Merge branch 'for-next' into for-linus

Documentation/sound/alsa/HD-Audio.txt

@@ -655,17 +655,6 @@ development branches in general while the development for the current
 and next kernels are found in for-linus and for-next branches,
 respectively.
 
-If you are using the latest Linus tree, it'd be better to pull the
-above GIT tree onto it.  If you are using the older kernels, an easy
-way to try the latest ALSA code is to build from the snapshot
-tarball.  There are daily tarballs and the latest snapshot tarball.
-All can be built just like normal alsa-driver release packages, that
-is, installed via the usual spells: configure, make and make
-install(-modules).  See INSTALL in the package.  The snapshot tarballs
-are found at:
-
-- ftp://ftp.suse.com/pub/people/tiwai/snapshot/
-
 
 Sending a Bug Report
 ~~~~~~~~~~~~~~~~~~~~
@@ -699,7 +688,12 @@ problems.
 alsa-info
 ~~~~~~~~~
 The script `alsa-info.sh` is a very useful tool to gather the audio
-device information.  You can fetch the latest version from:
+device information.  It's included in alsa-utils package.  The latest
+version can be found on git repository:
+
+- git://git.alsa-project.org/alsa-utils.git
+
+The script can be fetched directly from the following URL, too:
 
 - http://www.alsa-project.org/alsa-info.sh
 
@@ -836,15 +830,11 @@ can get a proc-file dump at the current state, get a list of control
 (mixer) elements, set/get the control element value, simulate the PCM
 operation, the jack plugging simulation, etc.
 
-The package is found in:
-
-- ftp://ftp.suse.com/pub/people/tiwai/misc/
-
-A git repository is available:
+The program is found in the git repository below:
 
 - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git
 
-See README file in the tarball for more details about hda-emu
+See README file in the repository for more details about hda-emu
 program.
 
 

Documentation/sound/alsa/compress_offload.txt

@@ -149,7 +149,7 @@ Gapless Playback
 ================
 When playing thru an album, the decoders have the ability to skip the encoder
 delay and padding and directly move from one track content to another. The end
-user can perceive this as gapless playback as we dont have silence while
+user can perceive this as gapless playback as we don't have silence while
 switching from one track to another
 
 Also, there might be low-intensity noises due to encoding. Perfect gapless is
@@ -184,7 +184,7 @@ Sequence flow for gapless would be:
 - Fill data of the first track
 - Trigger start
 - User-space finished sending all,
-- Indicaite next track data by sending set_next_track
+- Indicate next track data by sending set_next_track
 - Set metadata of the next track
 - then call partial_drain to flush most of buffer in DSP
 - Fill data of the next track

Documentation/sound/alsa/soc/dapm.txt

@@ -132,7 +132,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
 SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
 	ARRAY_SIZE(wm8731_output_mixer_controls)),
 
-If you dont want the mixer elements prefixed with the name of the mixer widget,
+If you don't want the mixer elements prefixed with the name of the mixer widget,
 you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
 as for SND_SOC_DAPM_MIXER.
 

Documentation/sound/alsa/soc/overview.txt

@@ -63,7 +63,7 @@ multiple re-usable component drivers :-
     and any audio DSP drivers for that platform.
 
   * Machine class driver: The machine driver class acts as the glue that
-    decribes and binds the other component drivers together to form an ALSA
+    describes and binds the other component drivers together to form an ALSA
     "sound card device". It handles any machine specific controls and
     machine level audio events (e.g. turning on an amp at start of playback).
 

Documentation/sound/alsa/timestamping.txt

@@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an
 interpolation of the results
 
 In some hardware-specific configuration, the system timestamp is
-latched by a low-level audio subsytem, and the information provided
+latched by a low-level audio subsystem, and the information provided
 back to the driver. Due to potential delays in the communication with
 the hardware, there is a risk of misalignment with the avail and delay
 information. To make sure applications are not confused, a

include/sound/hda_chmap.h

@@ -36,6 +36,8 @@ struct hdac_chmap_ops {
 	int (*chmap_validate)(struct hdac_chmap *hchmap, int ca,
 			int channels, unsigned char *chmap);
 
+	int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx);
+
 	void (*get_chmap)(struct hdac_device *hdac, int pcm_idx,
 					unsigned char *chmap);
 	void (*set_chmap)(struct hdac_device *hdac, int pcm_idx,

include/sound/hda_i915.h

@@ -10,8 +10,8 @@
 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
 int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
 void snd_hdac_i915_set_bclk(struct hdac_bus *bus);
-int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate);
-int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate);
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
 			   bool *audio_enabled, char *buffer, int max_bytes);
 int snd_hdac_i915_init(struct hdac_bus *bus);
 int snd_hdac_i915_exit(struct hdac_bus *bus);
@@ -28,12 +28,12 @@ static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
 static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 {
 }
-static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid,
-					   int rate)
+static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec,
+					   hda_nid_t nid, int rate)
 {
 	return 0;
 }
-static inline int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
 					 bool *audio_enabled, char *buffer,
 					 int max_bytes)
 {

include/uapi/sound/asound.h

@@ -672,7 +672,7 @@ enum {
 
 /* global timers (device member) */
 #define SNDRV_TIMER_GLOBAL_SYSTEM	0
-#define SNDRV_TIMER_GLOBAL_RTC		1
+#define SNDRV_TIMER_GLOBAL_RTC		1	/* unused */
 #define SNDRV_TIMER_GLOBAL_HPET		2
 #define SNDRV_TIMER_GLOBAL_HRTIMER	3
 

sound/core/Kconfig

@@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT
 	  Say Y here to use the HR-timer backend as the default sequencer
 	  timer.
 
-config SND_RTCTIMER
-	tristate "RTC Timer support"
-	depends on RTC
-	select SND_TIMER
-	help
-	  Say Y here to enable RTC timer support for ALSA.  ALSA uses
-	  the RTC timer as a precise timing source and maps the RTC
-	  timer to ALSA's timer interface.  The ALSA sequencer code also
-	  can use this timing source.
-
-	  To compile this driver as a module, choose M here: the module
-	  will be called snd-rtctimer.
-
-	  Note that this option is exclusive with the new RTC drivers
-	  (CONFIG_RTC_CLASS) since this requires the old API.
-
-config SND_SEQ_RTCTIMER_DEFAULT
-	bool "Use RTC as default sequencer timer"
-	depends on SND_RTCTIMER && SND_SEQUENCER
-	depends on !SND_SEQ_HRTIMER_DEFAULT
-	default y
-	help
-	  Say Y here to use the RTC timer as the default sequencer
-	  timer.  This is strongly recommended because it ensures
-	  precise MIDI timing even when the system timer runs at less
-	  than 1000 Hz.
-
-	  If in doubt, say Y.
-
 config SND_DYNAMIC_MINORS
 	bool "Dynamic device file minor numbers"
 	help

sound/core/Makefile

@@ -37,7 +37,6 @@ obj-$(CONFIG_SND) 		+= snd.o
 obj-$(CONFIG_SND_HWDEP)		+= snd-hwdep.o
 obj-$(CONFIG_SND_TIMER)		+= snd-timer.o
 obj-$(CONFIG_SND_HRTIMER)	+= snd-hrtimer.o
-obj-$(CONFIG_SND_RTCTIMER)	+= snd-rtctimer.o
 obj-$(CONFIG_SND_PCM)		+= snd-pcm.o
 obj-$(CONFIG_SND_DMAENGINE_PCM)	+= snd-pcm-dmaengine.o
 obj-$(CONFIG_SND_RAWMIDI)	+= snd-rawmidi.o

sound/core/compress_offload.c

@@ -288,9 +288,12 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf,
 	stream = &data->stream;
 	mutex_lock(&stream->device->lock);
 	/* write is allowed when stream is running or has been steup */
-	if (stream->runtime->state != SNDRV_PCM_STATE_SETUP &&
-	    stream->runtime->state != SNDRV_PCM_STATE_PREPARED &&
-			stream->runtime->state != SNDRV_PCM_STATE_RUNNING) {
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
+	case SNDRV_PCM_STATE_RUNNING:
+		break;
+	default:
 		mutex_unlock(&stream->device->lock);
 		return -EBADFD;
 	}
@@ -391,14 +394,13 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait)
 	int retval = 0;
 
 	if (snd_BUG_ON(!data))
-		return -EFAULT;
+		return POLLERR;
+
 	stream = &data->stream;
-	if (snd_BUG_ON(!stream))
-		return -EFAULT;
 
 	mutex_lock(&stream->device->lock);
 	if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) {
-		retval = -EBADFD;
+		retval = snd_compr_get_poll(stream) | POLLERR;
 		goto out;
 	}
 	poll_wait(f, &stream->runtime->sleep, wait);
@@ -421,10 +423,7 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait)
 			retval = snd_compr_get_poll(stream);
 		break;
 	default:
-		if (stream->direction == SND_COMPRESS_PLAYBACK)
-			retval = POLLOUT | POLLWRNORM | POLLERR;
-		else
-			retval = POLLIN | POLLRDNORM | POLLERR;
+		retval = snd_compr_get_poll(stream) | POLLERR;
 		break;
 	}
 out:
@@ -802,9 +801,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg)
 
 	if (snd_BUG_ON(!data))
 		return -EFAULT;
+
 	stream = &data->stream;
-	if (snd_BUG_ON(!stream))
-		return -EFAULT;
+
 	mutex_lock(&stream->device->lock);
 	switch (_IOC_NR(cmd)) {
 	case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION):

sound/core/hrtimer.c

@@ -38,37 +38,53 @@ static unsigned int resolution;
 struct snd_hrtimer {
 	struct snd_timer *timer;
 	struct hrtimer hrt;
-	atomic_t running;
+	bool in_callback;
 };
 
 static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
 {
 	struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
 	struct snd_timer *t = stime->timer;
-	unsigned long oruns;
-
-	if (!atomic_read(&stime->running))
-		return HRTIMER_NORESTART;
-
-	oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
-	snd_timer_interrupt(stime->timer, t->sticks * oruns);
+	ktime_t delta;
+	unsigned long ticks;
+	enum hrtimer_restart ret = HRTIMER_NORESTART;
+
+	spin_lock(&t->lock);
+	if (!t->running)
+		goto out; /* fast path */
+	stime->in_callback = true;
+	ticks = t->sticks;
+	spin_unlock(&t->lock);
+
+	/* calculate the drift */
+	delta = ktime_sub(hrt->base->get_time(), hrtimer_get_expires(hrt));
+	if (delta.tv64 > 0)
+		ticks += ktime_divns(delta, ticks * resolution);
+
+	snd_timer_interrupt(stime->timer, ticks);
+
+	spin_lock(&t->lock);
+	if (t->running) {
+		hrtimer_add_expires_ns(hrt, t->sticks * resolution);
+		ret = HRTIMER_RESTART;
+	}
 
-	if (!atomic_read(&stime->running))
-		return HRTIMER_NORESTART;
-	return HRTIMER_RESTART;
+	stime->in_callback = false;
+ out:
+	spin_unlock(&t->lock);
+	return ret;
 }
 
 static int snd_hrtimer_open(struct snd_timer *t)
 {
 	struct snd_hrtimer *stime;
 
-	stime = kmalloc(sizeof(*stime), GFP_KERNEL);
+	stime = kzalloc(sizeof(*stime), GFP_KERNEL);
 	if (!stime)
 		return -ENOMEM;
 	hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
 	stime->timer = t;
 	stime->hrt.function = snd_hrtimer_callback;
-	atomic_set(&stime->running, 0);
 	t->private_data = stime;
 	return 0;
 }
@@ -78,6 +94,11 @@ static int snd_hrtimer_close(struct snd_timer *t)
 	struct snd_hrtimer *stime = t->private_data;
 
 	if (stime) {
+		spin_lock_irq(&t->lock);
+		t->running = 0; /* just to be sure */
+		stime->in_callback = 1; /* skip start/stop */
+		spin_unlock_irq(&t->lock);
+
 		hrtimer_cancel(&stime->hrt);
 		kfree(stime);
 		t->private_data = NULL;
@@ -89,18 +110,19 @@ static int snd_hrtimer_start(struct snd_timer *t)
 {
 	struct snd_hrtimer *stime = t->private_data;
 
-	atomic_set(&stime->running, 0);
-	hrtimer_try_to_cancel(&stime->hrt);
+	if (stime->in_callback)
+		return 0;
 	hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution),
 		      HRTIMER_MODE_REL);
-	atomic_set(&stime->running, 1);
 	return 0;
 }
 
 static int snd_hrtimer_stop(struct snd_timer *t)
 {
 	struct snd_hrtimer *stime = t->private_data;
-	atomic_set(&stime->running, 0);
+
+	if (stime->in_callback)
+		return 0;
 	hrtimer_try_to_cancel(&stime->hrt);
 	return 0;
 }

sound/core/pcm_lib.c

@@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
 		snd_timer_interrupt(substream->timer, 1);
 #endif
  _end:
-	snd_pcm_stream_unlock_irqrestore(substream, flags);
 	kill_fasync(&runtime->fasync, SIGIO, POLL_IN);
+	snd_pcm_stream_unlock_irqrestore(substream, flags);
 }
 
 EXPORT_SYMBOL(snd_pcm_period_elapsed);
@@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
 	};
 	int err;
 
+	if (WARN_ON(pcm->streams[stream].chmap_kctl))
+		return -EBUSY;
 	info = kzalloc(sizeof(*info), GFP_KERNEL);
 	if (!info)
 		return -ENOMEM;

sound/core/pcm_native.c

@@ -3161,7 +3161,7 @@ static unsigned int snd_pcm_playback_poll(struct file *file, poll_table * wait)
 
 	substream = pcm_file->substream;
 	if (PCM_RUNTIME_CHECK(substream))
-		return -ENXIO;
+		return POLLOUT | POLLWRNORM | POLLERR;
 	runtime = substream->runtime;
 
 	poll_wait(file, &runtime->sleep, wait);
@@ -3200,7 +3200,7 @@ static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait)
 
 	substream = pcm_file->substream;
 	if (PCM_RUNTIME_CHECK(substream))
-		return -ENXIO;
+		return POLLIN | POLLRDNORM | POLLERR;
 	runtime = substream->runtime;
 
 	poll_wait(file, &runtime->sleep, wait);