Merge branch 'topic/misc' into for-linus

78 files changed, 2204 insertions, 1647 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 8c16d50f6cb6..221b81016dba 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -1545,7 +1545,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     Module for sound cards based on the C-Media CMI8786/8787/8788 chip:
     * Asound A-8788
-    * Asus Xonar DG
+    * Asus Xonar DG/DGX
     * AuzenTech X-Meridian
     * AuzenTech X-Meridian 2G
     * Bgears b-Enspirer
diff --git a/include/sound/asound.h b/include/sound/asound.h
index a2e4ff5ba9e9..0876a1e76aef 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -70,6 +70,20 @@ struct snd_aes_iec958 {
 
 /****************************************************************************
  *                                                                          *
+ *        CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort		    *
+ *                                                                          *
+ ****************************************************************************/
+
+struct snd_cea_861_aud_if {
+	unsigned char db1_ct_cc; /* coding type and channel count */
+	unsigned char db2_sf_ss; /* sample frequency and size */
+	unsigned char db3; /* not used, all zeros */
+	unsigned char db4_ca; /* channel allocation code */
+	unsigned char db5_dminh_lsv; /* downmix inhibit & level-shit values */
+};
+
+/****************************************************************************
+ *                                                                          *
  *      Section for driver hardware dependent interface - /dev/snd/hw?      *
  *                                                                          *
  ****************************************************************************/
diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h
index 20ebf3298eba..bb05c02f89b0 100644
--- a/include/sound/asoundef.h
+++ b/include/sound/asoundef.h
@@ -170,6 +170,47 @@
 #define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0)	/* condition not be used */
 #define IEC958_AES5_CON_CGMSA_COPYNEVER	(3<<0)	/* no copying is permitted */
 
+/****************************************************************************
+ *                                                                          *
+ *        CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort		    *
+ *                                                                          *
+ ****************************************************************************/
+#define CEA861_AUDIO_INFOFRAME_DB1CC		(7<<0) /* mask - channel count */
+#define CEA861_AUDIO_INFOFRAME_DB1CT		(0xf<<4) /* mask - coding type */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM (0<<4) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_IEC60958	(1<<4) /* IEC-60958 L-PCM */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AC3	(2<<4) /* AC-3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG1	(3<<4) /* MPEG1 Layers 1 & 2 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MP3	(4<<4) /* MPEG1 Layer 3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG2_MULTICH (5<<4) /* MPEG2 Multichannel */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AAC	(6<<4) /* AAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS	(7<<4) /* DTS */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ATRAC	(8<<4) /* ATRAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ONEBIT	(9<<4) /* One Bit Audio */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DOLBY_DIG_PLUS (10<<4) /* Dolby Digital + */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS_HD	(11<<4) /* DTS-HD */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MAT	(12<<4) /* MAT (MLP) */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DST	(13<<4) /* DST */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_WMA_PRO	(14<<4) /* WMA Pro */
+#define CEA861_AUDIO_INFOFRAME_DB2SF		(7<<2) /* mask - sample frequency */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM (0<<2) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_32000	(1<<2) /* 32kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_44100	(2<<2) /* 44.1kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_48000	(3<<2) /* 48kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_88200	(4<<2) /* 88.2kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_96000	(5<<2) /* 96kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_176400	(6<<2) /* 176.4kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_192000	(7<<2) /* 192kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SS		(3<<0) /* mask - sample size */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM (0<<0) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_16BIT	(1<<0) /* 16 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_20BIT	(2<<0) /* 20 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_24BIT	(3<<0) /* 24 bits */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH	(1<<7) /* mask - inhibit downmixing */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED (0<<7) /* stereo downmix permitted */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED (1<<7) /* stereo downmis prohibited */
+#define CEA861_AUDIO_INFOFRAME_DB5_LSV		(0xf<<3) /* mask - level-shift values */
+
 /*****************************************************************************
  *                                                                           *
  *                            MIDI v1.0 interface                            *
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 115313ef54d6..f5ded640b395 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -991,6 +991,8 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 			gpio_direction_output(pdata->reset_pin, 1);
 			chip->reset_pin = pdata->reset_pin;
 		}
+	} else {
+		chip->reset_pin = -EINVAL;
 	}
 
 	snd_card_set_dev(card, &pdev->dev);
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 471e1e3b0a99..a06b1651fcba 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -155,7 +155,7 @@ EXPORT_SYMBOL(snd_jack_new);
  * @jack:   The jack to configure
  * @parent: The device to set as parent for the jack.
  *
- * Set the parent for the jack input device in the device tree.  This
+ * Set the parent for the jack devices in the device tree.  This
  * function is only valid prior to registration of the jack.  If no
  * parent is configured then the parent device will be the sound card.
  */
@@ -179,6 +179,9 @@ EXPORT_SYMBOL(snd_jack_set_parent);
  * mapping is provided but keys are enabled in the jack type then
  * BTN_n numeric buttons will be reported.
  *
+ * If jacks are not reporting via the input API this call will have no
+ * effect.
+ *
  * Note that this is intended to be use by simple devices with small
  * numbers of keys that can be reported.  It is also possible to
  * access the input device directly - devices with complex input
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4d18941178e6..faedb1481b24 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1894,6 +1894,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t xfer = 0;
 	snd_pcm_uframes_t offset = 0;
+	snd_pcm_uframes_t avail;
 	int err = 0;
 
 	if (size == 0)
@@ -1917,13 +1918,12 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
 	}
 
 	runtime->twake = runtime->control->avail_min ? : 1;
+	if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+		snd_pcm_update_hw_ptr(substream);
+	avail = snd_pcm_playback_avail(runtime);
 	while (size > 0) {
 		snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
-		snd_pcm_uframes_t avail;
 		snd_pcm_uframes_t cont;
-		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-			snd_pcm_update_hw_ptr(substream);
-		avail = snd_pcm_playback_avail(runtime);
 		if (!avail) {
 			if (nonblock) {
 				err = -EAGAIN;
@@ -1971,6 +1971,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
 		offset += frames;
 		size -= frames;
 		xfer += frames;
+		avail -= frames;
 		if (runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
 		    snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) {
 			err = snd_pcm_start(substream);
@@ -2111,6 +2112,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t xfer = 0;
 	snd_pcm_uframes_t offset = 0;
+	snd_pcm_uframes_t avail;
 	int err = 0;
 
 	if (size == 0)
@@ -2141,13 +2143,12 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
 	}
 
 	runtime->twake = runtime->control->avail_min ? : 1;
+	if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+		snd_pcm_update_hw_ptr(substream);
+	avail = snd_pcm_capture_avail(runtime);
 	while (size > 0) {
 		snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
-		snd_pcm_uframes_t avail;
 		snd_pcm_uframes_t cont;
-		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-			snd_pcm_update_hw_ptr(substream);
-		avail = snd_pcm_capture_avail(runtime);
 		if (!avail) {
 			if (runtime->status->state ==
 			    SNDRV_PCM_STATE_DRAINING) {
@@ -2202,6 +2203,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
 		offset += frames;
 		size -= frames;
 		xfer += frames;
+		avail -= frames;
 	}
  _end_unlock:
 	runtime->twake = 0;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 3fe99e644eb8..53b5ada8f7c3 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1360,7 +1360,14 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream,
 
 static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state)
 {
-	substream->runtime->trigger_master = substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	switch (runtime->status->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_DISCONNECTED:
+	case SNDRV_PCM_STATE_SUSPENDED:
+		return -EBADFD;
+	}
+	runtime->trigger_master = substream;
 	return 0;
 }
 
@@ -1379,6 +1386,9 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state)
 		case SNDRV_PCM_STATE_RUNNING:
 			runtime->status->state = SNDRV_PCM_STATE_DRAINING;
 			break;
+		case SNDRV_PCM_STATE_XRUN:
+			runtime->status->state = SNDRV_PCM_STATE_SETUP;
+			break;
 		default:
 			break;
 		}
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index c70092043061..e9528333e36d 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -35,7 +35,7 @@
 #include <linux/sound.h>
 #include <linux/mutex.h>
 
-#define SNDRV_OSS_MINORS 128
+#define SNDRV_OSS_MINORS 256
 
 static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS];
 static DEFINE_MUTEX(sound_oss_mutex);
@@ -111,7 +111,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev,
 	int register1 = -1, register2 = -1;
 	struct device *carddev = snd_card_get_device_link(card);
 
-	if (card && card->number >= 8)
+	if (card && card->number >= SNDRV_MINOR_OSS_DEVICES)
 		return 0; /* ignore silently */
 	if (minor < 0)
 		return minor;
@@ -170,7 +170,7 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev)
 	int track2 = -1;
 	struct snd_minor *mptr;
 
-	if (card && card->number >= 8)
+	if (card && card->number >= SNDRV_MINOR_OSS_DEVICES)
 		return 0;
 	if (minor < 0)
 		return minor;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index ad079b63b8ba..8b5c36f4d303 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -117,6 +117,7 @@ struct loopback_pcm {
 	/* timer stuff */
 	unsigned int irq_pos;		/* fractional IRQ position */
 	unsigned int period_size_frac;
+	unsigned int last_drift;
 	unsigned long last_jiffies;
 	struct timer_list timer;
 };
@@ -264,6 +265,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd)
 			return err;
 		dpcm->last_jiffies = jiffies;
 		dpcm->pcm_rate_shift = 0;
+		dpcm->last_drift = 0;
 		spin_lock(&cable->lock);	
 		cable->running |= stream;
 		cable->pause &= ~stream;
@@ -444,34 +446,30 @@ static void copy_play_buf(struct loopback_pcm *play,
 	}
 }
 
-#define BYTEPOS_UPDATE_POSONLY	0
-#define BYTEPOS_UPDATE_CLEAR	1
-#define BYTEPOS_UPDATE_COPY	2
-
-static void loopback_bytepos_update(struct loopback_pcm *dpcm,
-				    unsigned int delta,
-				    unsigned int cmd)
+static inline unsigned int bytepos_delta(struct loopback_pcm *dpcm,
+					 unsigned int jiffies_delta)
 {
-	unsigned int count;
 	unsigned long last_pos;
+	unsigned int delta;
 
 	last_pos = byte_pos(dpcm, dpcm->irq_pos);
-	dpcm->irq_pos += delta * dpcm->pcm_bps;
-	count = byte_pos(dpcm, dpcm->irq_pos) - last_pos;
-	if (!count)
-		return;
-	if (cmd == BYTEPOS_UPDATE_CLEAR)
-		clear_capture_buf(dpcm, count);
-	else if (cmd == BYTEPOS_UPDATE_COPY)
-		copy_play_buf(dpcm->cable->streams[SNDRV_PCM_STREAM_PLAYBACK],
-			      dpcm->cable->streams[SNDRV_PCM_STREAM_CAPTURE],
-			      count);
-	dpcm->buf_pos += count;
-	dpcm->buf_pos %= dpcm->pcm_buffer_size;
+	dpcm->irq_pos += jiffies_delta * dpcm->pcm_bps;
+	delta = byte_pos(dpcm, dpcm->irq_pos) - last_pos;
+	if (delta >= dpcm->last_drift)
+		delta -= dpcm->last_drift;
+	dpcm->last_drift = 0;
 	if (dpcm->irq_pos >= dpcm->period_size_frac) {
 		dpcm->irq_pos %= dpcm->period_size_frac;
 		dpcm->period_update_pending = 1;
 	}
+	return delta;
+}
+
+static inline void bytepos_finish(struct loopback_pcm *dpcm,
+				  unsigned int delta)
+{
+	dpcm->buf_pos += delta;
+	dpcm->buf_pos %= dpcm->pcm_buffer_size;
 }
 
 static unsigned int loopback_pos_update(struct loopback_cable *cable)
@@ -481,7 +479,7 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
 	struct loopback_pcm *dpcm_capt =
 			cable->streams[SNDRV_PCM_STREAM_CAPTURE];
 	unsigned long delta_play = 0, delta_capt = 0;
-	unsigned int running;
+	unsigned int running, count1, count2;
 	unsigned long flags;
 
 	spin_lock_irqsave(&cable->lock, flags);
@@ -500,12 +498,13 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
 		goto unlock;
 		
 	if (delta_play > delta_capt) {
-		loopback_bytepos_update(dpcm_play, delta_play - delta_capt,
-					BYTEPOS_UPDATE_POSONLY);
+		count1 = bytepos_delta(dpcm_play, delta_play - delta_capt);
+		bytepos_finish(dpcm_play, count1);
 		delta_play = delta_capt;
 	} else if (delta_play < delta_capt) {
-		loopback_bytepos_update(dpcm_capt, delta_capt - delta_play,
-					BYTEPOS_UPDATE_CLEAR);
+		count1 = bytepos_delta(dpcm_capt, delta_capt - delta_play);
+		clear_capture_buf(dpcm_capt, count1);
+		bytepos_finish(dpcm_capt, count1);
 		delta_capt = delta_play;
 	}
 
@@ -513,8 +512,17 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
 		goto unlock;
 
 	/* note delta_capt == delta_play at this moment */
-	loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY);
-	loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY);
+	count1 = bytepos_delta(dpcm_play, delta_play);
+	count2 = bytepos_delta(dpcm_capt, delta_capt);
+	if (count1 < count2) {
+		dpcm_capt->last_drift = count2 - count1;
+		count1 = count2;
+	} else if (count1 > count2) {
+		dpcm_play->last_drift = count1 - count2;
+	}
+	copy_play_buf(dpcm_play, dpcm_capt, count1);
+	bytepos_finish(dpcm_play, count1);
+	bytepos_finish(dpcm_capt, count1);
  unlock:
 	spin_unlock_irqrestore(&cable->lock, flags);
 	return running;
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 87657dd7714c..ea995af6d049 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -31,6 +31,8 @@
 #define INTERRUPT_INTERVAL	16
 #define QUEUE_LENGTH		48
 
+static void pcm_period_tasklet(unsigned long data);
+
 /**
  * amdtp_out_stream_init - initialize an AMDTP output stream structure
  * @s: the AMDTP output stream to initialize
@@ -47,6 +49,7 @@ int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
 	s->flags = flags;
 	s->context = ERR_PTR(-1);
 	mutex_init(&s->mutex);
+	tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s);
 	s->packet_index = 0;
 
 	return 0;
@@ -164,6 +167,21 @@ void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
 }
 EXPORT_SYMBOL(amdtp_out_stream_set_pcm_format);
 
+/**
+ * amdtp_out_stream_pcm_prepare - prepare PCM device for running
+ * @s: the AMDTP output stream
+ *
+ * This function should be called from the PCM device's .prepare callback.
+ */
+void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s)
+{
+	tasklet_kill(&s->period_tasklet);
+	s->pcm_buffer_pointer = 0;
+	s->pcm_period_pointer = 0;
+	s->pointer_flush = true;
+}
+EXPORT_SYMBOL(amdtp_out_stream_pcm_prepare);
+
 static unsigned int calculate_data_blocks(struct amdtp_out_stream *s)
 {
 	unsigned int phase, data_blocks;
@@ -376,11 +394,21 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle)
 		s->pcm_period_pointer += data_blocks;
 		if (s->pcm_period_pointer >= pcm->runtime->period_size) {
 			s->pcm_period_pointer -= pcm->runtime->period_size;
-			snd_pcm_period_elapsed(pcm);
+			s->pointer_flush = false;