Merge commit 'perf/core' into perf/hw-breakpoint

Conflicts: kernel/Makefile kernel/trace/Makefile kernel/trace/trace.h samples/Makefile Merge reason: We need to be uptodate with the perf events development branch because we plan to rewrite the breakpoints API on top of perf events.
author: Frederic Weisbecker <fweisbec@gmail.com> 2009-10-18 01:09:09 +0200
committer: Frederic Weisbecker <fweisbec@gmail.com> 2009-10-18 01:12:33 +0200
commit: 0f8f86c7bdd1c954fbe153af437a0d91a6c5721a (patch)
tree: 94a8d419a470a4f9852ca397bb9bbe48db92ff5c /sound
parent: dca2d6ac09d9ef59ff46820d4f0c94b08a671202 (diff)
parent: f39cdf25bf77219676ec5360980ac40b1a7e144a (diff)
58 files changed, 866 insertions, 587 deletions
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index f0ebc971c68..1dd66ddffca 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -897,6 +897,15 @@ static int tas_create(struct i2c_adapter *adapter,
 	client = i2c_new_device(adapter, &info);
 	if (!client)
 		return -ENODEV;
+	/*
+	 * We know the driver is already loaded, so the device should be
+	 * already bound. If not it means binding failed, and then there
+	 * is no point in keeping the device instantiated.
+	 */
+	if (!client->driver) {
+		i2c_unregister_device(client);
+		return -ENODEV;
+	}
 
 	/*
 	 * Let i2c-core delete that device on driver removal.
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index dc78272fc39..1f0f8213e2d 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -937,6 +937,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci)
 	struct snd_ac97 *ac97;
 	int ret;
 
+	writel(0, aaci->base + AC97_POWERDOWN);
 	/*
 	 * Assert AACIRESET for 2us
 	 */
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 4e34d19ddbc..b4b48afb6de 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -137,9 +137,9 @@ static int pxa2xx_ac97_do_resume(struct snd_card *card)
 	return 0;
 }
 
-static int pxa2xx_ac97_suspend(struct platform_device *dev, pm_message_t state)
+static int pxa2xx_ac97_suspend(struct device *dev)
 {
-	struct snd_card *card = platform_get_drvdata(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	int ret = 0;
 
 	if (card)
@@ -148,9 +148,9 @@ static int pxa2xx_ac97_suspend(struct platform_device *dev, pm_message_t state)
 	return ret;
 }
 
-static int pxa2xx_ac97_resume(struct platform_device *dev)
+static int pxa2xx_ac97_resume(struct device *dev)
 {
-	struct snd_card *card = platform_get_drvdata(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	int ret = 0;
 
 	if (card)
@@ -159,9 +159,10 @@ static int pxa2xx_ac97_resume(struct platform_device *dev)
 	return ret;
 }
 
-#else
-#define pxa2xx_ac97_suspend	NULL
-#define pxa2xx_ac97_resume	NULL
+static struct dev_pm_ops pxa2xx_ac97_pm_ops = {
+	.suspend	= pxa2xx_ac97_suspend,
+	.resume		= pxa2xx_ac97_resume,
+};
 #endif
 
 static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
@@ -241,11 +242,12 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev)
 static struct platform_driver pxa2xx_ac97_driver = {
 	.probe		= pxa2xx_ac97_probe,
 	.remove		= __devexit_p(pxa2xx_ac97_remove),
-	.suspend	= pxa2xx_ac97_suspend,
-	.resume		= pxa2xx_ac97_resume,
 	.driver		= {
 		.name	= "pxa2xx-ac97",
 		.owner	= THIS_MODULE,
+#ifdef CONFIG_PM
+		.pm	= &pxa2xx_ac97_pm_ops,
+#endif
 	},
 };
 
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 59e5fbe6af5..ab73edf2c89 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1387,11 +1387,6 @@ static struct action_ops snd_pcm_action_drain_init = {
 	.post_action = snd_pcm_post_drain_init
 };
 
-struct drain_rec {
-	struct snd_pcm_substream *substream;
-	wait_queue_t wait;
-};
-
 static int snd_pcm_drop(struct snd_pcm_substream *substream);
 
 /*
@@ -1407,10 +1402,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 	struct snd_card *card;
 	struct snd_pcm_runtime *runtime;
 	struct snd_pcm_substream *s;
+	wait_queue_t wait;
 	int result = 0;
-	int i, num_drecs;
 	int nonblock = 0;
-	struct drain_rec *drec, drec_tmp, *d;
 
 	card = substream->pcm->card;
 	runtime = substream->runtime;
@@ -1433,38 +1427,10 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 	} else if (substream->f_flags & O_NONBLOCK)
 		nonblock = 1;
 
-	if (nonblock)
-		goto lock; /* no need to allocate waitqueues */
-
-	/* allocate temporary record for drain sync */
 	down_read(&snd_pcm_link_rwsem);
-	if (snd_pcm_stream_linked(substream)) {
-		drec = kmalloc(substream->group->count * sizeof(*drec), GFP_KERNEL);
-		if (! drec) {
-			up_read(&snd_pcm_link_rwsem);
-			snd_power_unlock(card);
-			return -ENOMEM;
-		}
-	} else
-		drec = &drec_tmp;
-
-	/* count only playback streams */
-	num_drecs = 0;
-	snd_pcm_group_for_each_entry(s, substream) {
-		runtime = s->runtime;
-		if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			d = &drec[num_drecs++];
-			d->substream = s;
-			init_waitqueue_entry(&d->wait, current);
-			add_wait_queue(&runtime->sleep, &d->wait);
-		}
-	}
-	up_read(&snd_pcm_link_rwsem);
-
- lock:
 	snd_pcm_stream_lock_irq(substream);
 	/* resume pause */
-	if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED)
+	if (runtime->status->state == SNDRV_PCM_STATE_PAUSED)
 		snd_pcm_pause(substream, 0);
 
 	/* pre-start/stop - all running streams are changed to DRAINING state */
@@ -1479,25 +1445,35 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 
 	for (;;) {
 		long tout;
+		struct snd_pcm_runtime *to_check;
 		if (signal_pending(current)) {
 			result = -ERESTARTSYS;
 			break;
 		}
-		/* all finished? */
-		for (i = 0; i < num_drecs; i++) {
-			runtime = drec[i].substream->runtime;
-			if (runtime->status->state == SNDRV_PCM_STATE_DRAINING)
+		/* find a substream to drain */
+		to_check = NULL;
+		snd_pcm_group_for_each_entry(s, substream) {
+			if (s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+				continue;
+			runtime = s->runtime;
+			if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
+				to_check = runtime;
 				break;
+			}
 		}
-		if (i == num_drecs)
-			break; /* yes, all drained */
-
+		if (!to_check)
+			break; /* all drained */
+		init_waitqueue_entry(&wait, current);
+		add_wait_queue(&to_check->sleep, &wait);
 		set_current_state(TASK_INTERRUPTIBLE);
 		snd_pcm_stream_unlock_irq(substream);
+		up_read(&snd_pcm_link_rwsem);
 		snd_power_unlock(card);
 		tout = schedule_timeout(10 * HZ);
 		snd_power_lock(card);
+		down_read(&snd_pcm_link_rwsem);
 		snd_pcm_stream_lock_irq(substream);
+		remove_wait_queue(&to_check->sleep, &wait);
 		if (tout == 0) {
 			if (substream->runtime->status->state == SNDRV_PCM_STATE_SUSPENDED)
 				result = -ESTRPIPE;
@@ -1512,16 +1488,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 
  unlock:
 	snd_pcm_stream_unlock_irq(substream);
-
-	if (!nonblock) {
-		for (i = 0; i < num_drecs; i++) {
-			d = &drec[i];
-			runtime = d->substream->runtime;
-			remove_wait_queue(&runtime->sleep, &d->wait);
-		}
-		if (drec != &drec_tmp)
-			kfree(drec);
-	}
+	up_read(&snd_pcm_link_rwsem);
 	snd_power_unlock(card);
 
 	return result;
@@ -3018,7 +2985,7 @@ static int snd_pcm_mmap_status_fault(struct vm_area_struct *area,
 	return 0;
 }
 
-static struct vm_operations_struct snd_pcm_vm_ops_status =
+static const struct vm_operations_struct snd_pcm_vm_ops_status =
 {
 	.fault =	snd_pcm_mmap_status_fault,
 };
@@ -3057,7 +3024,7 @@ static int snd_pcm_mmap_control_fault(struct vm_area_struct *area,
 	return 0;
 }
 
-static struct vm_operations_struct snd_pcm_vm_ops_control =
+static const struct vm_operations_struct snd_pcm_vm_ops_control =
 {
 	.fault =	snd_pcm_mmap_control_fault,
 };
@@ -3127,7 +3094,7 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area,
 	return 0;
 }
 
-static struct vm_operations_struct snd_pcm_vm_ops_data =
+static const struct vm_operations_struct snd_pcm_vm_ops_data =
 {
 	.open =		snd_pcm_mmap_data_open,
 	.close =	snd_pcm_mmap_data_close,
@@ -3151,7 +3118,7 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream,
  * mmap the DMA buffer on I/O memory area
  */
 #if SNDRV_PCM_INFO_MMAP_IOMEM
-static struct vm_operations_struct snd_pcm_vm_ops_data_mmio =
+static const struct vm_operations_struct snd_pcm_vm_ops_data_mmio =
 {
 	.open =		snd_pcm_mmap_data_open,
 	.close =	snd_pcm_mmap_data_close,
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index 6e7d09ae0e8..7d722a025d0 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -29,6 +29,8 @@ extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
 
 extern int use_internal_drums;
 
+static void snd_opl3_note_off_unsafe(void *p, int note, int vel,
+				     struct snd_midi_channel *chan);
 /*
  * The next table looks magical, but it certainly is not. Its values have
  * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception
@@ -242,16 +244,20 @@ void snd_opl3_timer_func(unsigned long data)
 	int again = 0;
 	int i;
 
-	spin_lock_irqsave(&opl3->sys_timer_lock, flags);
+	spin_lock_irqsave(&opl3->voice_lock, flags);
 	for (i = 0; i < opl3->max_voices; i++) {
 		struct snd_opl3_voice *vp = &opl3->voices[i];
 		if (vp->state > 0 && vp->note_off_check) {
 			if (vp->note_off == jiffies)
-				snd_opl3_note_off(opl3, vp->note, 0, vp->chan);
+				snd_opl3_note_off_unsafe(opl3, vp->note, 0,
+							 vp->chan);
 			else
 				again++;
 		}
 	}
+	spin_unlock_irqrestore(&opl3->voice_lock, flags);
+
+	spin_lock_irqsave(&opl3->sys_timer_lock, flags);
 	if (again) {
 		opl3->tlist.expires = jiffies + 1;	/* invoke again */
 		add_timer(&opl3->tlist);
@@ -658,15 +664,14 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice)
 /*
  * Release a note in response to a midi note off.
  */
-void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan)
+static void snd_opl3_note_off_unsafe(void *p, int note, int vel,
+				     struct snd_midi_channel *chan)
 {
   	struct snd_opl3 *opl3;
 
 	int voice;
 	struct snd_opl3_voice *vp;
 
-	unsigned long flags;
-
 	opl3 = p;
 
 #ifdef DEBUG_MIDI
@@ -674,12 +679,9 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan
 		   chan->number, chan->midi_program, note);
 #endif
 
-	spin_lock_irqsave(&opl3->voice_lock, flags);
-
 	if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) {
 		if (chan->drum_channel && use_internal_drums) {
 			snd_opl3_drum_switch(opl3, note, vel, 0, chan);
-			spin_unlock_irqrestore(&opl3->voice_lock, flags);
 			return;
 		}
 		/* this loop will hopefully kill all extra voices, because
@@ -697,6 +699,16 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan
 			snd_opl3_kill_voice(opl3, voice);
 		}
 	}
+}
+
+void snd_opl3_note_off(void *p, int note, int vel,
+		       struct snd_midi_channel *chan)
+{
+	struct snd_opl3 *opl3 = p;
+	unsigned long flags;
+
+	spin_lock_irqsave(&opl3->voice_lock, flags);
+	snd_opl3_note_off_unsafe(p, note, vel, chan);
 	spin_unlock_irqrestore(&opl3->voice_lock, flags);
 }
 
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index c52691c2fc4..9a88cdfd952 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -915,7 +915,7 @@ static int __devinit hal2_probe(struct platform_device *pdev)
 	return 0;
 }
 
-static int __exit hal2_remove(struct platform_device *pdev)
+static int __devexit hal2_remove(struct platform_device *pdev)
 {
 	struct snd_card *card = platform_get_drvdata(pdev);
 
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index e497525bc11..8691f4cf619 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -973,7 +973,7 @@ static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
 	return 0;
 }
 
-static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
+static int __devexit snd_sgio2audio_remove(struct platform_device *pdev)
 {
 	struct snd_card *card = platform_get_drvdata(pdev);
 
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 1edab7b4ea8..3136c88eacd 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -110,9 +110,6 @@ static void start_adc(struct cs4297a_state *s);
 // rather than 64k as some of the games work more responsively.
 // log base 2( buff sz = 32k).
 
-//static unsigned long defaultorder = 3;
-//MODULE_PARM(defaultorder, "i");
-
 //
 // Turn on/off debugging compilation by commenting out "#define CSDEBUG"
 //
diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c
index 107534477a2..8db6aefe15e 100644
--- a/sound/oss/sys_timer.c
+++ b/sound/oss/sys_timer.c
@@ -100,9 +100,6 @@ def_tmr_open(int dev, int mode)
 	curr_tempo = 60;
 	curr_timebase = 100;
 	opened = 1;
-
-	;
-
 	{
 		def_tmr.expires = (1) + jiffies;
 		add_timer(&def_tmr);
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 24585c6c6d0..4e2b925a94c 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -808,6 +808,8 @@ static struct pci_device_id snd_bt87x_ids[] = {
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC),
 	/* Leadtek Winfast tv 2000xp delux */
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC),
+	/* Pinnacle PCTV */
+	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x11bd, 0x0012, GENERIC),
 	/* Voodoo TV 200 */
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC),
 	/* Askey Computer Corp. MagicTView'99 */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index b1b3a644f73..75454648d50 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1037,7 +1037,7 @@ static int atc_line_front_unmute(struct ct_atc *atc, unsigned char state)
 
 static int atc_line_surround_unmute(struct ct_atc *atc, unsigned char state)
 {
-	return atc_daio_unmute(atc, state, LINEO4);
+	return atc_daio_unmute(atc, state, LINEO2);
 }
 
 static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
@@ -1047,7 +1047,7 @@ static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
 
 static int atc_line_rear_unmute(struct ct_atc *atc, unsigned char state)
 {
-	return atc_daio_unmute(atc, state, LINEO2);
+	return atc_daio_unmute(atc, state, LINEO4);
 }
 
 static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state)
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index da2065cd2c0..1305f7ca02c 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -950,7 +950,7 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
 	Control interface
 ******************************************************************************/
 
-#ifndef ECHOCARD_HAS_VMIXER
+#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN)
 
 /******************* PCM output volume *******************/
 static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
@@ -1003,6 +1003,19 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
+#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
+/* On the Mia this one controls the line-out volume */
+static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
+	.name = "Line Playback Volume",
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+		  SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+	.info = snd_echo_output_gain_info,
+	.get = snd_echo_output_gain_get,
+	.put = snd_echo_output_gain_put,
+	.tlv = {.p = db_scale_output_gain},
+};
+#else
 static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
 	.name = "PCM Playback Volume",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1012,9 +1025,10 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
 	.put = snd_echo_output_gain_put,
 	.tlv = {.p = db_scale_output_gain},
 };
-
 #endif
 
+#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */
+
 
 
 #ifdef ECHOCARD_HAS_INPUT_GAIN
@@ -2030,10 +2044,18 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
 	snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
 	if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
 		goto ctl_error;
-#else
-	if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0)
+#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&snd_echo_line_output_gain, chip));
+	if (err < 0)
 		goto ctl_error;
 #endif
+#else /* ECHOCARD_HAS_VMIXER */
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&snd_echo_pcm_output_gain, chip));
+	if (err < 0)
+		goto ctl_error;
+#endif /* ECHOCARD_HAS_VMIXER */
 
 #ifdef ECHOCARD_HAS_INPUT_GAIN
 	if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0)
diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c
index f3b9b45c9c1..f05c8c097aa 100644
--- a/sound/pci/echoaudio/mia.c
+++ b/sound/pci/echoaudio/mia.c
@@ -29,6 +29,7 @@
 #define ECHOCARD_HAS_ADAT	FALSE
 #define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
 #define ECHOCARD_HAS_MIDI
+#define ECHOCARD_HAS_LINE_OUT_GAIN
 
 /* Pipe indexes */
 #define PX_ANALOG_OUT	0	/* 8 */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 20a66f85f0a..c9ad182e1b4 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2303,6 +2303,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
  * white-list for enable_msi
  */
 static struct snd_pci_quirk msi_white_list[] __devinitdata = {
+	SND_PCI_QUIRK(0x103c, 0x30f7, "HP Pavilion dv4t-1300", 1),
 	SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1),
 	{}
 };
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 215e72a8711..2d603f6aba6 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -4032,6 +4032,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec)
 }
 
 /*
+ * HP Touchsmart
+ * port-A (0x11)      - front hp-out
+ * port-B (0x14)      - unused
+ * port-C (0x15)      - unused
+ * port-D (0x12)      - rear line out
+ * port-E (0x1c)      - front mic-in
+ * port-F (0x16)      - Internal speakers
+ * digital-mic (0x17) - Internal mic
+ */
+
+static struct hda_verb ad1984a_touchsmart_verbs[] = {
+	/* DACs; unmute as default */
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+	/* Port-A (HP) mixer - route only from analog mixer */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* Port-A pin */
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Port-A (HP) pin - always unmuted */
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Port-E (int speaker) mixer - route only from analog mixer */
+	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
+	/* Port-E pin */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	/* Port-F (int speaker) mixer - route only from analog mixer */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* Port-F pin */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Analog mixer; mute as default */
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+	/* Analog Mix output amp */
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* capture sources */
+	/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* unsolicited event for pin-sense */
+	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+	/* allow to touch GPIO1 (for mute control) */
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
+	/* internal mic - dmic */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* set magic COEFs for dmic */
+	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
+	{ } /* end */
+};
+
+static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+/*	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_amp_switch_get,
+		.put = ad1884a_mobile_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+	},
+	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+/* switch to external mic if plugged */
+static void ad1984a_touchsmart_automic(struct hda_codec *codec)
+{
+	if (snd_hda_codec_read(codec, 0x1c, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) {
+		snd_hda_codec_write(codec, 0x0c, 0,
+				     AC_VERB_SET_CONNECT_SEL, 0x4);
+	} else {
+		snd_hda_codec_write(codec, 0x0c, 0,
+				     AC_VERB_SET_CONNECT_SEL, 0x5);
+	}
+}
+
+
+/* unsolicited event for HP jack sensing */
+static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
+	unsigned int res)
+{
+	switch (res >> 26) {
+	case AD1884A_HP_EVENT:
+		ad1884a_hp_automute(codec);
+		break;
+	case AD1884A_MIC_EVENT:
+		ad1984a_touchsmart_automic(codec);
+		break;
+	}
+}
+
+/* initialize jack-sensing, too */
+static int ad1984a_touchsmart_init(struct hda_codec *codec)
+{
+	ad198x_init(codec);
+	ad1884a_hp_automute(codec);
+	ad1984a_touchsmart_automic(codec);
+	return 0;
+}
+
+
+/*
  */
 
 enum {
@@ -4039,6 +4160,7 @@ enum {
 	AD1884A_LAPTOP,
 	AD1884A_MOBILE,
 	AD1884A_THINKPAD,
+	AD1984A_TOUCHSMART,
 	AD1884A_MODELS
 };
 
@@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
 	[AD1884A_LAPTOP]	= "laptop",
 	[AD1884A_MOBILE]	= "mobile",
 	[AD1884A_THINKPAD]	= "thinkpad",
+	[AD1984A_TOUCHSMART]	= "touchsmart",
 };
 
 static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
@@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
 	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
+	SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
 	{}
 };
 
@@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec)
 		codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
 		codec->patch_ops.init = ad1984a_thinkpad_init;
 		break;
+	case AD1984A_TOUCHSMART:
+		spec->mixers[0] = ad1984a_touchsmart_mixers;
+		spec->init_verbs[0] = ad1984a_touchsmart_verbs;
+		spec->multiout.dig_out_nid = 0;
+		codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
+		codec->patch_ops.init = ad1984a_touchsmart_init;
+		/* set the upper-limit for mixer amp to 0dB for avoiding the
+		 * possible damage by overloading
+		 */
+		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
+					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+					  (1 << AC_AMPCAP_MUTE_SHIFT));
+		break;
 	}
 
 	return 0;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9d899eda44d..3fbbc8c01e7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -682,11 +682,13 @@ static struct hda_input_mux cxt5045_capture_source = {
 };
 
 static struct hda_input_mux cxt5045_capture_source_benq = {
-	.num_items = 3,
+	.num_items = 5,
 	.items = {
 		{ "IntMic", 0x1 },
 		{ "ExtMic", 0x2 },
 		{ "LineIn", 0x3 },
+		{ "CD",     0x4 },
+		{ "Mixer",  0x0 },
 	}
 };
 
@@ -811,11 +813,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = {
 };
 
 static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
+	HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
+
 	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
 	HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
 
+	HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
+
 	{}
 };
 
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index c8435c9a97f..9fb60276f5c 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -29,6 +29,9 @@
 #include "hda_codec.h"
 #include "hda_local.h"
 
+/* define below to restrict the supported rates and formats */
+/* #define LIMITED_RATE_FMT_SUPPORT */
+
 struct nvhdmi_spec {
 	struct hda_multi_out multiout;
 
@@ -60,6 +63,22 @@ static struct hda_verb nvhdmi_basic_init[] = {
 	{} /* terminator */
 };
 
+#ifdef LIMITED_RATE_FMT_SUPPORT
+/* support only the safe format and rate */
+#define SUPPORTED_RATES		SNDRV_PCM_RATE_48000
+#define SUPPORTED_MAXBPS	16
+#define SUPPORTED_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
+#else
+/* support all rates and formats */
+#define SUPPORTED_RATES \
+	(SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+	SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\
+	 SNDRV_PCM_RATE_192000)
+#define SUPPORTED_MAXBPS	24
+#define SUPPORTED_FORMATS \
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+#endif
+
 /*
  * Controls
  */
@@ -258,9 +277,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = {
 	.channels_min = 2,
 	.channels_max = 8,
 	.nid = Nv_Master_Convert_nid,
-	.rates = SNDRV_PCM_RATE_48000,
-	.maxbps = 16,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.rates = SUPPORTED_RATES,
+	.maxbps = SUPPORTED_MAXBPS,
+	.formats = SUPPORTED_FORMATS,
 	.ops = {
 		.open = nvhdmi_dig_playback_pcm_open,
 		.close = nvhdmi_dig_playback_pcm_close_8ch,
@@ -273,9 +292,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.nid = Nv_Master_Convert_nid,
-	.rates = SNDRV_PCM_RATE_48000,
-	.maxbps = 16,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.rates = SUPPORTED_RATES,
+	.maxbps = SUPPORTED_MAXBPS,
+	.formats = SUPPORTED_FORMATS,
 	.ops = {
 		.open = nvhdmi_dig_playback_pcm_open,
 		.close = nvhdmi_dig_playback_pcm_close_2ch,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7ed47f66ddd..c08ca660dab 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,7 +275,7 @@ struct alc_spec {
 	struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 
-	const struct hda_verb *init_verbs[5];	/* initialization verbs
+	const struct hda_verb *init_verbs[10];	/* initialization verbs
 						 * don't forget NULL
 						 * termination!
 						 */
@@ -1332,15 +1332,20 @@ do_sku:
 	 *	        when the external headphone out jack is plugged"
 	 */
 	if (!spec->autocfg.hp_pins[0]) {
+		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
-			spec->autocfg.hp_pins[0] = porta;
+			nid = porta;
 		else if (tmp == 1)
-			spec->autocfg.hp_pins[0] = porte;
+			nid = porte;
 		else if (tmp == 2)
-			spec->autocfg.hp_pins[0] = portd;
+			nid = portd;
 		else
 			return 1;
+		for (i = 0; i < spec->autocfg.line_outs; i++)
+			if (spec->autocfg.line_out_pins[i] == nid)
+				return 1;
+		spec->autocfg.hp_pins[0] = nid;
 	}
 
 	alc_init_auto_hp(codec);
@@ -1362,7 +1367,7 @@ static void alc_ssid_check(struct hda_codec *codec,
 }
 
 /*
- * Fix-up pin default configurations
+ * Fix-up pin default configurations and add default verbs
  */
 
 struct alc_pincfg {
@@ -1370,9 +1375,14 @@ struct alc_pincfg {
 	u32 val;
 };
 
-static void alc_fix_pincfg(struct hda_codec *codec,
+struct alc_fixup {
+	const struct alc_pincfg *pins;
+	const struct hda_verb *verbs;
+};
+
+static void alc_pick_fixup(struct hda_codec *codec,
 			   const struct snd_pci_quirk *quirk,
-			   const struct alc_pincfg **pinfix)
+			   const struct alc_fixup *fix)
 {
 	const struct alc_pincfg *cfg;
 
@@ -1380,9 +1390,14 @@ static void alc_fix_pincfg(struct hda_codec *codec,
 	if (!quirk)
 		return;
 
-	cfg = pinfix[quirk->value];
-	for (; cfg->nid; cfg++)
-		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+	fix += quirk->value;
+	cfg = fix->pins;
+	if (cfg) {
+		for (; cfg->nid; cfg++)
+			snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+	}
+	if (fix->verbs)
+		add_verb(codec->spec, fix->verbs);
 }
 
 /*
@@ -7927,8 +7942,9 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 
 static struct snd_kcontrol_new alc883_targa_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
@@ -7947,8 +7963,9 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = {
 
 static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -7960,6 +7977,15 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = {
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -9167,7 +9193,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.init_hook = alc882_targa_automute,
 	},
 	[ALC883_TARGA_8ch_DIG] = {
-		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
+		.mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer,
+			    alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
 				alc883_targa_verbs },
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
@@ -9581,11 +9608,13 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
 	{ }
 };
 
-static const struct alc_pincfg *alc882_pin_fixes[] = {
-	[PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
+static const struct alc_fixup alc882_fixups[] = {
+	[PINFIX_ABIT_AW9D_MAX] = {
+		.pins = alc882_abit_aw9d_pinfix
+	},
 };
 
-static struct snd_pci_quirk alc882_pinfix_tbl[] = {
+static struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
 	{}
 };
@@ -9857,7 +9886,7 @@ static int patch_alc882(struct hda_codec *codec)
 		board_config = ALC882_AUTO;
 	}
 
-	alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
+	alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups);
 
 	if (board_config == ALC882_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -12648,7 +12677,7 @@ static struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc268_toshiba_automute,
 	},
 	[ALC268_ACER] = {
-		.mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer,
+		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
 			    alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
@@ -12830,12 +12859,15 @@ static int patch_alc268(struct hda_codec *codec)
 		unsigned int wcap = get_wcaps(codec, 0x07);
 		int i;
 
+		spec->capsrc_nids = alc268_capsrc_nids;
 		/* get type */
 		wcap = get_wcaps_type(wcap);
 		if (spec->auto_mic ||
 		    wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
 			spec->adc_nids = alc268_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
+			if (spec->auto_mic)
+				fixup_automic_adc(codec);
 			if (spec->auto_mic || spec->input_mux->num_items == 1)
 				add_mixer(spec, alc268_capture_nosrc_mixer);
 			else
@@ -12845,7 +12877,6 @@ static int patch_alc268(struct hda_codec *codec)
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
 			add_mixer(spec, alc268_capture_mixer);
 		}
-		spec->capsrc_nids = alc268_capsrc_nids;
 		/* set default input source */
 		for (i = 0; i < spec->num_adc_nids; i++)
 			snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
@@ -13370,7 +13401,8 @@ static const char *alc269_models[ALC269_MODEL_LAST] = {
 	[ALC269_ASUS_EEEPC_P703]	= "eeepc-p703",
 	[ALC269_ASUS_EEEPC_P901]	= "eeepc-p901",
 	[ALC269_FUJITSU]		= "fujitsu",
-	[ALC269_LIFEBOOK]		= "lifebook"
+	[ALC269_LIFEBOOK]		= "lifebook",
+	[ALC269_AUTO]			= "auto",
 };
 
 static struct snd_pci_quirk alc269_cfg_tbl[] = {
@@ -14344,15 +14376,16 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec)
 static void alc861_auto_init_hp_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
 
-	pin = spec->autocfg.hp_pins[0];
-	if (pin)
-		alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
+	if (spec->autocfg.hp_outs)
+		alc861_auto_set_output_and_unmute(codec,
+						  spec->autocfg.hp_pins[0],
+						  PIN_HP,
 						  spec->multiout.hp_nid);
-	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+	if (spec->autocfg.speaker_outs)
+		alc861_auto_set_output_and_unmute(codec,
+						  spec->autocfg.speaker_pins[0],
+						  PIN_OUT,
 						  spec->multiout.dac_nids[0]);
 }
 
@@ -15145,7 +15178,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
 	SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
 	SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
-	SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
+	/*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
 	SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
 	SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
@@ -15538,6 +15571,29 @@ static void alc861vd_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
+enum {
+	ALC660VD_FIX_ASUS_GPIO1
+};
+
+/* reset GPIO1 */
+static const struct hda_verb alc660vd_fix_asus_gpio1_verbs[] = {
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+	{ }
+};
+
+static const struct alc_fixup alc861vd_fixups[] = {
+	[ALC660VD_FIX_ASUS_GPIO1] = {
+		.verbs = alc660vd_fix_asus_gpio1_verbs,
+	},
+};
+
+static struct snd_pci_quirk alc861vd_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
+	{}
+};
+
 static int patch_alc861vd(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
@@ -15559,6 +15615,8 @@ static int patch_alc861vd(struct hda_codec *codec)
 		board_config = ALC861VD_AUTO;
 	}
 
+	alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups);
+
 	if (board_config == ALC861VD_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc861vd_parse_auto_config(codec);
@@ -16839,6 +16897,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
 	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
 		      ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
 	SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
 		      ALC662_3ST_6ch_DIG),
@@ -17132,70 +17191,145 @@ static struct alc_config_preset alc662_presets[] = {
  * BIOS auto configuration
  */
 
+/* convert from MIX nid to DAC */
+static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
+{
+	if (nid == 0x0f)
+		return 0x02;
+	else if (nid >= 0x0c && nid <= 0x0e)
+		return nid - 0x0c + 0x02;
+	else
+		return 0;
+}
+
+/* get MIX nid connected to the given pin targeted to DAC */
+static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
+				   hda_nid_t dac)
+{
+	hda_nid_t mix[4];
+	int i, num;
+
+	num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
+	for (i = 0; i < num; i++) {
+		if (alc662_mix_to_dac(mix[i]) == dac)
+			return mix[i];
+	}
+	return 0;
+}
+
+/* look for an empty DAC slot */
+static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t srcs[5];
+	int i, j, num;
+
+	num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs));
+	if (num < 0)
+		return 0;
+	for (i = 0; i < num; i++) {
+		hda_nid_t nid = alc662_mix_to_dac(srcs[i]);
+		if (!nid)
+			continue;
+		for (j = 0; j < spec->multiout.num_dacs; j++)
+			if (spec->multiout.dac_nids[j] == nid)
+				break;
+		if (j >= spec->multiout.num_dacs)
+			return nid;
+	}
+	return 0;
+}
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int alc662_auto_fill_dac_nids(struct hda_codec *codec,
+				     const struct auto_pin_cfg *cfg)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+	hda_nid_t dac;
+
+	spec->multiout.dac_nids = spec->private_dac_nids;
+	for (i = 0; i < cfg->line_outs; i++) {
+		dac = alc662_look_for_dac(codec, cfg->line_out_pins[i]);
+		if (!dac)
+			continue;
+		spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+	}
+	return 0;
+}
+
+static int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx,
+			      hda_nid_t nid, unsigned int chs)
+{
+	char name[32];
+	sprintf(name, "%s Playback Volume", pfx);
+	return add_control(spec, ALC_CTL_WIDGET_VOL, name,
+			   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+}
+
+static int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx,
+			     hda_nid_t nid, unsigned int chs)
+{
+	char name[32];
+	sprintf(name, "%s Playback Switch", pfx);
+	return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+			   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT));
+}
+
+#define alc662_add_stereo_vol(spec, pfx, nid) \
+	alc662_add_vol_ctl(spec, pfx, nid, 3)
+#define alc662_add_stereo_sw(spec, pfx, nid) \
+	alc662_add_sw_ctl(spec, pfx, nid, 3)
+
 /* add playback controls from the parsed DAC table */
-static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
+static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
 					     const struct auto_pin_cfg *cfg)
 {
-	char name[32];
+	struct alc_spec *spec = codec->spec;
 	static const char *chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
-	hda_nid_t nid;
+	hda_nid_t nid, mix;
 	int i, err;
 
 	for (i = 0; i < cfg->line_outs; i++) {
-		if (!spec->multiout.dac_nids[i])
+		nid = spec->multiout.dac_nids[i];
+		if (!nid)
+			continue;
+		mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid);
+		if (!mix)
 			continue;
-		nid = alc880_idx_to_dac(i);
 		if (i == 2) {
 			/* Center/LFE */
-			err = add_control(spec, ALC_CTL_WIDGET_VOL,
-					  "Center Playback Volume",
-					  HDA_COMPOSE_AMP_VAL(nid, 1, 0,
-							      HDA_OUTPUT));
+			err = alc662_add_vol_ctl(spec, "Center", nid, 1);
 			if (err < 0)
 				return err;
-			err = add_control(spec, ALC_CTL_WIDGET_VOL,
-					  "LFE Playback Volume",
-					  HDA_COMPOSE_AMP_VAL(nid, 2, 0,
-							      HDA_OUTPUT));
+			err = alc662_add_vol_ctl(spec, "LFE", nid, 2);
 			if (err < 0)
 				return err;
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Center Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(0x0e, 1, 0,
-							      HDA_INPUT));
+			err = alc662_add_sw_ctl(spec, "Center", mix, 1);
 			if (err < 0)
 				return err;
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "LFE Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
-							      HDA_INPUT));
+			err = alc662_add_sw_ctl(spec, "LFE", mix, 2);
 			if (err < 0)
 				return err;
 		} else {
 			const char *pfx;
 			if (cfg->line_outs == 1 &&
 			    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
-				if (!cfg->hp_pins)
+				if (cfg->hp_outs)
 					pfx = "Speaker";
 				else
 					pfx = "PCM";
 			} else
 				pfx = chname[i];
-			sprintf(name, "%s Playback Volume", pfx);
-			err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
-					  HDA_COMPOSE_AMP_VAL(nid, 3, 0,
-							      HDA_OUTPUT));
+			err = alc662_add_vol_ctl(spec, pfx, nid, 3);
 			if (err < 0)
 				return err;
 			if (cfg->line_outs == 1 &&
 			    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
 				pfx = "Speaker";
-			sprintf(name, "%s Playback Switch", pfx);
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
-				HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
-						    3, 0, HDA_INPUT));
+			err = alc662_add_sw_ctl(spec, pfx, mix, 3);
 			if (err < 0)
 				return err;
 		}
@@ -17204,54 +17338,38 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 /* add playback controls for speaker and HP outputs */
-static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
+/* return DAC nid if any new DAC is assigned */
+static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
 					const char *pfx)
 {
-	hda_nid_t nid;
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid, mix;
 	int err;
-	char name[32];
 
 	if (!pin)
 		return 0;
-
-	if (pin == 0x17) {
-		/* ALC663 has a mono output pin on 0x17 */
+	nid = alc662_look_for_dac(codec, pin);
+	if (!nid) {
+		char name[32];
+		/* the corresponding DAC is already occupied */
+		if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
+			return 0; /* no way */
+		/* create a switch only */
 		sprintf(name, "%s Playback Switch", pfx);
-		err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
-				  HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT));
-		return err;
+		return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+				   HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
 	}
 
-	if (alc880_is_fixed_pin(pin)) {
-		nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
-		/* printk(KERN_DEBUG "DAC nid=%x\n",nid); */
-		/* specify the DAC as the extra output */
-		if (!spec->multiout.hp_nid)
-			spec->multiout.hp_nid = nid;
-		else
-			spec->multiout.extra_out_nid[0] = nid;
-		/* control HP volume/switch on the output mixer amp */
-		nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
-		sprintf(name, "%s Playback Volume", pfx);
-		err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
-				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
-		if (err < 0)
-			return err;
-		sprintf(name, "%s Playback Switch", pfx);
-		err = add_control(spec, ALC_CTL_BIND_MUTE, name,
-				  HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
-		if (err < 0)
-			return err;
-	} else if (alc880_is_multi_pin(pin)) {
-		/* set manual connection */
-		/* we have only a switch on HP-out PIN */
-		sprintf(name, "%s Playback Switch", pfx);
-		err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
-				  HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
-		if (err < 0)
-			return err;
-	}
-	return 0;
+	mix = alc662_dac_to_mix(codec, pin, nid);
+	if (!mix)
+		return 0;
+	err = alc662_add_vol_ctl(spec, pfx, nid, 3);
+	if (err < 0)
+		return err;
+	err = alc662_add_sw_ctl(spec, pfx, mix, 3);
+	if (err < 0)
+		return err;
+	return nid;
 }
 
 /* create playback/capture controls for input pins */
@@ -17260,30 +17378,35 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 
 static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid, int pin_type,
-					      int dac_idx)
+					      hda_nid_t dac)
 {
+	int i, num;
+	hda_nid_t srcs[4];
+
 	alc_set_pin_output(codec, nid, pin_type);
 	/* need the manual connection? */
-	if (alc880_is_multi_pin(nid)) {
-		struct alc_spec *spec = codec->spec;
-		int idx = alc880_multi_pin_idx(nid);
-		snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
-				    AC_VERB_SET_CONNECT_SEL,
-				    alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
+	num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs));
+	if (num <= 1)
+		return;
+	for (i = 0; i < num; i++) {
+		if (alc662_mix_to_dac(srcs[i]) != dac)
+			continue;
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i);
+		return;
 	}
 }
 
 static void alc662_auto_init_multi_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
+	int pin_type = get_pin_type(spec->autocfg.line_out_type);
 	int i;
 
 	for (i = 0; i <= HDA_SIDE; i++) {
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
-		int pin_type = get_pin_type(spec->autocfg.line_out_type);
 		if (nid)
 			alc662_auto_set_output_and_unmute(codec, nid, pin_type,
-							  i);
+					spec->multiout.dac_nids[i]);
 	}
 }
 
@@ -17293,12 +17416,13 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec)
 	hda_nid_t pin;
 
 	pin = spec->autocfg.hp_pins[0];
-	if (pin) /* connect to front */
-		/* use dac 0 */
-		alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+	if (pin)
+		alc662_auto_set_output_and_unmute(codec, pin, PIN_HP,
+						  spec->multiout.hp_nid);
 	pin = spec->autocfg.speaker_pins[0];
 	if (pin)
-		alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+		alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+					spec->multiout.extra_out_nid[0]);
 }
 
 #define ALC662_PIN_CD_NID		ALC880_PIN_CD_NID
@@ -17336,21 +17460,25 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	if (!spec->autocfg.line_outs)
 		return 0; /* can't find valid BIOS pin config */
 
-	err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+	err = alc662_auto_fill_dac_nids(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	err = alc662_auto_create_multi_out_ctls(spec, &spec->autocfg);
+	err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	err = alc662_auto_create_extra_out(spec,
+	err = alc662_auto_create_extra_out(codec,
 					   spec->autocfg.speaker_pins[0],
 					   "Speaker");
 	if (err < 0)
 		return err;
-	err = alc662_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
+	if (err)
+		spec->multiout.extra_out_nid[0] = err;
+	err = alc662_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
 					   "Headphone");
 	if (err < 0)
 		return err;
+	if (err)
+		spec->multiout.hp_nid = err;
 	err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e31e53dc696..66c0876bf73 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -158,6 +158,7 @@ enum {
 	STAC_D965_5ST_NO_FP,
 	STAC_DELL_3ST,
 	STAC_DELL_BIOS,
+	STAC_927X_VOLKNOB,
 	STAC_927X_MODELS
 };
 
@@ -182,8 +183,8 @@ struct sigmatel_jack {
 
 struct sigmatel_mic_route {
 	hda_nid_t pin;
-	unsigned char mux_idx;
-	unsigned char dmux_idx;
+	signed char mux_idx;
+	signed char dmux_idx;
 };
 
 struct sigmatel_spec {
@@ -864,10 +865,6 @@ static struct hda_verb stac92hd73xx_core_init[] = {
 };
 
 static struct hda_verb stac92hd83xxx_core_init[] = {
-	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x0},
-
 	/* power state controls amps */
 	{ 0x01, AC_VERB_SET_EAPD, 1 << 2},
 	{}
@@ -911,6 +908,16 @@ static struct hda_verb d965_core_init[] = {
 	{}
 };
 
+static struct hda_verb dell_3st_core_init[] = {
+	/* don't set delta bit */
+	{0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
+	/* unmute node 0x1b */
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* select node 0x03 as DAC */
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{}
+};
+
 static struct hda_verb stac927x_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -919,6 +926,14 @@ static struct hda_verb stac927x_core_init[] = {
 	{}
 };
 
+static struct hda_verb stac927x_volknob_core_init[] = {
+	/* don't set delta bit */
+	{0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
+	/* enable analog pc beep path */
+	{0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
+	{}
+};
+
 static struct hda_verb stac9205_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -1590,8 +1605,8 @@ static unsigned int ref92hd83xxx_pin_configs[10] = {
 };
 
 static unsigned int dell_s14_pin_configs[10] = {
-	0x02214030, 0x02211010, 0x02a19020, 0x01014050,
-	0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160,
+	0x0221403f, 0x0221101f, 0x02a19020, 0x90170110,
+	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a60160,
 	0x40f000f0, 0x40f000f0,
 };
 
@@ -1690,6 +1705,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "HP mini 1000", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b,
 		      "HP HDX", STAC_HP_HDX),  /* HDX16 */
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620,
+		      "HP dv6", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010,
 		      "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
@@ -2001,6 +2018,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 	[STAC_D965_5ST_NO_FP]  = d965_5st_no_fp_pin_configs,
 	[STAC_DELL_3ST]  = dell_3st_pin_configs,
 	[STAC_DELL_BIOS] = NULL,
+	[STAC_927X_VOLKNOB] = NULL,
 };
 
 static const char *stac927x_models[STAC_927X_MODELS] = {
@@ -2012,6 +2030,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = {
 	[STAC_D965_5ST_NO_FP]	= "5stack-no-fp",
 	[STAC_DELL_3ST]		= "dell-3stack",
 	[STAC_DELL_BIOS]	= "dell-bios",
+	[STAC_927X_VOLKNOB]	= "volknob",
 };
 
 static struct snd_pci_quirk stac927x_cfg_tbl[] = {
@@ -2047,6 +2066,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 			   "Intel D965", STAC_D965_5ST),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500,
 			   "Intel D965", STAC_D965_5ST),
+	/* volume-knob fixes */
+	SND_PCI_QUIRK_VENDOR(0x10cf, "FSC", STAC_927X_VOLKNOB),
 	{} /* terminator */
 };
 
@@ -3471,18 +3492,26 @@ static int set_mic_route(struct hda_codec *codec,
 			break;
 	if (i <= AUTO_PIN_FRONT_MIC) {
 		/* analog pin */
-		mic->dmux_idx = 0;
 		i = get_connection_index(codec, spec->mux_nids[0], pin);
 		if (i < 0)
 			return -1;
 		mic->mux_idx = i;
+		mic->dmux_idx = -1;
+		if (spec->dmux_nids)
+			mic->dmux_idx = get_connection_index(codec,
+							     spec->dmux_nids[0],
+							     spec->mux_nids[0]);
 	}  else if (spec->dmux_nids) {
 		/* digital pin */
-		mic->mux_idx = 0;
 		i = get_connection_index(codec, spec->dmux_nids[0], pin);
 		if (i < 0)
 			return -1;
 		mic->dmux_idx = i;
+		mic->mux_idx = -1;
+		if (spec->mux_nids)
+			mic->mux_idx = get_connection_index(codec,
+							    spec->mux_nids[0],
+							    spec->dmux_nids[0]);
 	}
 	return 0;
 }
@@ -4166,7 +4195,10 @@ static int stac92xx_init(struct hda_codec *codec)
 		stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0],
 				AC_PINCTL_OUT_EN);
 		/* fake event to set up pins */
-		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+		if (cfg->hp_pins[0])
+			stac_issue_unsol_event(codec, cfg->hp_pins[0]);
+		else if (cfg->line_out_pins[0])
+			stac_issue_unsol_event(codec, cfg->line_out_pins[0]);
 	} else {
 		stac92xx_auto_init_multi_out(codec);
 		stac92xx_auto_init_hp_out(codec);
@@ -4556,11 +4588,11 @@ static void stac92xx_mic_detect(struct hda_codec *codec)
 		mic = &spec->ext_mic;
 	else
 		mic = &spec->int_mic;
-	if (mic->dmux_idx)
+	if (mic->dmux_idx >= 0)
 		snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
 					  AC_VERB_SET_CONNECT_SEL,
 					  mic->dmux_idx);
-	else
+	if (mic->mux_idx >= 0)
 		snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0,
 					  AC_VERB_SET_CONNECT_SEL,
 					  mic->mux_idx);
@@ -4688,8 +4720,13 @@ static int stac92xx_resume(struct hda_codec *codec)
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
 	/* fake event to set up pins again to override cached values */
-	if (spec->hp_detect)
-		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+	if (spec->hp_detect) {
+		if (spec->autocfg.hp_pins[0])
+			stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+		else if (spec->autocfg.line_out_pins[0])
+			stac_issue_unsol_event(codec,
+					       spec->autocfg.line_out_pins[0]);
+	}
 	return 0;
 }
 
@@ -5016,7 +5053,7 @@ again:
 		spec->eapd_switch = 1;
 		break;
 	}
-	if (spec->board_config > STAC_92HD73XX_REF) {
+	if (spec->board_config != STAC_92HD73XX_REF) {
 		/* GPIO0 High = Enable EAPD */
 		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
 		spec->gpio_data = 0x01;
@@ -5066,7 +5103,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 
 	codec->spec = spec;
 	codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
-	spec->mono_nid = 0x19;
 	spec->digbeep_nid = 0x21;
 	spec->mux_nids = stac92hd83xxx_mux_nids;
 	spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids);
@@ -5242,7 +5278,7 @@ again:
 		stac92xx_set_config_regs(codec,
 				stac92hd71bxx_brd_tbl[spec->board_config]);
 
-	if (spec->board_config > STAC_92HD71BXX_REF) {
+	if (spec->board_config != STAC_92HD71BXX_REF) {
 		/* GPIO0 = EAPD */
 		spec->gpio_mask = 0x01;
 		spec->gpio_dir = 0x01;
@@ -5375,6 +5411,11 @@ again:
 	case STAC_HP_DV5:
 		snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
 		stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
+		/* HP dv6 gives the headphone pin as a line-out.  Thus we
+		 * need to set hp_detect flag here to force to enable HP
+		 * detection.
+		 */
+		spec->hp_detect = 1;
 		break;
 	case STAC_HP_HDX:
 		spec->num_dmics = 1;
@@ -5557,14 +5598,17 @@ static int patch_stac927x(struct hda_codec *codec)
 	spec->dac_list = stac927x_dac_nids;
 	spec->multiout.dac_nids = spec->dac_nids;
 
+	if (spec->board_config != STAC_D965_REF) {
+		/* GPIO0 High = Enable EAPD */
+		spec->eapd_mask = spec->gpio_mask = 0x01;
+		spec->gpio_dir = spec->gpio_data = 0x01;
+	}
+
 	switch (spec->board_config) {
 	case STAC_D965_3ST:
 	case STAC_D965_5ST:
 		/* GPIO0 High = Enable EAPD */
-		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01;
-		spec->gpio_data = 0x01;
 		spec->num_dmics = 0;
-
 		spec->init = d965_core_init;
 		break;
 	case STAC_DELL_BIOS:
@@ -5583,32 +5627,26 @@ static int patch_stac927x(struct hda_codec *codec)
 		snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130);
 		/* fallthru */
 	case STAC_DELL_3ST:
-		/* GPIO2 High = Enable EAPD */
-		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04;
-		spec->gpio_data = 0x04;
-		switch (codec->subsystem_id) {
-		case 0x1028022f:
-			/* correct EAPD to be GPIO0 */
-			spec->eapd_mask = spec->gpio_mask = 0x01;
-			spec->gpio_dir = spec->gpio_data = 0x01;
-			break;
-		};
+		if (codec->subsystem_id != 0x1028022f) {
+			/* GPIO2 High = Enable EAPD */
+			spec->eapd_mask = spec->gpio_mask = 0x04;
+			spec->gpio_dir = spec->gpio_data = 0x04;
+		}
 		spec->dmic_nids = stac927x_dmic_nids;
 		spec->num_dmics = STAC927X_NUM_DMICS;
 
-		spec->init = d965_core_init;
+		spec->init = dell_3st_core_init;
 		spec->dmux_nids = stac927x_dmux_nids;
 		spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
 		break;
+	case STAC_927X_VOLKNOB:
+		spec->num_dmics = 0;
+		spec->init = stac927x_volknob_core_init;
+		break;
 	default:
-		if (spec->board_config > STAC_D965_REF) {
-			/* GPIO0 High = Enable EAPD */
-			spec->eapd_mask = spec->gpio_mask = 0x01;
-			spec->gpio_dir = spec->gpio_data = 0x01;
-		}
 		spec->num_dmics = 0;
-
 		spec->init = stac927x_core_init;
+		break;
 	}
 
 	spec->num_caps = STAC927X_NUM_CAPS;
diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c
index 37564300b50..6da21a2bcad 100644
--- a/sound/pci/ice1712/amp.c
+++ b/sound/pci/ice1712/amp.c
@@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice)
 
 	/* only use basic functionality for now */
 
-	ice->num_total_dacs = 2;	/* only PSDOUT0 is connected */
+	/* VT1616 6ch codec connected to PSDOUT0 using packed mode */
+	ice->num_total_dacs = 6;
 	ice->num_total_adcs = 2;
 
-	/* Chaintech AV-710 has another codecs, which need initialization */
-	/* initialize WM8728 codec */
+	/* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4
+	   (shared with the SPDIF output). Mixer control for this codec
+	   is not yet supported. */
 	if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) {
 		for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2)
 			wm_put(ice, wm_inits[i], wm_inits[i+1]);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index cecf1ffeeaa..d74033a2cfb 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2259,7 +2259,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol,
 }
 
 static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = {
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Multi Track Peak",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info = snd_ice1712_pro_peak_info,
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index af6e0014862..10fc92c0557 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -648,7 +648,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
 	    (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) {
 		/* running? we cannot change the rate now... */
 		spin_unlock_irqrestore(&ice->reg_lock, flags);
-		return -EBUSY;
+		return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY;
 	}
 	if (!force && is_pro_rate_locked(ice)) {
 		spin_unlock_irqrestore(&ice->reg_lock, flags);
@@ -1294,7 +1294,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device)
 
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 					      snd_dma_pci_data(ice->pci),
-					      64*1024, 64*1024);
+					      256*1024, 256*1024);
 
 	ice->pcm = pcm;
 
@@ -1408,7 +1408,7 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device)
 
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 					      snd_dma_pci_data(ice->pci),
-					      64*1024, 64*1024);
+					      256*1024, 256*1024);
 
 	ice->pcm_ds = pcm;
 
@@ -2110,7 +2110,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol,
 }
 
 static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = {
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Multi Track Peak",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.info = snd_vt1724_pro_peak_info,
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 171ada53520..754867ed478 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1954,6 +1954,18 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 		.name = "Sony S1XP",
 		.type = AC97_TUNE_INV_EAPD
 	},
+	{
+		.subvendor = 0x104d,
+		.subdevice = 0x81c0,
+		.name = "Sony VAIO VGN-T350P", /*AD1981B*/
+		.type = AC97_TUNE_INV_EAPD
+	},
+	{
+		.subvendor = 0x104d,
+		.subdevice = 0x81c5,
+		.name = "Sony VAIO VGN-B1VP", /*AD1981B*/
+		.type = AC97_TUNE_INV_EAPD
+	},
  	{
 		.subvendor = 0x1043,
 		.subdevice = 0x80f3,
diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h
index 012c010c8c8..51afc048961 100644
--- a/sound/pci/lx6464es/lx6464es.h
+++ b/sound/pci/lx6464es/lx6464es.h
@@ -86,7 +86,6 @@ struct lx6464es {
 
 	/* messaging */
 	spinlock_t		msg_lock;          /* message spinlock */
-	atomic_t	        send_message_locked;
 	struct lx_rmh           rmh;
 
 	/* configuration */
@@ -95,7 +94,6 @@ struct lx6464es {
 	uint                    hardware_running[2];
 	u32                     board_sample_rate; /* sample rate read from
 						    * board */
-	u32                     sample_rate;	   /* our sample rate */
 	u16                     pcm_granularity;   /* board blocksize */
 
 	/* dma */
diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
index 5812780d6e8..3086b751da4 100644
--- a/sound/pci/lx6464es/lx_core.c
+++ b/sound/pci/lx6464es/lx_core.c
@@ -314,98 +314,6 @@ static inline void lx_message_dump(struct lx_rmh *rmh)
 #define XILINX_POLL_NO_SLEEP    100
 #define XILINX_POLL_ITERATIONS  150
 
-#if 0 /* not used now */
-static int lx_message_send(struct lx6464es *chip, struct lx_rmh *rmh)
-{
-	u32 reg = ED_DSP_TIMED_OUT;
-	int dwloop;
-	int answer_received;
-
-	if (lx_dsp_reg_read(chip, eReg_CSM) & (Reg_CSM_MC | Reg_CSM_MR)) {
-		snd_printk(KERN_ERR LXP "PIOSendMessage eReg_CSM %x\n", reg);
-		return -EBUSY;
-	}
-
-	/* write command */
-	lx_dsp_reg_writebuf(chip, eReg_CRM1, rmh->cmd, rmh->cmd_len);
-
-	snd_BUG_ON(atomic_read(&chip->send_message_locked) != 0);
-	atomic_set(&chip->send_message_locked, 1);
-
-	/* MicoBlaze gogogo */
-	lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC);
-
-	/* wait for interrupt to answer */
-	for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS; ++dwloop) {
-		answer_received = atomic_read(&chip->send_message_locked);
-		if (answer_received == 0)
-			break;
-		msleep(1);
-	}
-
-	if (answer_received == 0) {
-		/* in Debug mode verify Reg_CSM_MR */
-		snd_BUG_ON(!(lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR));
-
-		/* command finished, read status */
-		if (rmh->dsp_stat == 0)
-			reg = lx_dsp_reg_read(chip, eReg_CRM1);
-		else
-			reg = 0;
-	} else {
-		int i;
-		snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! "
-			   "Interrupts disabled?\n");
-
-		/* attente bit Reg_CSM_MR */
-		for (i = 0; i != XILINX_POLL_ITERATIONS; i++) {
-			if ((lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR)) {
-				if (rmh->dsp_stat == 0)
-					reg = lx_dsp_reg_read(chip, eReg_CRM1);
-				else
-					reg = 0;
-				goto polling_successful;
-			}
-
-			if (i > XILINX_POLL_NO_SLEEP)
-				msleep(1);
-		}
-		snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! "
-			   "polling failed\n");
-
-polling_successful:
-		atomic_set(&chip->send_message_locked, 0);
-	}
-
-	if ((reg & ERROR_VALUE) == 0) {
-		/* read response */
-		if (rmh->stat_len) {
-			snd_BUG_ON(rmh->stat_len >= (REG_CRM_NUMBER-1));
-
-			lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat,
-					   rmh->stat_len);
-		}
-	} else
-		snd_printk(KERN_WARNING LXP "lx_message_send: error_value %x\n",
-			   reg);
-
-	/* clear Reg_CSM_MR */
-	lx_dsp_reg_write(chip, eReg_CSM, 0);
-
-	switch (reg) {
-	case ED_DSP_TIMED_OUT:
-		snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n");
-		return -ETIMEDOUT;
-
-	case ED_DSP_CRASHED:
-		snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n");
-		return -EAGAIN;
-	}
-
-	lx_message_dump(rmh);
-	return 0;
-}
-#endif /* not used now */
 
 static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh)
 {
@@ -423,7 +331,7 @@ static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh)
 	/* MicoBlaze gogogo */
 	lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC);
 
-	/* wait for interrupt to answer */
+	/* wait for device to answer */
 	for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS * 1000; ++dwloop) {
 		if (lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR) {
 			if (rmh->dsp_stat == 0)
@@ -1175,10 +1083,6 @@ static int lx_interrupt_ack(struct lx6464es *chip, u32 *r_irqsrc,
 		*r_async_escmd = 1;
 	}
 
-	if (irqsrc & MASK_SYS_STATUS_CMD_DONE)
-		/* xilinx command notification */
-		atomic_set(&chip->send_message_locked, 0);
-
 	if (irq_async) {
 		/* snd_printd("interrupt: async event pending\n"); */
 		*r_async_pending = 1;
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index acfa4760da4..91683a34903 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1626,7 +1626,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct via82xx *chip = snd_kcontrol_chip(kcontrol);
-	unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+	unsigned int idx = kcontrol->id.subdevice;
 
 	ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0];
 	ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1];
@@ -1646,7 +1646,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct via82xx *chip = snd_kcontrol_chip(kcontrol);
-	unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+	unsigned int idx = kcontrol->id.subdevice;
 	unsigned long port = chip->port + 0x10 * idx;
 	unsigned char val;
 	int i, change = 0;
@@ -1705,11 +1705,12 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata =
 };
 
 static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
-	.name = "VIA DXS Playback Volume",
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.device = 0,
+	/* .subdevice set later */
+	.name = "PCM Playback Volume",
 	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
 		   SNDRV_CTL_ELEM_ACCESS_TLV_READ),
-	.count = 4,
 	.info = snd_via8233_dxs_volume_info,
 	.get = snd_via8233_dxs_volume_get,
 	.put = snd_via8233_dxs_volume_put,
@@ -1936,10 +1937,18 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip)
 		}
 		else /* Using DXS when PCM emulation is enabled is really weird */
 		{
-			/* Standalone DXS controls */
-			err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip));
-			if (err < 0)
-				return err;
+			for (i = 0; i < 4; ++i) {
+				struct snd_kcontrol *kctl;
+
+				kctl = snd_ctl_new1(
+					&snd_via8233_dxs_volume_control, chip);
+				if (!kctl)
+					return -ENOMEM;
+				kctl->id.subdevice = i;
+				err = snd_ctl_add(chip->card, kctl);
+				if (err < 0)
+					return err;
+			}
 		}
 	}
 	/* select spdif data slot 10/11 */
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 835fa19ed46..d06f780bd7e 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -59,6 +59,18 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
 	strlcpy(info.type, "keywest", I2C_NAME_SIZE);
 	info.addr = keywest_ctx->addr;
 	keywest_ctx->client = i2c_new_device(adapter, &info);
+	if (!keywest_ctx->client)
+		return -ENODEV;
+	/*
+	 * We know the driver is already loaded, so the device should be
+	 * already bound. If not it means binding failed, and then there
+	 * is no point in keeping the device instantiated.
+	 */
+	if (!keywest_ctx->client->driver) {
+		i2c_unregister_device(keywest_ctx->client);
+		keywest_ctx->client = NULL;
+		return -ENODEV;
+	}
 	
 	/*
 	 * Let i2c-core delete that device on driver removal.
@@ -86,7 +98,7 @@ static const struct i2c_device_id keywest_i2c_id[] = {
 	{ }
 };
 
-struct i2c_driver keywest_driver = {
+static struct i2c_driver keywest_driver = {
 	.driver = {
 		.name = "PMac Keywest Audio",
 	},
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ac927ffdc96..97f1a251e44 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -7,15 +7,6 @@ config SND_BF5XX_I2S
 	  mode (supports single stereo In/Out).
 	  You will also need to select the audio interfaces to support below.
 
-config SND_BF5XX_TDM
-	tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
-	depends on (BLACKFIN && SND_SOC)
-	help
-	  Say Y or M if you want to add support for codecs attached to
-	  the Blackfin SPORT (synchronous serial ports) interface in TDM
-	  mode.
-	  You will also need to select the audio interfaces to support below.
-
 config SND_BF5XX_SOC_SSM2602
 	tristate "SoC SSM2602 Audio support for BF52x ezkit"
 	depends on SND_BF5XX_I2S
@@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE
 	  Enter the GPIO used to control AD73311's SE pin. Acceptable
 	  values are 0 to 7
 
+config SND_BF5XX_TDM
+	tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
+	depends on (BLACKFIN && SND_SOC)
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the Blackfin SPORT (synchronous serial ports) interface in TDM
+	  mode.
+	  You will also need to select the audio interfaces to support below.
+
+config SND_BF5XX_SOC_AD1836
+	tristate "SoC AD1836 Audio support for BF5xx"
+	depends on SND_BF5XX_TDM
+	select SND_BF5XX_SOC_TDM
+	select SND_SOC_AD1836
+	help
+	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
+config SND_BF5XX_SOC_AD1938
+	tristate "SoC AD1938 Audio support for Blackfin"
+	depends on SND_BF5XX_TDM
+	select SND_BF5XX_SOC_TDM
+	select SND_SOC_AD1938
+	help
+	  Say Y if you want to add support for AD1938 codec on Blackfin.
+
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
 	depends on BLACKFIN
@@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT
 	  Say y if you want AC97 driver to support up to 5.1 channel audio.
 	  this mode will consume much more memory for DMA.
 
+config SND_BF5XX_HAVE_COLD_RESET
+	bool "BOARD has COLD Reset GPIO"
+	depends on SND_BF5XX_AC97
+	default y if BFIN548_EZKIT
+	default n if !BFIN548_EZKIT
+
+config SND_BF5XX_RESET_GPIO_NUM
+	int "Set a GPIO for cold reset"
+	depends on SND_BF5XX_HAVE_COLD_RESET
+	range 0 159
+	default 19 if BFIN548_EZKIT
+	default 5 if BFIN537_STAMP
+	default 0
+	help
+	  Set the correct GPIO for RESET the sound chip.
+
+config SND_BF5XX_SOC_AD1980
+	tristate "SoC AD1980/1 Audio support for BF5xx"
+	depends on SND_BF5XX_AC97
+	select SND_BF5XX_SOC_AC97
+	select SND_SOC_AD1980
+	help
+	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
 config SND_BF5XX_SOC_SPORT
 	tristate
 
@@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97
 	select SND_SOC_AC97_BUS
 	select SND_BF5XX_SOC_SPORT
 
-config SND_BF5XX_SOC_AD1836
-	tristate "SoC AD1836 Audio support for BF5xx"
-	depends on SND_BF5XX_TDM
-	select SND_BF5XX_SOC_TDM
-	select SND_SOC_AD1836
-	help
-	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1980
-	tristate "SoC AD1980/1 Audio support for BF5xx"
-	depends on SND_BF5XX_AC97
-	select SND_BF5XX_SOC_AC97
-	select SND_SOC_AD1980
-	help
-	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1938
-        tristate "SoC AD1938 Audio support for Blackfin"
-        depends on SND_BF5XX_TDM
-        select SND_BF5XX_SOC_TDM
-        select SND_SOC_AD1938
-        help
-          Say Y if you want to add support for AD1938 codec on Blackfin.
-
 config SND_BF5XX_SPORT_NUM
 	int "Set a SPORT for Sound chip"
 	depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
@@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM
 	default 0
 	help
 	  Set the correct SPORT for sound chip.
-
-config SND_BF5XX_HAVE_COLD_RESET
-	bool "BOARD has COLD Reset GPIO"
-	depends on SND_BF5XX_AC97
-	default y if BFIN548_EZKIT
-	default n if !BFIN548_EZKIT
-
-config SND_BF5XX_RESET_GPIO_NUM
-	int "Set a GPIO for cold reset"
-	depends on SND_BF5XX_HAVE_COLD_RESET
-	range 0 159
-	default 19 if BFIN548_EZKIT
-	default 5 if BFIN537_STAMP
-	default 0
-	help
-	  Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 2758b9017a7..e6932297873 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -277,7 +277,11 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
 	if (!dai->active)
 		return 0;
 
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+	ret = sport_set_multichannel(sport, 16, 0x3FF, 1);
+#else
 	ret = sport_set_multichannel(sport, 16, 0x1F, 1);
+#endif
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		return -EBUSY;
@@ -334,7 +338,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev,
 		goto sport_err;
 	}
 	/*SPORT works in TDM mode to simulate AC97 transfers*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+	ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1);
+#else
 	ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
+#endif
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		ret = -EBUSY;
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 3f2a911fe0c..a1f97dd809d 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -1,5 +1,5 @@
 /*
- * linux/sound/arm/bf5xx-ac97.h
+ * sound/soc/blackfin/bf5xx-ac97.h
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 876abade27e..084b68884ad 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = {
  * TFS. When Port G is selected and EMAC then there is a conflict between
  * the PHY interrupt line and TFS.  Current settings prevent the conflict
  * by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
  */
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
 #define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
 #endif
 
 static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
@@ -227,7 +227,8 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct platform_device *pdev,
+			struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	peripheral_free_list(&sport_req[sport_num][0]);
@@ -236,36 +237,31 @@ static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
 #ifdef CONFIG_PM
 static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
 {
-	struct sport_device *sport =
-		(struct sport_device *)dai->private_data;
 
 	pr_debug("%s : sport %d\n", __func__, dai->id);
-	if (!dai->active)
-		return 0;
+
 	if (dai->capture.active)
-		sport_rx_stop(sport);
+		sport_rx_stop(sport_handle);
 	if (dai->playback.active)
-		sport_tx_stop(sport);
+		sport_tx_stop(sport_handle);
 	return 0;
 }
 
 static int bf5xx_i2s_resume(struct snd_soc_dai *dai)
 {
 	int ret;
-	struct sport_device *sport =
-		(struct sport_device *)dai->private_data;
 
 	pr_debug("%s : sport %d\n", __func__, dai->id);
-	if (!dai->active)
-		return 0;
 
-	ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0);
+	ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1,
+				      bf5xx_i2s.rcr2, 0, 0);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		return -EBUSY;
 	}
 
-	ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0);
+	ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1,
+				      bf5xx_i2s.tcr2, 0, 0);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		return -EBUSY;
diff --git a/sound/soc/blackfin/bf5xx-i2s.h b/sound/soc/blackfin/bf5xx-i2s.h
index 7107d1a0b06..264ecdcba35 100644
--- a/sound/soc/blackfin/bf5xx-i2s.h
+++ b/sound/soc/blackfin/bf5xx-i2s.h
@@ -1,5 +1,5 @@
 /*
- * linux/sound/arm/bf5xx-i2s.h
+ * sound/soc/blackfin/bf5xx-i2s.h
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
index 469ce7fab20..99051ff0954 100644
--- a/sound/soc/blackfin/bf5xx-sport.c
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -326,7 +326,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport)
 
 int sport_tx_start(struct sport_device *sport)
 {
-	unsigned flags;
+	unsigned long flags;
 	pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__,
 			sport->tx_run, sport->rx_run);
 	if (sport->tx_run)
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 3096badf09a..ff546e91a22 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = {
  * TFS. When Port G is selected and EMAC then there is a conflict between
  * the PHY interrupt line and TFS.  Current settings prevent the conflict
  * by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
  */
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
 #define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
 #endif
 
 static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 3612bb92df9..c48485f2c55 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -18,7 +18,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
@@ -252,8 +251,7 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi)
 
 static struct spi_driver ad1836_spi_driver = {
 	.driver = {
-		.name	= "ad1836-spi",
-		.bus	= &spi_bus_type,
+		.name	= "ad1836",
 		.owner	= THIS_MODULE,
 	},
 	.probe		= ad1836_spi_probe,
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
index e62b27701a4..34b30efc3cb 100644
--- a/sound/soc/codecs/ad1938.c
+++ b/sound/soc/codecs/ad1938.c
@@ -28,7 +28,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
@@ -457,7 +456,6 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi)
 static struct spi_driver ad1938_spi_driver = {
 	.driver = {
 		.name	= "ad1938",
-		.bus	= &spi_bus_type,
 		.owner	= THIS_MODULE,
 	},
 	.probe		= ad1938_spi_probe,
@@ -516,6 +514,7 @@ static int ad1938_register(struct ad1938_priv *ad1938)
 	codec->num_dai = 1;
 	codec->write = ad1938_write_reg;
 	codec->read = ad1938_read_reg_cache;
+	codec->set_bias_level = ad1938_set_bias_level;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3ff0373dff8..593d5b9c9f0 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -579,7 +579,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
 	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
 			    WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
 	SOC_DAPM_SINGLE("PGA Capture Switch",
-			WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+			WM8350_LEFT_INPUT_VOLUME, 14, 1, 1),
 };
 
 /* Right Input Mixer */
@@ -589,7 +589,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
 	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
 			    WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
 	SOC_DAPM_SINGLE("PGA Capture Switch",
-			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1),
 };
 
 /* Left Mic Mixer */
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d80d414cfbb..5ad677ce80d 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -595,6 +595,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* Mono Capture mixer-mux */
 	{"Capture Right Mixer", "Stereo", "Capture Right Mux"},
+	{"Capture Left Mixer", "Stereo", "Capture Left Mux"},
 	{"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"},
 	{"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"},
 	{"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index da97aae475a..1ef2454c520 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940,
 	codec->reg_cache = &wm8940->reg_cache;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
-	if (ret == 0) {
+	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
 	}
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index d8a013ab317..98d663afc97 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -12,7 +12,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index c64e55aa63b..686e5aa9720 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1027,7 +1027,7 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream,
 		       - wm9081->fs);
 	for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
 		cur_val = abs((wm9081->sysclk_rate /
-			       clk_sys_rates[i].ratio) - wm9081->fs);;
+			       clk_sys_rates[i].ratio) - wm9081->fs);
 		if (cur_val < best_val) {
 			best = i;
 			best_val = cur_val;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 12a6c549ee6..4ae70704802 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -97,22 +97,19 @@ enum {
 	DAVINCI_MCBSP_WORD_32,
 };
 
-static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
-	.name = "I2S PCM Stereo out",
-};
-
-static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
-	.name = "I2S PCM Stereo in",
-};
-
 struct davinci_mcbsp_dev {
+	/*
+	 * dma_params must be first because rtd->dai->cpu_dai->private_data
+	 * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+	 * davinci_pcm_open.
+	 */
+	struct davinci_pcm_dma_params	dma_params[2];
 	void __iomem			*base;
 #define MOD_DSP_A	0
 #define MOD_DSP_B	1
 	int				mode;
 	u32				pcr;
 	struct clk			*clk;
-	struct davinci_pcm_dma_params	*dma_params[2];
 };
 
 static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
 	toggle_clock(dev, playback);
 }
 
-static int davinci_i2s_startup(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *cpu_dai)
-{
-	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
-	cpu_dai->dma_data = dev->dma_params[substream->stream];
-	return 0;
-}
-
 #define DEFAULT_BITPERSAMPLE	16
 
 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -353,8 +342,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
-	struct davinci_pcm_dma_params *dma_params = dai->dma_data;
 	struct davinci_mcbsp_dev *dev = dai->private_data;
+	struct davinci_pcm_dma_params *dma_params =
+					&dev->dma_params[substream->stream];
 	struct snd_interval *i = NULL;
 	int mcbsp_word_length;
 	unsigned int rcr, xcr, srgr;
@@ -472,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
 static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
-	.startup 	= davinci_i2s_startup,
 	.shutdown	= davinci_i2s_shutdown,
 	.prepare	= davinci_i2s_prepare,
 	.trigger	= davinci_i2s_trigger,
@@ -534,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 
 	dev->base = (void __iomem *)IO_ADDRESS(mem->start);
 
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
+	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
 	    (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
 
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
+	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
 	    (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
 
 	/* first TX, then RX */
@@ -549,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 		ret = -ENXIO;
 		goto err_free_mem;
 	}
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start;
+	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!res) {
@@ -557,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 		ret = -ENXIO;
 		goto err_free_mem;
 	}
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start;
+	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
 
 	davinci_i2s_dai.private_data = dev;
 	ret = snd_soc_register_dai(&davinci_i2s_dai);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index eca22d7829d..5d1f98a4c97 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
 		printk(KERN_ERR "GBLCTL write error\n");
 }
 
-static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
-						struct snd_soc_dai *cpu_dai)
-{
-	struct davinci_audio_dev *dev = cpu_dai->private_data;
-	cpu_dai->dma_data = dev->dma_params[substream->stream];
-	return 0;
-}
-
 static void mcasp_start_rx(struct davinci_audio_dev *dev)
 {
 	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
@@ -386,17 +378,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
 
 static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
 {
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (dev->txnumevt)	/* enable FIFO */
+			mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+								FIFO_ENABLE);
 		mcasp_start_tx(dev);
-	else
+	} else {
+		if (dev->rxnumevt)	/* enable FIFO */
+			mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+								FIFO_ENABLE);
 		mcasp_start_rx(dev);
-
-	/* enable FIFO */
-	if (dev->txnumevt)
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
-	if (dev->rxnumevt)
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+	}
 }
 
 static void mcasp_stop_rx(struct davinci_audio_dev *dev)
@@ -413,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev)
 
 static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
 {
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (dev->txnumevt)	/* disable FIFO */
+			mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+								FIFO_ENABLE);
 		mcasp_stop_tx(dev);
-	else
+	} else {
+		if (dev->rxnumevt)	/* disable FIFO */
+			mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+								FIFO_ENABLE);
 		mcasp_stop_rx(dev);
-
-	/* disable FIFO */
-	if (dev->txnumevt)
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
-	if (dev->rxnumevt)
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+	}
 }
 
 static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -512,34 +504,49 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 				       int channel_size)
 {
 	u32 fmt = 0;
+	u32 mask, rotate;
 
 	switch (channel_size) {
 	case DAVINCI_AUDIO_WORD_8:
 		fmt = 0x03;
+		rotate = 6;
+		mask = 0x000000ff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_12:
 		fmt = 0x05;
+		rotate = 5;
+		mask = 0x00000fff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_16:
 		fmt = 0x07;
+		rotate = 4;
+		mask = 0x0000ffff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_20:
 		fmt = 0x09;
+		rotate = 3;
+		mask = 0x000fffff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_24:
 		fmt = 0x0B;
+		rotate = 2;
+		mask = 0x00ffffff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_28:
 		fmt = 0x0D;
+		rotate = 1;
+		mask = 0x0fffffff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_32:
 		fmt = 0x0F;
+		rotate = 0;
+		mask = 0xffffffff;
 		break;
 
 	default:
@@ -550,6 +557,13 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 					RXSSZ(fmt), RXSSZ(0x0F));
 	mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
 					TXSSZ(fmt), TXSSZ(0x0F));
+	mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate),
+							TXROT(7));
+	mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate),
+							RXROT(7));
+	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask);
+	mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask);
+
 	return 0;
 }
 
@@ -638,7 +652,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
 			printk(KERN_ERR "playback tdm slot %d not supported\n",
 				dev->tdm_slots);
 
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF);
 		mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
 	} else {
 		/* bit stream is MSB first with no delay */
@@ -655,7 +668,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
 			printk(KERN_ERR "capture tdm slot %d not supported\n",
 				dev->tdm_slots);
 
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF);
 		mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
 	}
 }
@@ -700,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 {
 	struct davinci_audio_dev *dev = cpu_dai->private_data;
 	struct davinci_pcm_dma_params *dma_params =
-					dev->dma_params[substream->stream];
+					&dev->dma_params[substream->stream];
 	int word_length;
 	u8 numevt;
 
@@ -778,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 }
 
 static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
-	.startup 	= davinci_mcasp_startup,
 	.trigger	= davinci_mcasp_trigger,
 	.hw_params	= davinci_mcasp_hw_params,
 	.set_fmt	= davinci_mcasp_set_dai_fmt,
@@ -829,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	struct resource *mem, *ioarea, *res;
 	struct snd_platform_data *pdata;
 	struct davinci_audio_dev *dev;
-	int count = 0;
 	int ret = 0;
 
 	dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL);
 	if (!dev)
 		return	-ENOMEM;
 
-	dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2,
-								GFP_KERNEL);
-	if (!dma_data) {
-		ret = -ENOMEM;
-		goto err_release_dev;
-	}
-
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!mem) {
 		dev_err(&pdev->dev, "no mem resource?\n");
@@ -877,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	dev->txnumevt = pdata->txnumevt;
 	dev->rxnumevt = pdata->rxnumevt;
 
-	dma_data[count].name = "I2S PCM Stereo out";
-	dma_data[count].eventq_no = pdata->eventq_no;
-	dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
+	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+	dma_data->eventq_no = pdata->eventq_no;
+	dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
 							io_v2p(dev->base));
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count];
 
 	/* first TX, then RX */
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -890,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		goto err_release_region;
 	}
 
-	dma_data[count].channel = res->start;
-	count++;
-	dma_data[count].name = "I2S PCM Stereo in";
-	dma_data[count].eventq_no = pdata->eventq_no;
-	dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
+	dma_data->channel = res->start;
+
+	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+	dma_data->eventq_no = pdata->eventq_no;
+	dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
 							io_v2p(dev->base));
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count];
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!res) {
@@ -904,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		goto err_release_region;
 	}
 
-	dma_data[count].channel = res->start;
+	dma_data->channel = res->start;
 	davinci_mcasp_dai[pdata->op_mode].private_data = dev;
 	davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
 	ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
@@ -916,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 err_release_region:
 	release_mem_region(mem->start, (mem->end - mem->start) + 1);
 err_release_data:
-	kfree(dma_data);
-err_release_dev:
 	kfree(dev);
 
 	return ret;
@@ -926,7 +925,6 @@ err_release_dev:
 static int davinci_mcasp_remove(struct platform_device *pdev)
 {
 	struct snd_platform_data *pdata = pdev->dev.platform_data;
-	struct davinci_pcm_dma_params *dma_data;
 	struct davinci_audio_dev *dev;
 	struct resource *mem;
 
@@ -939,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	release_mem_region(mem->start, (mem->end - mem->start) + 1);
 
-	dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
-	kfree(dma_data);
 	kfree(dev);
 
 	return 0;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 554354c1cc2..9d179cc88f7 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -39,10 +39,15 @@ enum {
 };
 
 struct davinci_audio_dev {
+	/*
+	 * dma_params must be first because rtd->dai->cpu_dai->private_data
+	 * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+	 * davinci_pcm_open.
+	 */
+	struct davinci_pcm_dma_params dma_params[2];
 	void __iomem *base;
 	int sample_rate;
 	struct clk *clk;
-	struct davinci_pcm_dma_params *dma_params[2];
 	unsigned int codec_fmt;
 
 	/* McASP specific data */
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 091dacb78b4..c73a915f233 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
 static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 {
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
 	struct edmacc_param p_ram;
 	int ret;
 
-	if (!dma_data)
-		return -ENODEV;
-
-	prtd->params = dma_data;
-
 	/* Request master DMA channel */
 	ret = edma_alloc_channel(prtd->params->channel,
 				  davinci_pcm_dma_irq, substream,
@@ -145,7 +138,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	prtd->master_lch = ret;
 
 	/* Request parameter RAM reload slot */
-	ret = edma_alloc_slot(EDMA_SLOT_ANY);
+	ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY);
 	if (ret < 0) {
 		edma_free_channel(prtd->master_lch);
 		return ret;
@@ -162,8 +155,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	 * so davinci_pcm_enqueue_dma() takes less time in IRQ.
 	 */
 	edma_read_slot(prtd->slave_lch, &p_ram);
-	p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
-	p_ram.link_bcntrld = prtd->slave_lch << 5;
+	p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->master_lch));
+	p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5;
 	edma_write_slot(prtd->slave_lch, &p_ram);
 
 	return 0;
@@ -244,6 +237,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct davinci_runtime_data *prtd;
 	int ret = 0;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data;
+	struct davinci_pcm_dma_params *params = &pa[substream->stream];
+	if (!params)
+		return -ENODEV;
 
 	snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
 	/* ensure that buffer size is a multiple of period size */
@@ -257,6 +255,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
 		return -ENOMEM;
 
 	spin_lock_init(&prtd->lock);
+	prtd->params = params;
 
 	runtime->private_data = prtd;
 
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 63d96253c73..8746606efc8 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -17,7 +17,6 @@
 
 
 struct davinci_pcm_dma_params {
-	char *name;			/* stream identifier */
 	int channel;			/* sync dma channel ID */
 	unsigned short acnt;
 	dma_addr_t dma_addr;		/* device physical address for DMA */
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 9ff62e3a9b1..6096d22283e 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	int size, irq, rc;
 	const __be32 *prop;
 	void __iomem *regs;
+	int ret;
 
 	/* Fetch the registers and IRQ of the PSC */
 	irq = irq_of_parse_and_map(op->node, 0);
@@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	/* Allocate and initialize the driver private data */
 	psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
 	if (!psc_dma) {
-		iounmap(regs);
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto out_unmap;
 	}
 
 	/* Get the PSC ID */
 	prop = of_get_property(op->node, "cell-index", &size);
-	if (!prop || size < sizeof *prop)
-		return -ENODEV;
+	if (!prop || size < sizeof *prop) {
+		ret = -ENODEV;
+		goto out_free;
+	}
 
 	spin_lock_init(&psc_dma->lock);
 	mutex_init(&psc_dma->mutex);
@@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	if (!psc_dma->capture.bcom_task ||
 	    !psc_dma->playback.bcom_task) {
 		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
-		iounmap(regs);
-		kfree(psc_dma);
-		return -ENODEV;
+		ret = -ENODEV;
+		goto out_free;
 	}
 
 	/* Disable all interrupts and reset the PSC */
@@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
 			  &psc_dma_bcom_irq_tx, IRQF_SHARED,
 			  "psc-dma-playback", &psc_dma->playback);
 	if (rc) {
-		free_irq(psc_dma->irq, psc_dma);
-		free_irq(psc_dma->capture.irq,
-			 &psc_dma->capture);
-		free_irq(psc_dma->playback.irq,
-			 &psc_dma->playback);
-		return -ENODEV;
+		ret = -ENODEV;
+		goto out_irq;
 	}
 
 	/* Save what we've done so it can be found again later */
@@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op)
 
 	/* Tell the ASoC OF helpers about it */
 	return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+out_irq:
+	free_irq(psc_dma->irq, psc_dma);
+	free_irq(psc_dma->capture.irq, &psc_dma->capture);
+	free_irq(psc_dma->playback.irq, &psc_dma->playback);
+out_free:
+	kfree(psc_dma);
+out_unmap:
+	iounmap(regs);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
 
diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c
index 3806ff2c0cd..ccdefe60e75 100644
--- a/sound/soc/imx/mxc-ssi.c
+++ b/sound/soc/imx/mxc-ssi.c
@@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		break;
 	}
 
-	/* sync */
-	if (!(fmt & SND_SOC_DAIFMT_ASYNC))
-		scr |= SSI_SCR_SYN;
-
-	/* tdm - only for stereo atm */
-	if (fmt & SND_SOC_DAIFMT_TDM)
-		scr |= SSI_SCR_NET;
-
 	if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
 		SSI1_STCR = stcr;
 		SSI1_SRCR = srcr;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 6375b4ea525..dcb3181bb34 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701
 
 config SND_PXA2XX_SOC_IMOTE2
        tristate "SoC Audio support for IMote 2"
-       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
        select SND_PXA2XX_SOC_I2S
        select SND_SOC_WM8940
        help
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 5b9ed646478..d11a6d7e384 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -351,7 +351,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
 			do_div(tmp, freq_out);
 			val = tmp;
 
-			val = (val << 16) | 64;;
+			val = (val << 16) | 64;
 			ssp_write_reg(ssp, SSACDD, val);
 
 			ssacd |= (0x6 << 4);
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index aa7af0b8d42..9bc4aa35caa 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -230,6 +230,8 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
 }
 
+#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
+
 /*
  * Wait for the LR signal to allow synchronisation to the L/R clock
  * from the codec. May only be needed for slave mode.
@@ -237,19 +239,21 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s)
 {
 	u32 iiscon;
-	unsigned long timeout = jiffies + msecs_to_jiffies(5);
+	unsigned long loops = msecs_to_loops(5);
 
 	pr_debug("Entered %s\n", __func__);
 
-	while (1) {
+	while (--loops) {
 		iiscon = readl(i2s->regs + S3C2412_IISCON);
 		if (iiscon & S3C2412_IISCON_LRINDEX)
 			break;
 
-		if (timeout < jiffies) {
-			printk(KERN_ERR "%s: timeout\n", __func__);
-			return -ETIMEDOUT;
-		}
+		cpu_relax();
+	}
+
+	if (!loops) {
+		printk(KERN_ERR "%s: timeout\n", __func__);
+		return -ETIMEDOUT;
 	}
 
 	return 0;
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index 8e79a416db5..c215d32d632 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -67,7 +67,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
 {
 	int ret = 0;
 #ifdef ENFORCE_RATES
-	struct snd_pcm_runtime *runtime = substream->runtime;;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 #endif
 
 	mutex_lock(&clk_lock);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 0d8b08ef873..8de6f9dec4a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
 
 		/* connected jack or spk ? */
 		if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
-			widget->id == snd_soc_dapm_line)
+		    (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources)))
 			return 1;
 	}
 
@@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 			return 1;
 
 		/* connected jack ? */
-		if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+		if (widget->id == snd_soc_dapm_mic ||
+		    (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks)))
 			return 1;
 	}
 
@@ -1131,9 +1132,10 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 	ret = snprintf(buf, PAGE_SIZE, "%s: %s  in %d out %d\n",
 		       w->name, w->power ? "On" : "Off", in, out);
 
-	if (w->active && w->sname)
-		ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n",
-				w->sname);
+	if (w->sname)
+		ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+				w->sname,
+				w->active ? "active" : "inactive");
 
 	list_for_each_entry(p, &w->sources, list_sink) {
 		if (p->connect)
diff --git a/sound/sound_core.c b/sound/sound_core.c
index bb4b88e606b..49c99818659 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -29,7 +29,7 @@ MODULE_DESCRIPTION("Core sound module");
 MODULE_AUTHOR("Alan Cox");
 MODULE_LICENSE("GPL");
 
-static char *sound_nodename(struct device *dev)
+static char *sound_devnode(struct device *dev, mode_t *mode)
 {
 	if (MAJOR(dev->devt) == SOUND_MAJOR)
 		return NULL;
@@ -50,7 +50,7 @@ static int __init init_soundcore(void)
 		return PTR_ERR(sound_class);
 	}
 
-	sound_class->nodename = sound_nodename;
+	sound_class->devnode = sound_devnode;
 
 	return 0;
 }
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index ab5a3ac2ac4..9efcfd08d74 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -898,6 +898,11 @@ static struct snd_kcontrol_new usb_feature_unit_ctl = {
  * build a feature control
  */
 
+static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str)
+{
+	return strlcat(kctl->id.name, str, sizeof(kctl->id.name));
+}
+
 static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
 			      unsigned int ctl_mask, int control,
 			      struct usb_audio_term *iterm, int unitid)
@@ -978,13 +983,13 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
 		 */
 		if (! mapped_name && ! (state->oterm.type >> 16)) {
 			if ((state->oterm.type & 0xff00) == 0x0100) {
-				len = strlcat(kctl->id.name, " Capture", sizeof(kctl->id.name));
+				len = append_ctl_name(kctl, " Capture");
 			} else {
-				len = strlcat(kctl->id.name + len, " Playback", sizeof(kctl->id.name));
+				len = append_ctl_name(kctl, " Playback");
 			}
 		}
-		strlcat(kctl->id.name + len, control == USB_FEATURE_MUTE ? " Switch" : " Volume",
-			sizeof(kctl->id.name));
+		append_ctl_name(kctl, control == USB_FEATURE_MUTE ?
+				" Switch" : " Volume");
 		if (control == USB_FEATURE_VOLUME) {
 			kctl->tlv.c = mixer_vol_tlv;
 			kctl->vd[0].access |= 
@@ -1143,7 +1148,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc,
 		len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0);
 	if (! len)
 		len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1);
-	strlcat(kctl->id.name + len, " Volume", sizeof(kctl->id.name));
+	append_ctl_name(kctl, " Volume");
 
 	snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n",
 		    cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
@@ -1400,8 +1405,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned
 			if (! len)
 				strlcpy(kctl->id.name, name, sizeof(kctl->id.name));
 		}
-		strlcat(kctl->id.name, " ", sizeof(kctl->id.name));
-		strlcat(kctl->id.name, valinfo->suffix, sizeof(kctl->id.name));
+		append_ctl_name(kctl, " ");
+		append_ctl_name(kctl, valinfo->suffix);
 
 		snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n",
 			    cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
@@ -1610,9 +1615,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi
 			strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
 
 		if ((state->oterm.type & 0xff00) == 0x0100)
-			strlcat(kctl->id.name, " Capture Source", sizeof(kctl->id.name));
+			append_ctl_name(kctl, " Capture Source");
 		else
-			strlcat(kctl->id.name, " Playback Source", sizeof(kctl->id.name));
+			append_ctl_name(kctl, " Playback Source");
 	}
 
 	snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n",
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index fd44946ce4b..99f33766cd5 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -154,7 +154,7 @@ static void usb_stream_hwdep_vm_close(struct vm_area_struct *area)
 	snd_printdd(KERN_DEBUG "%i\n", atomic_read(&us122l->mmap_count));
 }
 
-static struct vm_operations_struct usb_stream_hwdep_vm_ops = {
+static const struct vm_operations_struct usb_stream_hwdep_vm_ops = {
 	.open = usb_stream_hwdep_vm_open,
 	.fault = usb_stream_hwdep_vm_fault,
 	.close = usb_stream_hwdep_vm_close,
diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c
index f3d8f71265d..52e04b2f35d 100644
--- a/sound/usb/usx2y/usX2Yhwdep.c
+++ b/sound/usb/usx2y/usX2Yhwdep.c
@@ -53,7 +53,7 @@ static int snd_us428ctls_vm_fault(struct vm_area_struct *area,
 	return 0;
 }
 
-static struct vm_operations_struct us428ctls_vm_ops = {
+static const struct vm_operations_struct us428ctls_vm_ops = {
 	.fault = snd_us428ctls_vm_fault,
 };
 
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c
index 117946f2deb..4b2304c2e02 100644
--- a/sound/usb/usx2y/usx2yhwdeppcm.c
+++ b/sound/usb/usx2y/usx2yhwdeppcm.c
@@ -697,7 +697,7 @@ static int snd_usX2Y_hwdep_pcm_vm_fault(struct vm_area_struct *area,
 }
 
 
-static struct vm_operations_struct snd_usX2Y_hwdep_pcm_vm_ops = {
+static const struct vm_operations_struct snd_usX2Y_hwdep_pcm_vm_ops = {
 	.open = snd_usX2Y_hwdep_pcm_vm_open,
 	.close = snd_usX2Y_hwdep_pcm_vm_close,
 	.fault = snd_usX2Y_hwdep_pcm_vm_fault,
author	Frederic Weisbecker <fweisbec@gmail.com>	2009-10-18 01:09:09 +0200
committer	Frederic Weisbecker <fweisbec@gmail.com>	2009-10-18 01:12:33 +0200
commit	0f8f86c7bdd1c954fbe153af437a0d91a6c5721a (patch)
tree	94a8d419a470a4f9852ca397bb9bbe48db92ff5c /sound
parent	dca2d6ac09d9ef59ff46820d4f0c94b08a671202 (diff)
parent	f39cdf25bf77219676ec5360980ac40b1a7e144a (diff)