off-by: Takashi Iwai --- sound/usb/caiaq/input.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index a213813487bd77..9efb92e4090dba 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -664,15 +664,17 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) for (i = 0; i < input->keycodemax; i++) __set_bit(dev->keycode[i], input->keybit); + dev->input_dev = input; + ret = input_register_device(input); if (ret < 0) goto exit_free_idev; - dev->input_dev = input; return 0; exit_free_idev: input_free_device(input); + dev->input_dev = NULL; return ret; } From e653510a27e63b41a5bae3c46eb093375e17ca2d Mon Sep 17 00:00:00 2001 From: William Light Date: Mon, 10 Oct 2011 15:54:23 +0000 Subject: [PATCH 43/45] ALSA: snd-usb-caiaq: Add support for Maschine This adds partial support for the Maschine controller by Native Instruments. Supported now are the 1x1 MIDI interface and the 41 buttons, 11 endless rotary encoders, and 16 pressure-sensitive drum pads. Still to work on are the dimmable LEDs and the two monochrome screens. Signed-off-by: William Light Signed-off-by: Takashi Iwai --- sound/usb/Kconfig | 2 + sound/usb/caiaq/device.c | 8 ++- sound/usb/caiaq/device.h | 1 + sound/usb/caiaq/input.c | 151 ++++++++++++++++++++++++++++++++++++++- 4 files changed, 160 insertions(+), 2 deletions(-) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 8beb77563da252..3efc21c3d67c61 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -67,6 +67,7 @@ config SND_USB_CAIAQ * Native Instruments Guitar Rig mobile * Native Instruments Traktor Kontrol X1 * Native Instruments Traktor Kontrol S4 + * Native Instruments Maschine Controller To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. @@ -85,6 +86,7 @@ config SND_USB_CAIAQ_INPUT * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 * Native Instruments Traktor Kontrol S4 + * Native Instruments Maschine Controller config SND_USB_US122L tristate "Tascam US-122L USB driver" diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 45bc4a2dc6f0dc..3eb605bd95038f 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -50,7 +50,8 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, Session I/O}," "{Native Instruments, GuitarRig mobile}" "{Native Instruments, Traktor Kontrol X1}" - "{Native Instruments, Traktor Kontrol S4}"); + "{Native Instruments, Traktor Kontrol S4}" + "{Native Instruments, Maschine Controller}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -146,6 +147,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_TRAKTORAUDIO2 }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_MASCHINECONTROLLER + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 3f9c6339ae90fa..562b0bff9c417d 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -18,6 +18,7 @@ #define USB_PID_TRAKTORKONTROLX1 0x2305 #define USB_PID_TRAKTORKONTROLS4 0xbaff #define USB_PID_TRAKTORAUDIO2 0x041d +#define USB_PID_MASCHINECONTROLLER 0x0808 #define EP1_BUFSIZE 64 #define EP4_BUFSIZE 512 diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 9efb92e4090dba..26a121b42c3c15 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -67,6 +67,61 @@ static unsigned short keycode_kore[] = { KEY_BRL_DOT5 }; +#define MASCHINE_BUTTONS (42) +#define MASCHINE_BUTTON(X) ((X) + BTN_MISC) +#define MASCHINE_PADS (16) +#define MASCHINE_PAD(X) ((X) + ABS_PRESSURE) + +static unsigned short keycode_maschine[] = { + MASCHINE_BUTTON(40), /* mute */ + MASCHINE_BUTTON(39), /* solo */ + MASCHINE_BUTTON(38), /* select */ + MASCHINE_BUTTON(37), /* duplicate */ + MASCHINE_BUTTON(36), /* navigate */ + MASCHINE_BUTTON(35), /* pad mode */ + MASCHINE_BUTTON(34), /* pattern */ + MASCHINE_BUTTON(33), /* scene */ + KEY_RESERVED, /* spacer */ + + MASCHINE_BUTTON(30), /* rec */ + MASCHINE_BUTTON(31), /* erase */ + MASCHINE_BUTTON(32), /* shift */ + MASCHINE_BUTTON(28), /* grid */ + MASCHINE_BUTTON(27), /* > */ + MASCHINE_BUTTON(26), /* < */ + MASCHINE_BUTTON(25), /* restart */ + + MASCHINE_BUTTON(21), /* E */ + MASCHINE_BUTTON(22), /* F */ + MASCHINE_BUTTON(23), /* G */ + MASCHINE_BUTTON(24), /* H */ + MASCHINE_BUTTON(20), /* D */ + MASCHINE_BUTTON(19), /* C */ + MASCHINE_BUTTON(18), /* B */ + MASCHINE_BUTTON(17), /* A */ + + MASCHINE_BUTTON(0), /* control */ + MASCHINE_BUTTON(2), /* browse */ + MASCHINE_BUTTON(4), /* < */ + MASCHINE_BUTTON(6), /* snap */ + MASCHINE_BUTTON(7), /* autowrite */ + MASCHINE_BUTTON(5), /* > */ + MASCHINE_BUTTON(3), /* sampling */ + MASCHINE_BUTTON(1), /* step */ + + MASCHINE_BUTTON(15), /* 8 softkeys */ + MASCHINE_BUTTON(14), + MASCHINE_BUTTON(13), + MASCHINE_BUTTON(12), + MASCHINE_BUTTON(11), + MASCHINE_BUTTON(10), + MASCHINE_BUTTON(9), + MASCHINE_BUTTON(8), + + MASCHINE_BUTTON(16), /* note repeat */ + MASCHINE_BUTTON(29) /* play */ +}; + #define KONTROLX1_INPUTS (40) #define KONTROLS4_BUTTONS (12 * 8) #define KONTROLS4_AXIS (46) @@ -218,6 +273,29 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, input_report_abs(input_dev, ABS_HAT3Y, i); input_sync(input_dev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + /* 4 under the left screen */ + input_report_abs(input_dev, ABS_HAT0X, decode_erp(buf[21], buf[20])); + input_report_abs(input_dev, ABS_HAT0Y, decode_erp(buf[15], buf[14])); + input_report_abs(input_dev, ABS_HAT1X, decode_erp(buf[9], buf[8])); + input_report_abs(input_dev, ABS_HAT1Y, decode_erp(buf[3], buf[2])); + + /* 4 under the right screen */ + input_report_abs(input_dev, ABS_HAT2X, decode_erp(buf[19], buf[18])); + input_report_abs(input_dev, ABS_HAT2Y, decode_erp(buf[13], buf[12])); + input_report_abs(input_dev, ABS_HAT3X, decode_erp(buf[7], buf[6])); + input_report_abs(input_dev, ABS_HAT3Y, decode_erp(buf[1], buf[0])); + + /* volume */ + input_report_abs(input_dev, ABS_RX, decode_erp(buf[17], buf[16])); + /* tempo */ + input_report_abs(input_dev, ABS_RY, decode_erp(buf[11], buf[10])); + /* swing */ + input_report_abs(input_dev, ABS_RZ, decode_erp(buf[5], buf[4])); + + input_sync(input_dev); + break; } } @@ -400,6 +478,25 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev, input_sync(dev->input_dev); } +#define MASCHINE_MSGBLOCK_SIZE 2 + +static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *dev, + const unsigned char *buf, + unsigned int len) +{ + unsigned int i, pad_id; + uint16_t pressure; + + for (i = 0; i < MASCHINE_PADS; i++) { + pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); + pad_id = pressure >> 12; + + input_report_abs(dev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + } + + input_sync(dev->input_dev); +} + static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) { struct snd_usb_caiaqdev *dev = urb->context; @@ -425,6 +522,13 @@ static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): snd_usb_caiaq_tks4_dispatch(dev, buf, urb->actual_length); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + if (urb->actual_length < (MASCHINE_PADS * MASCHINE_MSGBLOCK_SIZE)) + goto requeue; + + snd_usb_caiaq_maschine_dispatch(dev, buf, urb->actual_length); + break; } requeue: @@ -444,6 +548,7 @@ static int snd_usb_caiaq_input_open(struct input_dev *idev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0) return -EIO; break; @@ -462,6 +567,7 @@ static void snd_usb_caiaq_input_close(struct input_dev *idev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): usb_kill_urb(dev->ep4_in_urb); break; } @@ -652,6 +758,50 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); + input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) | + BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) | + BIT_MASK(ABS_HAT2X) | BIT_MASK(ABS_HAT2Y) | + BIT_MASK(ABS_HAT3X) | BIT_MASK(ABS_HAT3Y) | + BIT_MASK(ABS_RX) | BIT_MASK(ABS_RY) | + BIT_MASK(ABS_RZ); + + BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_maschine)); + memcpy(dev->keycode, keycode_maschine, sizeof(keycode_maschine)); + input->keycodemax = ARRAY_SIZE(keycode_maschine); + + for (i = 0; i < MASCHINE_PADS; i++) { + input->absbit[0] |= MASCHINE_PAD(i); + input_set_abs_params(input, MASCHINE_PAD(i), 0, 0xfff, 5, 10); + } + + input_set_abs_params(input, ABS_HAT0X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT0Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT1X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT1Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT2X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT2Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT3X, 0, 999, 0, 10); + input_set_abs_params(input, ABS_HAT3Y, 0, 999, 0, 10); + input_set_abs_params(input, ABS_RX, 0, 999, 0, 10); + input_set_abs_params(input, ABS_RY, 0, 999, 0, 10); + input_set_abs_params(input, ABS_RZ, 0, 999, 0, 10); + + dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!dev->ep4_in_urb) { + ret = -ENOMEM; + goto exit_free_idev; + } + + usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_rcvbulkpipe(usb_dev, 0x4), + dev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, dev); + + snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + break; + default: /* no input methods supported on this device */ goto exit_free_idev; @@ -690,4 +840,3 @@ void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev) input_unregister_device(dev->input_dev); dev->input_dev = NULL; } - From d09c06c6fc240261dde65198774b279d89c35459 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Oct 2011 08:19:09 +0200 Subject: [PATCH 44/45] ALSA: usb-audio - Fix possible access over audio_feature_info[] array The audio_feature_info[] array should contain all entries for UAC2_FU_*, but currently a few last entries are missing. Even though, the driver tries to probe these entries in parse_audio_feature_unit() and may access the range over the array. This patch fixes the bug by limiting the loop size properly using ARRAY_SIZE() instead of a hard-coded magic number. Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b13b7ac5bad919..60f65ace7474dd 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1259,7 +1259,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0); } } else { /* UAC_VERSION_2 */ - for (i = 0; i < 30/2; i++) { + for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) { unsigned int ch_bits = 0; unsigned int ch_read_only = 0; From dde7ad8dee274763c8958769779aea8c993c950e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Oct 2011 10:00:22 +0200 Subject: [PATCH 45/45] ALSA: Update the sound git tree URL Now back to kernel.org but without -2.6 suffix. Signed-off-by: Takashi Iwai --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 0b4ccdd35bbb18..a676343a27bc4e 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5990,7 +5990,7 @@ M: Jaroslav Kysela M: Takashi Iwai L: alsa-devel@alsa-project.org (moderated for non-subscribers) W: http://www.alsa-project.org/ -T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git T: git git://git.alsa-project.org/alsa-kernel.git S: Maintained F: Documentation/sound/