Merge branch 'master' of git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6

The SmartMedia FTL code depends on new kfifo bits from 2.6.33

52 files changed, 851 insertions, 409 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1497dce1b04a..656e474dca47 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -172,14 +172,15 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 	return v;
 }
 
-static inline void aaci_chan_wait_ready(struct aaci_runtime *aacirun)
+static inline void
+aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask)
 {
 	u32 val;
 	int timeout = 5000;
 
 	do {
 		val = readl(aacirun->base + AACI_SR);
-	} while (val & (SR_TXB|SR_RXB) && timeout--);
+	} while (val & mask && timeout--);
 }
 
 
@@ -208,8 +209,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
 			writel(0, aacirun->base + AACI_IE);
 			return;
 		}
-		ptr = aacirun->ptr;
 
+		spin_lock(&aacirun->lock);
+
+		ptr = aacirun->ptr;
 		do {
 			unsigned int len = aacirun->fifosz;
 			u32 val;
@@ -217,9 +220,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
 			if (aacirun->bytes <= 0) {
 				aacirun->bytes += aacirun->period;
 				aacirun->ptr = ptr;
-				spin_unlock(&aaci->lock);
+				spin_unlock(&aacirun->lock);
 				snd_pcm_period_elapsed(aacirun->substream);
-				spin_lock(&aaci->lock);
+				spin_lock(&aacirun->lock);
 			}
 			if (!(aacirun->cr & CR_EN))
 				break;
@@ -245,7 +248,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
 					ptr = aacirun->start;
 			}
 		} while(1);
+
 		aacirun->ptr = ptr;
+
+		spin_unlock(&aacirun->lock);
 	}
 
 	if (mask & ISR_URINTR) {
@@ -263,6 +269,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
 			return;
 		}
 
+		spin_lock(&aacirun->lock);
+
 		ptr = aacirun->ptr;
 		do {
 			unsigned int len = aacirun->fifosz;
@@ -271,9 +279,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
 			if (aacirun->bytes <= 0) {
 				aacirun->bytes += aacirun->period;
 				aacirun->ptr = ptr;
-				spin_unlock(&aaci->lock);
+				spin_unlock(&aacirun->lock);
 				snd_pcm_period_elapsed(aacirun->substream);
-				spin_lock(&aaci->lock);
+				spin_lock(&aacirun->lock);
 			}
 			if (!(aacirun->cr & CR_EN))
 				break;
@@ -301,6 +309,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
 		} while (1);
 
 		aacirun->ptr = ptr;
+
+		spin_unlock(&aacirun->lock);
 	}
 }
 
@@ -310,7 +320,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
 	u32 mask;
 	int i;
 
-	spin_lock(&aaci->lock);
 	mask = readl(aaci->base + AACI_ALLINTS);
 	if (mask) {
 		u32 m = mask;
@@ -320,7 +329,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
 			}
 		}
 	}
-	spin_unlock(&aaci->lock);
 
 	return mask ? IRQ_HANDLED : IRQ_NONE;
 }
@@ -330,63 +338,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
 /*
  * ALSA support.
  */
-
-struct aaci_stream {
-	unsigned char codec_idx;
-	unsigned char rate_idx;
-};
-
-static struct aaci_stream aaci_streams[] = {
-	[ACSTREAM_FRONT] = {
-		.codec_idx	= 0,
-		.rate_idx	= AC97_RATES_FRONT_DAC,
-	},
-	[ACSTREAM_SURROUND] = {
-		.codec_idx	= 0,
-		.rate_idx	= AC97_RATES_SURR_DAC,
-	},
-	[ACSTREAM_LFE] = {
-		.codec_idx	= 0,
-		.rate_idx	= AC97_RATES_LFE_DAC,
-	},
-};
-
-static inline unsigned int aaci_rate_mask(struct aaci *aaci, int streamid)
-{
-	struct aaci_stream *s = aaci_streams + streamid;
-	return aaci->ac97_bus->codec[s->codec_idx]->rates[s->rate_idx];
-}
-
-static unsigned int rate_list[] = {
-	5512, 8000, 11025, 16000, 22050, 32000, 44100,
-	48000, 64000, 88200, 96000, 176400, 192000
-};
-
-/*
- * Double-rate rule: we can support double rate iff channels == 2
- *  (unimplemented)
- */
-static int
-aaci_rule_rate_by_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule)
-{
-	struct aaci *aaci = rule->private;
-	unsigned int rate_mask = SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_5512;
-	struct snd_interval *c = hw_param_interval(p, SNDRV_PCM_HW_PARAM_CHANNELS);
-
-	switch (c->max) {
-	case 6:
-		rate_mask &= aaci_rate_mask(aaci, ACSTREAM_LFE);
-	case 4:
-		rate_mask &= aaci_rate_mask(aaci, ACSTREAM_SURROUND);
-	case 2:
-		rate_mask &= aaci_rate_mask(aaci, ACSTREAM_FRONT);
-	}
-
-	return snd_interval_list(hw_param_interval(p, rule->var),
-				 ARRAY_SIZE(rate_list), rate_list,
-				 rate_mask);
-}
-
 static struct snd_pcm_hardware aaci_hw_info = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
@@ -400,10 +351,7 @@ static struct snd_pcm_hardware aaci_hw_info = {
 	 */
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 
-	/* should this be continuous or knot? */
-	.rates			= SNDRV_PCM_RATE_CONTINUOUS,
-	.rate_max		= 48000,
-	.rate_min		= 4000,
+	/* rates are setup from the AC'97 codec */
 	.channels_min		= 2,
 	.channels_max		= 6,
 	.buffer_bytes_max	= 64 * 1024,
@@ -423,6 +371,12 @@ static int __aaci_pcm_open(struct aaci *aaci,
 	aacirun->substream = substream;
 	runtime->private_data = aacirun;
 	runtime->hw = aaci_hw_info;
+	runtime->hw.rates = aacirun->pcm->rates;
+	snd_pcm_limit_hw_rates(runtime);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+	    aacirun->pcm->r[1].slots)
+		snd_ac97_pcm_double_rate_rules(runtime);
 
 	/*
 	 * FIXME: ALSA specifies fifo_size in bytes.  If we're in normal
@@ -433,17 +387,6 @@ static int __aaci_pcm_open(struct aaci *aaci,
 	 */
 	runtime->hw.fifo_size = aaci->fifosize * 2;
 
-	/*
-	 * Add rule describing hardware rate dependency
-	 * on the number of channels.
-	 */
-	ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				  aaci_rule_rate_by_channels, aaci,
-				  SNDRV_PCM_HW_PARAM_CHANNELS,
-				  SNDRV_PCM_HW_PARAM_RATE, -1);
-	if (ret)
-		goto out;
-
 	ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED,
 			  DRIVER_NAME, aaci);
 	if (ret)
@@ -498,6 +441,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *params)
 {
 	int err;
+	struct aaci *aaci = substream->private_data;
 
 	aaci_pcm_hw_free(substream);
 	if (aacirun->pcm_open) {
@@ -507,18 +451,22 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	err = snd_pcm_lib_malloc_pages(substream,
 				       params_buffer_bytes(params));
-	if (err < 0)
-		goto out;
+	if (err >= 0) {
+		unsigned int rate = params_rate(params);
+		int dbl = rate > 48000;
 
-	err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
-				params_channels(params),
-				aacirun->pcm->r[0].slots);
-	if (err)
-		goto out;
+		err = snd_ac97_pcm_open(aacirun->pcm, rate,
+					params_channels(params),
+					aacirun->pcm->r[dbl].slots);
 
-	aacirun->pcm_open = 1;
+		aacirun->pcm_open = err == 0;
+		aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
+		aacirun->fifosz = aaci->fifosize * 4;
+
+		if (aacirun->cr & CR_COMPACT)
+			aacirun->fifosz >>= 1;
+	}
 
- out:
 	return err;
 }
 
@@ -527,7 +475,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct aaci_runtime *aacirun = runtime->private_data;
 
-	aacirun->start	= (void *)runtime->dma_area;
+	aacirun->start	= runtime->dma_area;
 	aacirun->end	= aacirun->start + snd_pcm_lib_buffer_bytes(substream);
 	aacirun->ptr	= aacirun->start;
 	aacirun->period	=
@@ -613,7 +561,6 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream)
 static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream,
 				       struct snd_pcm_hw_params *params)
 {
-	struct aaci *aaci = substream->private_data;
 	struct aaci_runtime *aacirun = substream->runtime->private_data;
 	unsigned int channels = params_channels(params);
 	int ret;
@@ -627,14 +574,9 @@ static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream,
 	 * Enable FIFO, compact mode, 16 bits per sample.
 	 * FIXME: double rate slots?
 	 */
-	if (ret >= 0) {
-		aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
+	if (ret >= 0)
 		aacirun->cr |= channels_to_txmask[channels];
 
-		aacirun->fifosz	= aaci->fifosize * 4;
-		if (aacirun->cr & CR_COMPACT)
-			aacirun->fifosz >>= 1;
-	}
 	return ret;
 }
 
@@ -646,7 +588,7 @@ static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun)
 	ie &= ~(IE_URIE|IE_TXIE);
 	writel(ie, aacirun->base + AACI_IE);
 	aacirun->cr &= ~CR_EN;
-	aaci_chan_wait_ready(aacirun);
+	aaci_chan_wait_ready(aacirun, SR_TXB);
 	writel(aacirun->cr, aacirun->base + AACI_TXCR);
 }
 
@@ -654,7 +596,7 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun)
 {
 	u32 ie;
 
-	aaci_chan_wait_ready(aacirun);
+	aaci_chan_wait_ready(aacirun, SR_TXB);
 	aacirun->cr |= CR_EN;
 
 	ie = readl(aacirun->base + AACI_IE);
@@ -665,12 +607,12 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun)
 
 static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 {
-	struct aaci *aaci = substream->private_data;
 	struct aaci_runtime *aacirun = substream->runtime->private_data;
 	unsigned long flags;
 	int ret = 0;
 
-	spin_lock_irqsave(&aaci->lock, flags);
+	spin_lock_irqsave(&aacirun->lock, flags);
+
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		aaci_pcm_playback_start(aacirun);
@@ -697,7 +639,8 @@ static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cm
 	default:
 		ret = -EINVAL;
 	}
-	spin_unlock_irqrestore(&aaci->lock, flags);
+
+	spin_unlock_irqrestore(&aacirun->lock, flags);
 
 	return ret;
 }
@@ -716,23 +659,14 @@ static struct snd_pcm_ops aaci_playback_ops = {
 static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream,
 				      struct snd_pcm_hw_params *params)
 {
-	struct aaci *aaci = substream->private_data;
 	struct aaci_runtime *aacirun = substream->runtime->private_data;
 	int ret;
 
 	ret = aaci_pcm_hw_params(substream, aacirun, params);
-
-	if (ret >= 0) {
-		aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
-
+	if (ret >= 0)
 		/* Line in record: slot 3 and 4 */
 		aacirun->cr |= CR_SL3 | CR_SL4;
 
-		aacirun->fifosz = aaci->fifosize * 4;
-
-		if (aacirun->cr & CR_COMPACT)
-			aacirun->fifosz >>= 1;
-	}
 	return ret;
 }
 
@@ -740,7 +674,7 @@ static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun)
 {
 	u32 ie;
 
-	aaci_chan_wait_ready(aacirun);
+	aaci_chan_wait_ready(aacirun, SR_RXB);
 
 	ie = readl(aacirun->base + AACI_IE);
 	ie &= ~(IE_ORIE | IE_RXIE);
@@ -755,7 +689,7 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun)
 {
 	u32 ie;
 
-	aaci_chan_wait_ready(aacirun);
+	aaci_chan_wait_ready(aacirun, SR_RXB);
 
 #ifdef DEBUG
 	/* RX Timeout value: bits 28:17 in RXCR */
@@ -772,12 +706,11 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun)
 
 static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 {
-	struct aaci *aaci = substream->private_data;
 	struct aaci_runtime *aacirun = substream->runtime->private_data;
 	unsigned long flags;
 	int ret = 0;
 
-	spin_lock_irqsave(&aaci->lock, flags);
+	spin_lock_irqsave(&aacirun->lock, flags);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -806,7 +739,7 @@ static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd
 		ret = -EINVAL;
 	}
 
-	spin_unlock_irqrestore(&aaci->lock, flags);
+	spin_unlock_irqrestore(&aacirun->lock, flags);
 
 	return ret;
 }
@@ -889,6 +822,12 @@ static struct ac97_pcm ac97_defs[] __devinitdata = {
 					  (1 << AC97_SLOT_PCM_SRIGHT) |
 					  (1 << AC97_SLOT_LFE),
 			},
+			[1] = {
+				.slots	= (1 << AC97_SLOT_PCM_LEFT) |
+					  (1 << AC97_SLOT_PCM_RIGHT) |
+					  (1 << AC97_SLOT_PCM_LEFT_0) |
+					  (1 << AC97_SLOT_PCM_RIGHT_0),
+			},
 		},
 	},
 	[1] = {	/* PCM in */
@@ -1001,7 +940,6 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev)
 
 	aaci = card->private_data;
 	mutex_init(&aaci->ac97_sem);
-	spin_lock_init(&aaci->lock);
 	aaci->card = card;
 	aaci->dev = dev;
 
@@ -1028,7 +966,7 @@ static int __devinit aaci_init_pcm(struct aaci *aaci)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops);
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops);
 		snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
-						      NULL, 0, 64 * 104);
+						      NULL, 0, 64 * 1024);
 	}
 
 	return ret;
@@ -1088,12 +1026,14 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
 	/*
 	 * Playback uses AACI channel 0
 	 */
+	spin_lock_init(&aaci->playback.lock);
 	aaci->playback.base = aaci->base + AACI_CSCH1;
 	aaci->playback.fifo = aaci->base + AACI_DR1;
 
 	/*
 	 * Capture uses AACI channel 0
 	 */
+	spin_lock_init(&aaci->capture.lock);
 	aaci->capture.base = aaci->base + AACI_CSCH1;
 	aaci->capture.fifo = aaci->base + AACI_DR1;
 
diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h
index 924f69c1c44c..6a4a2eebdda1 100644
--- a/sound/arm/aaci.h
+++ b/sound/arm/aaci.h
@@ -202,6 +202,7 @@
 struct aaci_runtime {
 	void			__iomem *base;
 	void			__iomem *fifo;
+	spinlock_t		lock;
 
 	struct ac97_pcm		*pcm;
 	int			pcm_open;
@@ -232,7 +233,6 @@ struct aaci {
 	struct snd_ac97		*ac97;
 
 	u32			maincr;
-	spinlock_t		lock;
 
 	struct aaci_runtime	playback;
 	struct aaci_runtime	capture;
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index c15682a2f9db..475455c76610 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -5,6 +5,7 @@ config SND_TIMER
 config SND_PCM
 	tristate
 	select SND_TIMER
+	select GCD
 
 config SND_HWDEP
 	tristate
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 30f410832a25..a27545b23ee9 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -758,7 +758,7 @@ int snd_interval_ratnum(struct snd_interval *i,
 		int diff;
 		if (q == 0)
 			q = 1;
-		den = div_down(num, q);
+		den = div_up(num, q);
 		if (den < rats[k].den_min)
 			continue;
 		if (den > rats[k].den_max)
@@ -794,7 +794,7 @@ int snd_interval_ratnum(struct snd_interval *i,
 			i->empty = 1;
 			return -EINVAL;
 		}
-		den = div_up(num, q);
+		den = div_down(num, q);
 		if (den > rats[k].den_max)
 			continue;
 		if (den < rats[k].den_min)
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 29ab46a12e11..25b0641e6b8c 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1918,13 +1918,13 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
 
 	err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
 					   hw->rate_min, hw->rate_max);
-	 if (err < 0)
-		 return err;
+	if (err < 0)
+		return err;
 
 	err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
 					   hw->period_bytes_min, hw->period_bytes_max);
-	 if (err < 0)
-		 return err;
+	if (err < 0)
+		return err;
 
 	err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS,
 					   hw->periods_min, hw->periods_max);
diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c
index ca8068b63d6c..b01d9481d632 100644
--- a/sound/core/pcm_timer.c
+++ b/sound/core/pcm_timer.c
@@ -20,6 +20,7 @@
  */
 
 #include <linux/time.h>
+#include <linux/gcd.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/timer.h>
@@ -28,22 +29,6 @@
  *  Timer functions
  */
 
-/* Greatest common divisor */
-static unsigned long gcd(unsigned long a, unsigned long b)
-{
-	unsigned long r;
-	if (a < b) {
-		r = a;
-		a = b;
-		b = r;
-	}
-	while ((r = a % b) != 0) {
-		a = b;
-		b = r;
-	}
-	return b;
-}
-
 void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream)
 {
 	unsigned long rate, mult, fsize, l, post;
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 7872a02f6ca9..563d1967a0ad 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void)
 	unregister_chrdev(major, "alsa");
 }
 
-module_init(alsa_sound_init)
-module_exit(alsa_sound_exit)
+subsys_initcall(alsa_sound_init);
+module_exit(alsa_sound_exit);
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 7fe12264ff80..0c164e5e4322 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
 	default:
 		return -EINVAL;
 	}
-	if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+	if (minor < 0 || minor >= SNDRV_OSS_MINORS)
 		return -EINVAL;
 	return minor;
 }
diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c
index cb9aa4c4edd0..4be562b2cf21 100644
--- a/sound/isa/msnd/msnd_midi.c
+++ b/sound/isa/msnd/msnd_midi.c
@@ -162,7 +162,7 @@ int snd_msndmidi_new(struct snd_card *card, int device)
 	err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi);
 	if (err < 0)
 		return err;
-	mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL);
+	mpu = kzalloc(sizeof(*mpu), GFP_KERNEL);
 	if (mpu == NULL) {
 		snd_device_free(card, rmidi);
 		return -ENOMEM;
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 96678d5d3834..0c40951b6523 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -377,12 +377,13 @@ init_arrays(struct snd_emu8000 *emu)
 static void __devinit
 size_dram(struct snd_emu8000 *emu)
 {
-	int i, size;
+	int i, size, detected_size;
 
 	if (emu->dram_checked)
 		return;
 
 	size = 0;
+	detected_size = 0;
 
 	/* write out a magic number */
 	snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE);
@@ -414,7 +415,9 @@ size_dram(struct snd_emu8000 *emu)
 		/*snd_emu8000_read_wait(emu);*/
 		EMU8000_SMLD_READ(emu); /* discard stale data  */
 		if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2)
-			break; /* we must have wrapped around */
+			break; /* no memory at this address */
+
+		detected_size = size;
 
 		snd_emu8000_read_wait(emu);
 
@@ -442,9 +445,9 @@ size_dram(struct snd_emu8000 *emu)
 	snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE);
 
 	snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n",
-		    emu->port1, size/1024);
+		    emu->port1, detected_size/1024);
 
-	emu->mem_size = size;
+	emu->mem_size = detected_size;
 	emu->dram_checked = 1;
 }
 
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 8691f4cf6191..f1d9d16b5486 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -609,7 +609,7 @@ static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
 	/* alloc virtual 'dma' area */
 	if (runtime->dma_area)
 		vfree(runtime->dma_area);
-	runtime->dma_area = vmalloc(size);
+	runtime->dma_area = vmalloc_user(size);
 	if (runtime->dma_area == NULL)
 		return -ENOMEM;
 	runtime->dma_bytes = size;
diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c
index 08274c995d06..727bdb9ba2dc 100644
--- a/sound/oss/dev_table.c
+++ b/sound/oss/dev_table.c
@@ -67,14 +67,15 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
 		return -(EBUSY);
 	}
 	d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver)));
-
-	if (sound_nblocks < 1024)
-		sound_nblocks++;
+	sound_nblocks++;
+	if (sound_nblocks >= MAX_MEM_BLOCKS)
+		sound_nblocks = MAX_MEM_BLOCKS - 1;
 
 	op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations)));
+	sound_nblocks++;
+	if (sound_nblocks >= MAX_MEM_BLOCKS)
+		sound_nblocks = MAX_MEM_BLOCKS - 1;
 
-	if (sound_nblocks < 1024)
-		sound_nblocks++;
 	if (d == NULL || op == NULL) {
 		printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name);
 		sound_unload_audiodev(num);
@@ -128,9 +129,10 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
 	   until you unload sound! */
 	   
 	op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations)));
+	sound_nblocks++;
+	if (sound_nblocks >= MAX_MEM_BLOCKS)
+		sound_nblocks = MAX_MEM_BLOCKS - 1;
 
-	if (sound_nblocks < 1024)
-		sound_nblocks++;
 	if (op == NULL) {
 		printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name);
 		return -ENOMEM;
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 83f5ee236b12..e19dd5dcc2de 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -269,7 +269,7 @@ static int pss_reset_dsp(pss_confdata * devc)
 	unsigned long   i, limit = jiffies + HZ/10;
 
 	outw(0x2000, REG(PSS_CONTROL));
-	for (i = 0; i < 32768 && (limit-jiffies >= 0); i++)
+	for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
 		inw(REG(PSS_CONTROL));
 	outw(0x0000, REG(PSS_CONTROL));
 	return 1;
@@ -369,11 +369,11 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size
 		outw(0, REG(PSS_DATA));
 
 		limit = jiffies + HZ/10;
-		for (i = 0; i < 32768 && (limit - jiffies >= 0); i++)
+		for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
 			val = inw(REG(PSS_STATUS));
 
 		limit = jiffies + HZ/10;
-		for (i = 0; i < 32768 && (limit-jiffies >= 0); i++)
+		for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
 		{
 			val = inw(REG(PSS_STATUS));
 			if (val & 0x4000)
diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h
index 55271fbe7f49..9d35c4c65b9b 100644
--- a/sound/oss/sound_config.h
+++ b/sound/oss/sound_config.h
@@ -142,4 +142,6 @@ static inline int translate_mode(struct file *file)
 #define TIMER_ARMED	121234
 #define TIMER_NOT_ARMED	1
 
+#define MAX_MEM_BLOCKS 1024
+
 #endif
diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c
index 61aaedae6b7e..c62530943888 100644
--- a/sound/oss/soundcard.c
+++ b/sound/oss/soundcard.c
@@ -56,7 +56,7 @@
 /*
  * Table for permanently allocated memory (used when unloading the module)
  */
-void *          sound_mem_blocks[1024];
+void *          sound_mem_blocks[MAX_MEM_BLOCKS];
 int             sound_nblocks = 0;
 
 /* Persistent DMA buffers */
@@ -574,7 +574,7 @@ static int __init oss_init(void)
 				      NULL, "%s%d", dev_list[i].name, j);
 	}
 
-	if (sound_nblocks >= 1024)
+	if (sound_nblocks >= MAX_MEM_BLOCKS - 1)
 		printk(KERN_ERR "Sound warning: Deallocation table was too small.\n");
 	
 	return 0;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index c11920623009..a7630e9edf8a 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -83,6 +83,7 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = {
 { 0x4e534300, 0xffffff00, "National Semiconductor", NULL, NULL },
 { 0x50534300, 0xffffff00, "Philips",		NULL,	NULL },
 { 0x53494c00, 0xffffff00, "Silicon Laboratory",	NULL,	NULL },
+{ 0x53544d00, 0xffffff00, "STMicroelectronics",	NULL,	NULL },
 { 0x54524100, 0xffffff00, "TriTech",		NULL,	NULL },
 { 0x54584e00, 0xffffff00, "Texas Instruments",	NULL,	NULL },
 { 0x56494100, 0xffffff00, "VIA Technologies",   NULL,	NULL },
@@ -161,6 +162,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
 { 0x4e534350, 0xffffffff, "LM4550",		patch_lm4550,  	NULL }, // volume wrap fix 
 { 0x50534304, 0xffffffff, "UCB1400",		patch_ucb1400,	NULL },
 { 0x53494c20, 0xffffffe0, "Si3036,8",		mpatch_si3036,	mpatch_si3036, AC97_MODEM_PATCH },
+{ 0x53544d02, 0xffffffff, "ST7597",		NULL,		NULL },
 { 0x54524102, 0xffffffff, "TR28022",		NULL,		NULL },
 { 0x54524103, 0xffffffff, "TR28023",		NULL,		NULL },
 { 0x54524106, 0xffffffff, "TR28026",		NULL,		NULL },
@@ -213,6 +215,14 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg)
 {
 	/* filter some registers for buggy codecs */
 	switch (ac97->id) {
+	case AC97_ID_ST_AC97_ID4:
+		if (reg == 0x08)
+			return 0;
+		/* fall through */
+	case AC97_ID_ST7597:
+		if (reg == 0x22 || reg == 0x7a)
+			return 1;
+		/* fall through */
 	case AC97_ID_AK4540:
 	case AC97_ID_AK4542:
 		if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c)
diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h
index c129492c82b3..d603147c4a96 100644
--- a/sound/pci/ac97/ac97_id.h
+++ b/sound/pci/ac97/ac97_id.h
@@ -62,3 +62,5 @@
 #define AC97_ID_CM9761_78	0x434d4978
 #define AC97_ID_CM9761_82	0x434d4982
 #define AC97_ID_CM9761_83	0x434d4983
+#define AC97_ID_ST7597		0x53544d02
+#define AC97_ID_ST_AC97_ID4	0x53544d04
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 139cf3b2b9d7..d9266bae2849 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1870,6 +1870,7 @@ static unsigned int ad1981_jacks_blacklist[] = {
 	0x10140554, /* Thinkpad T42p/R50p */
 	0x10140567, /* Thinkpad T43p 2668-G7U */
 	0x10140581, /* Thinkpad X41-2527 */
+	0x10280160, /* Dell Dimension 2400 */
 	0x104380b0, /* Asus A7V8X-MX */
 	0x11790241, /* Toshiba Satellite A-15 S127 */
 	0x144dc01a, /* Samsung NP-X20C004/SEG */
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index d6752dff2a44..42b4fbbd8e2b 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -297,6 +297,7 @@ static struct pci_device_id snd_atiixp_ids[] = {
 MODULE_DEVICE_TABLE(pci, snd_atiixp_ids);
 
 static struct snd_pci_quirk atiixp_quirks[] __devinitdata = {
+	SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0),
 	SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0),
 	{ } /* terminator */
 };
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index cb65bd0dd35b..459c1f62783b 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -166,18 +166,7 @@ static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm)
 
 static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index)
 {
-	struct ct_vm *vm;
-	void *kvirt_addr;
-	unsigned long phys_addr;
-
-	vm = atc->vm;
-	kvirt_addr = vm->get_ptp_virt(vm, index);
-	if (kvirt_addr == NULL)
-		phys_addr = (~0UL);
-	else
-		phys_addr = virt_to_phys(kvirt_addr);
-
-	return phys_addr;
+	return atc->vm->get_ptp_phys(atc->vm, index);
 }
 
 static unsigned int convert_format(snd_pcm_format_t snd_format)
@@ -1669,7 +1658,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
 	}
 
 	/* Set up device virtual memory management object */
-	err = ct_vm_create(&atc->vm);
+	err = ct_vm_create(&atc->vm, pci);
 	if (err < 0)
 		goto error1;
 
diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c
index 6b78752e9503..65da6e466f80 100644
--- a/sound/pci/ctxfi/ctvmem.c
+++ b/sound/pci/ctxfi/ctvmem.c
@@ -138,7 +138,7 @@ ct_vm_map(struct ct_vm *vm, struct snd_pcm_substream *substream, int size)
 		return NULL;
 	}
 
-	ptp = vm->ptp[0];
+	ptp = (unsigned long *)vm->ptp[0].area;
 	pte_start = (block->addr >> CT_PAGE_SHIFT);
 	pages = block->size >> CT_PAGE_SHIFT;
 	for (i = 0; i < pages; i++) {
@@ -158,25 +158,25 @@ static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block)
 }
 
 /* *
- * return the host (kmalloced) addr of the @index-th device
- * page talbe page on success, or NULL on failure.
- * The first returned NULL indicates the termination.
+ * return the host physical addr of the @index-th device
+ * page table page on success, or ~0UL on failure.
+ * The first returned ~0UL indicates the termination.
  * */
-static void *
-ct_get_ptp_virt(struct ct_vm *vm, int index)
+static dma_addr_t
+ct_get_ptp_phys(struct ct_vm *vm, int index)
 {
-	void *addr;
+	dma_addr_t addr;
 
-	addr = (index >= CT_PTP_NUM) ? NULL : vm->ptp[index];
+	addr = (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr;
 
 	return addr;
 }
 
-int ct_vm_create(struct ct_vm **rvm)
+int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci)
 {
 	struct ct_vm *vm;
 	struct ct_vm_block *block;
-	int i;
+	int i, err = 0;
 
 	*rvm = NULL;
 
@@ -188,23 +188,21 @@ int ct_vm_create(struct ct_vm **rvm)
 
 	/* Allocate page table pages */
 	for (i = 0; i < CT_PTP_NUM; i++) {
-		vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL);
-		if (!vm->ptp[i])
+		err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+					  snd_dma_pci_data(pci),
+					  PAGE_SIZE, &vm->ptp[i]);
+		if (err < 0)
 			break;
 	}
-	if (!i) {
+	if (err < 0) {
 		/* no page table pages are allocated */
-		kfree(vm);
+		ct_vm_destroy(vm);
 		return -ENOMEM;
 	}
 	vm->size = CT_ADDRS_PER_PAGE * i;
-	/* Initialise remaining ptps */
-	for (; i < CT_PTP_NUM; i++)
-		vm->ptp[i] = NULL;
-
 	vm->map = ct_vm_map;
 	vm->unmap = ct_vm_unmap;
-	vm->get_ptp_virt = ct_get_ptp_virt;
+	vm->get_ptp_phys = ct_get_ptp_phys;
 	INIT_LIST_HEAD(&vm->unused);
 	INIT_LIST_HEAD(&vm->used);
 	block = kzalloc(sizeof(*block), GFP_KERNEL);
@@ -242,7 +240,7 @@ void ct_vm_destroy(struct ct_vm *vm)
 
 	/* free allocated page table pages */
 	for (i = 0; i < CT_PTP_NUM; i++)
-		kfree(vm->ptp[i]);
+		snd_dma_free_pages(&vm->ptp[i]);
 
 	vm->size = 0;
 
diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h
index 01e4fd0386a3..b23adfca4de6 100644
--- a/sound/pci/ctxfi/ctvmem.h
+++ b/sound/pci/ctxfi/ctvmem.h
@@ -22,6 +22,8 @@
 
 #include <linux/mutex.h>
 #include <linux/list.h>
+#include <linux/pci.h>
+#include <sound/memalloc.h>
 
 /* The chip can handle the page table of 4k pages
  * (emu20k1 can handle even 8k pages, but we don't use it right now)
@@ -41,7 +43,7 @@ struct snd_pcm_substream;
 
 /* Virtual memory management object for card device */
 struct ct_vm {
-	void *ptp[CT_PTP_NUM];		/* Device page table pages */
+	struct snd_dma_buffer ptp[CT_PTP_NUM];	/* Device page table pages */
 	unsigned int size;		/* Available addr space in bytes */
 	struct list_head unused;	/* List of unused blocks */
 	struct list_head used;		/* List of used blocks */
@@ -52,10 +54,10 @@ struct ct_vm {
 				   int size);
 	/* Unmap device logical addr area. */
 	void (*unmap)(struct ct_vm *, struct ct_vm_block *block);
-	void *(*get_ptp_virt)(struct ct_vm *vm, int index);
+	dma_addr_t (*get_ptp_phys)(struct ct_vm *vm, int index);
 };
 
-int ct_vm_create(struct ct_vm **rvm);
+int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci);
 void ct_vm_destroy(struct ct_vm *vm);
 
 #endif /* CTVMEM_H */
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 5fe34a8d8c81..e4581a42ace5 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -42,7 +42,7 @@ static void snd_hda_generate_beep(struct work_struct *work)
 		return;
 
 	/* generate tone */
-	snd_hda_codec_write_cache(codec, beep->nid, 0,
+	snd_hda_codec_write(codec, beep->nid, 0,
 			AC_VERB_SET_BEEP_CONTROL, beep->tone);
 }
 
@@ -119,7 +119,7 @@ static void snd_hda_do_detach(struct hda_beep *beep)
 	beep->dev = NULL;
 	cancel_work_sync(&beep->beep_work);
 	/* turn off beep for sure */
-	snd_hda_codec_write_cache(beep->codec, beep->nid, 0,
+	snd_hda_codec_write(beep->codec, beep->nid, 0,
 				  AC_VERB_SET_BEEP_CONTROL, 0);
 }
 
@@ -192,7 +192,7 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
 		beep->enabled = enable;
 		if (!enable) {
 			/* turn off beep */
-			snd_hda_codec_write_cache(beep->codec, beep->nid, 0,
+			snd_hda_codec_write(beep->codec, beep->nid, 0,
 						  AC_VERB_SET_BEEP_CONTROL, 0);
 		}
 		if (beep->mode == HDA_BEEP_MODE_SWREG) {
@@ -239,8 +239,12 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 	mutex_init(&beep->mutex);
 
 	if (beep->mode == HDA_BEEP_MODE_ON) {
-		beep->enabled = 1;
-		snd_hda_do_register(&beep->register_work);
+		int err = snd_hda_do_attach(beep);
+		if (err < 0) {
+			kfree(beep);
+			codec->beep = NULL;
+			return err;
+		}
 	}
 
 	return 0;
@@ -253,7 +257,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 	if (beep) {
 		cancel_work_sync(&beep->register_work);
 		cancel_delayed_work(&beep->unregister_work);
-		if (beep->enabled)
+		if (beep->dev)
 			snd_hda_do_detach(beep);
 		codec->beep = NULL;
 		kfree(beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 9cfdb771928c..f98b47cd6cfb 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1086,11 +1086,6 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-	/* audio codec should override the mixer name */
-	if (codec->afg || !*codec->bus->card->mixername)
-		snprintf(codec->bus->card->mixername,
-			 sizeof(codec->bus->card->mixername),
-			 "%s %s", codec->vendor_name, codec->chip_name);
 
 	if (is_generic_config(codec)) {
 		err = snd_hda_parse_generic_codec(codec);
@@ -1109,6 +1104,11 @@ int snd_hda_codec_configure(struct hda_codec *codec)
  patched:
 	if (!err && codec->patch_ops.unsol_event)
 		err = init_unsol_queue(codec->bus);
+	/* audio codec should override the mixer name */
+	if (!err && (codec->afg || !*codec->bus->card->mixername))
+		snprintf(codec->bus->card->mixername,
+			 sizeof(codec->bus->card->mixername),
+			 "%s %s", codec->vendor_name, codec->chip_name);
 	return err;
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
@@ -1327,11 +1327,13 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
  */
 u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 {
-	u32 pincap = snd_hda_query_pin_caps(codec, nid);
-
-	if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
-		snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+	u32 pincap;
 
+	if (!codec->no_trigger_sense) {
+		pincap = snd_hda_query_pin_caps(codec, nid);
+		if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
+			snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+	}
 	return snd_hda_codec_read(codec, nid, 0,
 				  AC_VERB_GET_PIN_SENSE, 0);
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 1d541b7f5547..0a770a28e71f 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -817,6 +817,7 @@ struct hda_codec {
 	unsigned int pin_amp_workaround:1; /* pin out-amp takes index
 					    * (e.g. Conexant codecs)
 					    */
+	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int power_transition :1; /* power-state in transition */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9b56f937913e..ff6da6f386d1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -356,6 +356,7 @@ struct azx_dev {
 					 */
 	unsigned char stream_tag;	/* assigned stream */
 	unsigned char index;		/* stream index */
+	int device;			/* last device number assigned to */
 
 	unsigned int opened :1;
 	unsigned int running :1;
@@ -425,6 +426,7 @@ struct azx {
 
 	/* flags */
 	int position_fix;
+	int poll_count;
 	unsigned int running :1;
 	unsigned int initialized :1;
 	unsigned int single_cmd :1;
@@ -505,7 +507,7 @@ static char *driver_short_names[] __devinitdata = {
 #define get_azx_dev(substream) (substream->runtime->private_data)
 
 static int azx_acquire_irq(struct azx *chip, int do_disconnect);
-
+static int azx_send_cmd(struct hda_bus *bus, unsigned int val);
 /*
  * Interface for HD codec
  */
@@ -663,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 {
 	struct azx *chip = bus->private_data;
 	unsigned long timeout;
+	int do_poll = 0;
 
  again:
 	timeout = jiffies + msecs_to_jiffies(1000);
 	for (;;) {
-		if (chip->polling_mode) {
+		if (chip->polling_mode || do_poll) {
 			spin_lock_irq(&chip->reg_lock);
 			azx_update_rirb(chip);
 			spin_unlock_irq(&chip->reg_lock);
@@ -675,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		if (!chip->rirb.cmds[addr]) {
 			smp_rmb();
 			bus->rirb_error = 0;
+
+			if (!do_poll)
+				chip->poll_count = 0;
 			return chip->rirb.res[addr]; /* the last value */
 		}
 		if (time_after(jiffies, timeout))
@@ -687,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 	}
 
+	if (!chip->polling_mode && chip->poll_count < 2) {
+		snd_printdd(SFX "azx_get_response timeout, "
+			   "polling the codec once: last cmd=0x%08x\n",
+			   chip->last_cmd[addr]);
+		do_poll = 1;
+		chip->poll_count++;
+		goto again;
+	}
+
+
 	if (!chip->polling_mode) {
 		snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
 			   "switching to polling mode: last cmd=0x%08x\n",
@@ -1441,10 +1457,13 @@ static int __devinit azx_codec_configure(struct azx *chip)
  */
 
 /* assign a stream for the PCM */
-static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
+static inline struct azx_dev *
+azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
 {
 	int dev, i, nums;
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+	struct azx_dev *res = NULL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		dev = chip->playback_index_offset;
 		nums = chip->playback_streams;
 	} else {
@@ -1453,10 +1472,15 @@ static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
 	}
 	for (i = 0; i < nums; i++, dev++)
 		if (!chip->azx_dev[dev].opened) {
-			chip->azx_dev[dev].opened = 1;
-			return &chip->azx_dev[dev];
+			res = &chip->azx_dev[dev];
+			if (res->device == substream->pcm->device)
+				break;
 		}
-	return NULL;
+	if (res) {
+		res->opened = 1;
+		res->device = substream->pcm->device;
+	}
+	return res;
 }
 
 /* release the assigned stream */
@@ -1505,7 +1529,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	int err;
 
 	mutex_lock(&chip->open_mutex);
-	azx_dev = azx_assign_device(chip, substream->stream);
+	azx_dev = azx_assign_device(chip, substream);
 	if (azx_dev == NULL) {
 		mutex_unlock(&chip->open_mutex);
 		return -EBUSY;
@@ -1869,6 +1893,9 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 
 	if (!bdl_pos_adj[chip->dev_index])
 		return 1; /* no delayed ack */
+	if (WARN_ONCE(!azx_dev->period_bytes,
+		      "hda-intel: zero azx_dev->period_bytes"))
+		return 0; /* this shouldn't happen! */
 	if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
 		return 0; /* NG - it's below the period boundary */
 	return 1; /* OK, it's fine */
@@ -2034,7 +2061,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
 {
 	if (request_irq(chip->pci->irq, azx_interrupt,
 			chip->msi ? 0 : IRQF_SHARED,
-			"HDA Intel", chip)) {
+			"hda_intel", chip)) {
 		printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
 		       "disabling device\n", chip->pci->irq);
 		if (do_disconnect)
@@ -2322,6 +2349,8 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
  * white/black-list for enable_msi
  */
 static struct snd_pci_quirk msi_black_list[] __devinitdata = {
+	SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
+	SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
 	{}
 };
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1a36137e13ec..69a941c7b158 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1186,6 +1186,8 @@ static int patch_ad1986a(struct hda_codec *codec)
 	 */
 	spec->multiout.no_share_stream = 1;
 
+	codec->no_trigger_sense = 1;
+
 	return 0;
 }
 
@@ -1371,6 +1373,8 @@ static int patch_ad1983(struct hda_codec *codec)
 
 	codec->patch_ops = ad198x_patch_ops;
 
+	codec->no_trigger_sense = 1;
+
 	return 0;
 }
 
@@ -1813,6 +1817,9 @@ static int patch_ad1981(struct hda_codec *codec)
 		codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
 		break;
 	}
+
+	codec->no_trigger_sense = 1;
+
 	return 0;
 }
 
@@ -3118,6 +3125,8 @@ static int patch_ad1988(struct hda_codec *codec)
 #endif
 	spec->vmaster_nid = 0x04;
 
+	codec->no_trigger_sense = 1;
+
 	return 0;
 }
 
@@ -3330,6 +3339,8 @@ static int patch_ad1884(struct hda_codec *codec)
 
 	codec->patch_ops = ad198x_patch_ops;
 
+	codec->no_trigger_sense = 1;
+
 	return 0;
 }
 
@@ -4287,6 +4298,8 @@ static int patch_ad1884a(struct hda_codec *codec)
 		break;
 	}
 
+	codec->no_trigger_sense = 1;
+
 	return 0;
 }
 
@@ -4623,6 +4636,9 @@ static int patch_ad1882(struct hda_codec *codec)
 		spec->mixers[2] = ad1882_6stack_mixers;
 		break;
 	}
+
+	codec->no_trigger_sense = 1;
+
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 4b200da1bd18..fe0423c39598 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -66,6 +66,7 @@ struct cs_spec {
 /* available models */
 enum {
 	CS420X_MBP55,
+	CS420X_IMAC27,
 	CS420X_AUTO,
 	CS420X_MODELS
 };
@@ -827,7 +828,8 @@ static void cs_automute(struct hda_codec *codec)
 				    AC_VERB_SET_PIN_WIDGET_CONTROL,
 				    hp_present ? 0 : PIN_OUT);
 	}
-	if (spec->board_config == CS420X_MBP55) {
+	if (spec->board_config == CS420X_MBP55 ||
+	    spec->board_config == CS420X_IMAC27) {
 		unsigned int gpio = hp_present ? 0x02 : 0x08;
 		snd_hda_codec_write(codec, 0x01, 0,
 				    AC_VERB_SET_GPIO_DATA, gpio);
@@ -1069,12 +1071,14 @@ static int cs_parse_auto_config(struct hda_codec *codec)
 
 static const char *cs420x_models[CS420X_MODELS] = {
 	[CS420X_MBP55] = "mbp55",
+	[CS420X_IMAC27] = "imac27",
 	[CS420X_AUTO] = "auto",
 };
 
 
 static struct snd_pci_quirk cs420x_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
+	SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),
 	{} /* terminator */
 };
 
@@ -1097,8 +1101,23 @@ static struct cs_pincfg mbp55_pincfgs[] = {
 	{} /* terminator */
 };
 
+static struct cs_pincfg imac27_pincfgs[] = {
+	{ 0x09, 0x012b4050 },
+	{ 0x0a, 0x90100140 },
+	{ 0x0b, 0x90100142 },
+	{ 0x0c, 0x018b3020 },
+	{ 0x0d, 0x90a00110 },
+	{ 0x0e, 0x400000f0 },
+	{ 0x0f, 0x01cbe030 },
+	{ 0x10, 0x014be060 },
+	{ 0x12, 0x01ab9070 },
+	{ 0x15, 0x400000f0 },
+	{} /* terminator */
+};
+
 static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
 	[CS420X_MBP55] = mbp55_pincfgs,
+	[CS420X_IMAC27] = imac27_pincfgs,
 };
 
 static void fix_pincfg(struct hda_codec *codec, int model)
@@ -1128,6 +1147,7 @@ static int patch_cs420x(struct hda_codec *codec)
 		fix_pincfg(codec, spec->board_config);
 
 	switch (spec->board_config) {
+	case CS420X_IMAC27:
 	case CS420X_MBP55:
 		/* GPIO1 = headphones */
 		/* GPIO3 = speakers */
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a09c03c3f62b..c578c28f368e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -29,6 +29,7 @@
 
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_beep.h"
 
 #define CXT_PIN_DIR_IN              0x00
 #define CXT_PIN_DIR_OUT             0x01
@@ -111,6 +112,7 @@ struct conexant_spec {
 	unsigned int dell_automute;
 	unsigned int port_d_mode;
 	unsigned char ext_mic_bias;
+	unsigned int dell_vostro;
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -476,6 +478,7 @@ static void conexant_free(struct hda_codec *codec)
 		snd_array_free(&spec->jacks);
 	}
 #endif
+	snd_hda_detach_beep_device(codec);
 	kfree(codec->spec);
 }
 
@@ -2109,9 +2112,12 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	int val;
+	hda_nid_t nid = kcontrol->private_value & 0xff;
+	int inout = (kcontrol->private_value & 0x100) ?
+		AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT;
 
-	val = snd_hda_codec_read(codec, 0x17, 0,
-		AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT);
+	val = snd_hda_codec_read(codec, nid, 0,
+		AC_VERB_GET_AMP_GAIN_MUTE, inout);
 
 	ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN;
 	return 0;
@@ -2123,6 +2129,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
 	unsigned int idx;
+	hda_nid_t nid = kcontrol->private_value & 0xff;
+	int inout = (kcontrol->private_value & 0x100) ?
+		AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT;
 
 	if (!imux->num_items)
 		return 0;
@@ -2130,9 +2139,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
 	if (idx >= imux->num_items)
 		idx = imux->num_items - 1;
 
-	snd_hda_codec_write_cache(codec, 0x17, 0,
+	snd_hda_codec_write_cache(codec, nid, 0,
 		AC_VERB_SET_AMP_GAIN_MUTE,
-		AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT |
+		AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout |
 			imux->items[idx].index);
 
 	return 1;
@@ -2201,10 +2210,11 @@ static struct snd_kcontrol_new cxt5066_mixers[] = {
 
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Analog Mic Boost Capture Enum",
+		.name = "Ext Mic Boost Capture Enum",
 		.info = cxt5066_mic_boost_mux_enum_info,
 		.get = cxt5066_mic_boost_mux_enum_get,
 		.put = cxt5066_mic_boost_mux_enum_put,
+		.private_value = 0x17,
 	},
 
 	HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others),
@@ -2212,6 +2222,19 @@ static struct snd_kcontrol_new cxt5066_mixers[] = {
 	{}
 };
 
+static struct snd_kcontrol_new cxt5066_vostro_mixers[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Int Mic Boost Capture Enum",
+		.info = cxt5066_mic_boost_mux_enum_info,
+		.get = cxt5066_mic_boost_mux_enum_get,
+		.put = cxt5066_mic_boost_mux_enum_put,
+		.private_value = 0x23 | 0x100,
+	},
+	HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
+	{}
+};
+
 static struct hda_verb cxt5066_init_verbs[] = {
 	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
@@ -2397,11 +2420,16 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
 /* initialize jack-sensing, too */
 static int cxt5066_init(struct hda_codec *codec)
 {
+	struct conexant_spec *spec = codec->spec;
+
 	snd_printdd("CXT5066: init\n");
 	conexant_init(codec);
 	if (codec->patch_ops.unsol_event) {
 		cxt5066_hp_automute(codec);
-		cxt5066_automic(codec);
+		if (spec->dell_vostro)
+			cxt5066_vostro_automic(codec);
+		else
+			cxt5066_automic(codec);
 	}
 	return 0;
 }
@@ -2500,7 +2528,10 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->init_verbs[0] = cxt5066_init_verbs_vostro;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+		spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers;
 		spec->port_d_mode = 0;
+		spec->dell_vostro = 1;
+		snd_hda_attach_beep_device(codec, 0x13);
 
 		/* no S/PDIF out */
 		spec->multiout.dig_out_nid = 0;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index aeed4cc5aa79..da34095c707f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -131,8 +131,8 @@ enum {
 enum {
 	ALC269_BASIC,
 	ALC269_QUANTA_FL1,
-	ALC269_ASUS_EEEPC_P703,
-	ALC269_ASUS_EEEPC_P901,
+	ALC269_ASUS_AMIC,
+	ALC269_ASUS_DMIC,
 	ALC269_FUJITSU,
 	ALC269_LIFEBOOK,
 	ALC269_AUTO,
@@ -188,6 +188,8 @@ enum {
 	ALC663_ASUS_MODE4,
 	ALC663_ASUS_MODE5,
 	ALC663_ASUS_MODE6,
+	ALC663_ASUS_MODE7,
+	ALC663_ASUS_MODE8,
 	ALC272_DELL,
 	ALC272_DELL_ZM1,
 	ALC272_SAMSUNG_NC10,
@@ -335,6 +337,9 @@ struct alc_spec {
 	/* hooks */
 	void (*init_hook)(struct hda_codec *codec);
 	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	void (*power_hook)(struct hda_codec *codec, int power);
+#endif
 
 	/* for pin sensing */
 	unsigned int sense_updated: 1;
@@ -386,6 +391,7 @@ struct alc_config_preset {
 	void (*init_hook)(struct hda_codec *);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	struct hda_amp_list *loopbacks;
+	void (*power_hook)(struct hda_codec *codec, int power);
 #endif
 };
 
@@ -898,6 +904,7 @@ static void setup_preset(struct hda_codec *codec,
 	spec->unsol_event = preset->unsol_event;
 	spec->init_hook = preset->init_hook;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
+	spec->power_hook = preset->power_hook;
 	spec->loopback.amplist = preset->loopbacks;
 #endif
 
@@ -1086,6 +1093,16 @@ static void alc889_coef_init(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
 }
 
+/* turn on/off EAPD control (only if available) */
+static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
+{
+	if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+		return;
+	if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+				    on ? 2 : 0);
+}
+
 static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
 	unsigned int tmp;
@@ -1103,25 +1120,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
 	case ALC_INIT_DEFAULT:
 		switch (codec->vendor_id) {
 		case 0x10ec0260:
-			snd_hda_codec_write(codec, 0x0f, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
-			snd_hda_codec_write(codec, 0x10, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
+			set_eapd(codec, 0x0f, 1);
+			set_eapd(codec, 0x10, 1);
 			break;
 		case 0x10ec0262:
 		case 0x10ec0267:
 		case 0x10ec0268:
 		case 0x10ec0269:
+		case 0x10ec0270:
 		case 0x10ec0272:
 		case 0x10ec0660:
 		case 0x10ec0662:
 		case 0x10ec0663:
 		case 0x10ec0862:
 		case 0x10ec0889:
-			snd_hda_codec_write(codec, 0x14, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
-			snd_hda_codec_write(codec, 0x15, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
+			set_eapd(codec, 0x14, 1);
+			set_eapd(codec, 0x15, 1);
 			break;
 		}
 		switch (codec->vendor_id) {
@@ -1223,6 +1237,8 @@ static void alc_init_auto_mic(struct hda_codec *codec)
 			return; /* invalid entry */
 		}
 	}
+	if (!ext || !fixed)
+		return;
 	if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
 		return; /* no unsol support */
 	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
@@ -1663,9 +1679,6 @@ static struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
 /*  some bit here disables the other DACs. Init=0x4900 */
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x08},
 	{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
-/* Enable amplifiers */
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
 /* DMIC fix
  * This laptop has a stereo digital microphone. The mics are only 1cm apart
  * which makes the stereo useless. However, either the mic or the ALC889
@@ -1778,6 +1791,25 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+	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),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+
 static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1808,6 +1840,14 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x1b;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void alc889_power_eapd(struct hda_codec *codec, int power)
+{
+	set_eapd(codec, 0x14, power);
+	set_eapd(codec, 0x15, power);
+}
+#endif
+
 /*
  * ALC880 3-stack model
  *
@@ -3601,12 +3641,29 @@ static void alc_free(struct hda_codec *codec)
 	snd_hda_detach_beep_device(codec);
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int alc_suspend(struct hda_codec *codec, pm_message_t state)
+{
+	struct alc_spec *spec = codec->spec;
+	if (spec && spec->power_hook)
+		spec->power_hook(codec, 0);
+	return 0;
+}
+#endif
+
 #ifdef SND_HDA_NEEDS_RESUME
 static int alc_resume(struct hda_codec *codec)
 {
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	struct alc_spec *spec = codec->spec;
+#endif
 	codec->patch_ops.init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	if (spec && spec->power_hook)
+		spec->power_hook(codec, 1);
+#endif
 	return 0;
 }
 #endif
@@ -3623,6 +3680,7 @@ static struct hda_codec_ops alc_patch_ops = {
 	.resume = alc_resume,
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
+	.suspend = alc_suspend,
 	.check_power_status = alc_check_power_status,
 #endif
 };
@@ -4761,6 +4819,49 @@ static void fixup_automic_adc(struct hda_codec *codec)
 	spec->auto_mic = 0; /* disable auto-mic to be sure */
 }
 
+/* choose the ADC/MUX containing the input pin and initialize the setup */
+static void fixup_single_adc(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t pin;
+	int i;
+
+	/* search for the input pin; there must be only one */
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		if (spec->autocfg.input_pins[i]) {
+			pin = spec->autocfg.input_pins[i];
+			break;
+		}
+	}
+	if (!pin)
+		return;
+
+	/* set the default connection to that pin */
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		hda_nid_t cap = spec->capsrc_nids ?
+			spec->capsrc_nids[i] : spec->adc_nids[i];
+		int idx;
+
+		idx = get_connection_index(codec, cap, pin);
+		if (idx < 0)
+			continue;
+		/* use only this ADC */
+		if (spec->capsrc_nids)
+			spec->capsrc_nids += i;
+		spec->adc_nids += i;
+		spec->num_adc_nids = 1;
+		/* select or unmute this route */
+		if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+			snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+						 HDA_AMP_MUTE, 0);
+		} else {
+			snd_hda_codec_write_cache(codec, cap, 0,
+					  AC_VERB_SET_CONNECT_SEL, idx);
+		}
+		return;
+	}
+}
+
 static void set_capture_mixer(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -4773,14 +4874,15 @@ static void set_capture_mixer(struct hda_codec *codec)
 		  alc_capture_mixer3 },
 	};
 	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
-		int mux;
-		if (spec->auto_mic) {
-			mux = 0;
+		int mux = 0;
+		if (spec->auto_mic)
 			fixup_automic_adc(codec);
-		} else if (spec->input_mux && spec->input_mux->num_items > 1)
-			mux = 1;
-		else
-			mux = 0;
+		else if (spec->input_mux) {
+			if (spec->input_mux->num_items > 1)
+				mux = 1;
+			else if (spec->input_mux->num_items == 1)
+				fixup_single_adc(codec);
+		}
 		spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
 	}
 }
@@ -7043,8 +7145,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
 	HDA_BIND_MUTE   ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
-	HDA_BIND_MUTE   ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -7445,6 +7547,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
 	/* Front Mic pin: input vref at 80% */
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
@@ -7629,6 +7732,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 }
 
+static void alc885_mb5_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x14, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+	snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+
+}
+
+static void alc885_mb5_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	/* Headphone insertion or removal. */
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc885_mb5_automute(codec);
+}
+
 static void alc885_imac91_automute(struct hda_codec *codec)
 {
  	unsigned int present;
@@ -8919,7 +9043,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
 	SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
 	SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8  */
-	SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO),
 	SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
@@ -9075,6 +9199,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &mb5_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc885_mb5_unsol_event,
+		.init_hook = alc885_mb5_automute,
 	},
 	[ALC885_MACPRO] = {
 		.mixers = { alc882_macpro_mixer },
@@ -9282,6 +9408,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.dac_nids = alc883_dac_nids,
 		.adc_nids = alc883_adc_nids_alt,
 		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+		.capsrc_nids = alc883_capsrc_nids,
 		.dig_out_nid = ALC883_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
@@ -9351,6 +9478,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
 		.channel_mode = alc883_3ST_6ch_modes,
 		.need_dac_fix = 1,
+		.const_channel_count = 6,
 		.num_mux_defs =
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_2_capture_sources,
@@ -9378,10 +9506,11 @@ static struct alc_config_preset alc882_presets[] = {
 		.init_hook = alc_automute_amp,
 	},
 	[ALC888_ACER_ASPIRE_8930G] = {
-		.mixers = { alc888_base_mixer,
+		.mixers = { alc889_acer_aspire_8930g_mixer,
 				alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
-				alc889_acer_aspire_8930g_verbs },
+				alc889_acer_aspire_8930g_verbs,
+				alc889_eapd_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
@@ -9398,6 +9527,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.unsol_event = alc_automute_amp_unsol_event,
 		.setup = alc889_acer_aspire_8930g_setup,
 		.init_hook = alc_automute_amp,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+		.power_hook = alc889_power_eapd,
+#endif
 	},
 	[ALC888_ACER_ASPIRE_7730G] = {
 		.mixers = { alc883_3ST_6ch_mixer,
@@ -9428,6 +9560,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.dac_nids = alc883_dac_nids,
 		.adc_nids = alc883_adc_nids_alt,
 		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+		.capsrc_nids = alc883_capsrc_nids,
 		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
 		.channel_mode = alc883_sixstack_modes,
 		.input_mux = &alc883_capture_source,
@@ -9489,6 +9622,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.dac_nids = alc883_dac_nids,
 		.adc_nids = alc883_adc_nids_alt,
 		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+		.capsrc_nids = alc883_capsrc_nids,
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_lenovo_101e_capture_source,
@@ -9668,6 +9802,7 @@ static struct alc_config_preset alc882_presets[] = {
 			alc880_gpio1_init_verbs },
 		.adc_nids = alc883_adc_nids,
 		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+		.capsrc_nids = alc883_capsrc_nids,
 		.dac_nids = alc883_dac_nids,
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.channel_mode = alc889A_mb31_6ch_modes,
@@ -10248,7 +10383,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+	spec->autocfg.speaker_pins[0] = 0x14;
 }
 
 static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -10678,6 +10813,13 @@ static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
 	{}
 };
 
+static struct hda_verb alc262_lenovo_3000_init_verbs[] = {
+	/* Front Mic pin: input vref at 50% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{}
+};
+
 static struct hda_input_mux alc262_fujitsu_capture_source = {
 	.num_items = 3,
 	.items = {
@@ -11113,7 +11255,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 #define alc262_auto_create_input_ctls \
-	alc880_auto_create_input_ctls
+	alc882_auto_create_input_ctls
 
 /*
  * generic initialization of ADC, input mixers and output mixers
@@ -11652,9 +11794,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hp_t5735_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HP_RP5700] = {
 		.mixers = { alc262_hp_rp5700_mixer },
@@ -11720,7 +11862,8 @@ static struct alc_config_preset alc262_presets[] = {
 	[ALC262_LENOVO_3000] = {
 		.mixers = { alc262_lenovo_3000_mixer },
 		.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
-				alc262_lenovo_3000_unsol_verbs },
+				alc262_lenovo_3000_unsol_verbs,
+				alc262_lenovo_3000_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
 		.dac_nids = alc262_dac_nids,
 		.hp_nid = 0x03,
@@ -12404,6 +12547,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x02;
 		break;
 	case 0x15:
+	case 0x21:
 		dac = 0x03;
 		break;
 	default:
@@ -12857,7 +13001,7 @@ static int patch_alc268(struct hda_codec *codec)
 	int board_config;
 	int i, has_beep, err;
 
-	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
 
@@ -13232,10 +13376,12 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
 /* toggle speaker-output according to the hp-jack state */
 static void alc269_speaker_automute(struct hda_codec *codec)
 {
+	struct alc_spec *spec = codec->spec;
+	unsigned int nid = spec->autocfg.hp_pins[0];
 	unsigned int present;
 	unsigned char bits;
 
-	present = snd_hda_jack_detect(codec, 0x15);
+	present = snd_hda_jack_detect(codec, nid);
 	bits = present ? AMP_IN_MUTE(0) : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
 				AMP_IN_MUTE(0), bits);
@@ -13460,8 +13606,8 @@ static void alc269_auto_init(struct hda_codec *codec)
 static const char *alc269_models[ALC269_MODEL_LAST] = {
 	[ALC269_BASIC]			= "basic",
 	[ALC269_QUANTA_FL1]		= "quanta",
-	[ALC269_ASUS_EEEPC_P703]	= "eeepc-p703",
-	[ALC269_ASUS_EEEPC_P901]	= "eeepc-p901",
+	[ALC269_ASUS_AMIC]		= "asus-amic",
+	[ALC269_ASUS_DMIC]		= "asus-dmic",
 	[ALC269_FUJITSU]		= "fujitsu",
 	[ALC269_LIFEBOOK]		= "lifebook",
 	[ALC269_AUTO]			= "auto",
@@ -13470,18 +13616,41 @@ static const char *alc269_models[ALC269_MODEL_LAST] = {
 static struct snd_pci_quirk alc269_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
 	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
-		      ALC269_ASUS_EEEPC_P703),
-        SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_EEEPC_P703),
-        SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_EEEPC_P703),
-        SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_EEEPC_P703),
-        SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_EEEPC_P703),
-        SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_EEEPC_P703),
-        SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_EEEPC_P703),
+		      ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80JT", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
-		      ALC269_ASUS_EEEPC_P901),
+		      ALC269_ASUS_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
-		      ALC269_ASUS_EEEPC_P901),
-        SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_EEEPC_P901),
+		      ALC269_ASUS_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_ASUS_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_ASUS_DMIC),
 	SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
 	{}
@@ -13511,7 +13680,7 @@ static struct alc_config_preset alc269_presets[] = {
 		.setup = alc269_quanta_fl1_setup,
 		.init_hook = alc269_quanta_fl1_init_hook,
 	},
-	[ALC269_ASUS_EEEPC_P703] = {
+	[ALC269_ASUS_AMIC] = {
 		.mixers = { alc269_eeepc_mixer },
 		.cap_mixer = alc269_epc_capture_mixer,
 		.init_verbs = { alc269_init_verbs,
@@ -13525,7 +13694,7 @@ static struct alc_config_preset alc269_presets[] = {
 		.setup = alc269_eeepc_amic_setup,
 		.init_hook = alc269_eeepc_inithook,
 	},
-	[ALC269_ASUS_EEEPC_P901] = {
+	[ALC269_ASUS_DMIC] = {
 		.mixers = { alc269_eeepc_mixer },
 		.cap_mixer = alc269_epc_capture_mixer,
 		.init_verbs = { alc269_init_verbs,
@@ -14763,6 +14932,8 @@ static int patch_alc861(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc861_pcm_digital_playback;
 	spec->stream_digital_capture = &alc861_pcm_digital_capture;
 
+	if (!spec->cap_mixer)
+		set_capture_mixer(codec);
 	set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
 
 	spec->vmaster_nid = 0x03;
@@ -15401,7 +15572,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0);
 }
 
 
@@ -16160,6 +16331,52 @@ static struct snd_kcontrol_new alc663_g50v_mixer[] = {
 	{ } /* end */
 };
 
+static struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
+	.ops = &snd_hda_bind_sw,
+	.values = {
+		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+static struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
+	.ops = &snd_hda_bind_sw,
+	.values = {
+		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+static struct snd_kcontrol_new alc663_mode7_mixer[] = {
+	HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+	HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+	HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+	HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct snd_kcontrol_new alc663_mode8_mixer[] = {
+	HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+	HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+	HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+	HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+
 static struct snd_kcontrol_new alc662_chmode_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -16447,6 +16664,45 @@ static struct hda_verb alc272_dell_init_verbs[] = {
 	{}
 };
 
+static struct hda_verb alc663_mode7_init_verbs[] = {
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
+static struct hda_verb alc663_mode8_init_verbs[] = {
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
 static struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
@@ -16626,6 +16882,54 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
 	}
 }
 
+static void alc663_two_hp_m7_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present1, present2;
+
+	present1 = snd_hda_codec_read(codec, 0x1b, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+	present2 = snd_hda_codec_read(codec, 0x21, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+
+	if (present1 || present2) {
+		snd_hda_codec_write_cache(codec, 0x14, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+		snd_hda_codec_write_cache(codec, 0x17, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+	} else {
+		snd_hda_codec_write_cache(codec, 0x14, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		snd_hda_codec_write_cache(codec, 0x17, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	}
+}
+
+static void alc663_two_hp_m8_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present1, present2;
+
+	present1 = snd_hda_codec_read(codec, 0x21, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+	present2 = snd_hda_codec_read(codec, 0x15, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+
+	if (present1 || present2) {
+		snd_hda_codec_write_cache(codec, 0x14, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+		snd_hda_codec_write_cache(codec, 0x17, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+	} else {
+		snd_hda_codec_write_cache(codec, 0x14, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		snd_hda_codec_write_cache(codec, 0x17, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	}
+}
+
 static void alc663_m51va_unsol_event(struct hda_codec *codec,
 					   unsigned int res)
 {
@@ -16645,7 +16949,7 @@ static void alc663_m51va_setup(struct hda_codec *codec)
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
-	spec->int_mic.mux_idx = 1;
+	spec->int_mic.mux_idx = 9;
 	spec->auto_mic = 1;
 }
 
@@ -16657,7 +16961,17 @@ static void alc663_m51va_inithook(struct hda_codec *codec)
 
 /* ***************** Mode1 ******************************/
 #define alc663_mode1_unsol_event	alc663_m51va_unsol_event
-#define alc663_mode1_setup		alc663_m51va_setup
+
+static void alc663_mode1_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
+}
+
 #define alc663_mode1_inithook		alc663_m51va_inithook
 
 /* ***************** Mode2 ******************************/
@@ -16674,7 +16988,7 @@ static void alc662_mode2_unsol_event(struct hda_codec *codec,
 	}
 }
 
-#define alc662_mode2_setup	alc663_m51va_setup
+#define alc662_mode2_setup	alc663_mode1_setup
 
 static void alc662_mode2_inithook(struct hda_codec *codec)
 {
@@ -16695,7 +17009,7 @@ static void alc663_mode3_unsol_event(struct hda_codec *codec,
 	}
 }
 
-#define alc663_mode3_setup	alc663_m51va_setup
+#define alc663_mode3_setup	alc663_mode1_setup
 
 static void alc663_mode3_inithook(struct hda_codec *codec)
 {
@@ -16716,7 +17030,7 @@ static void alc663_mode4_unsol_event(struct hda_codec *codec,
 	}
 }
 
-#define alc663_mode4_setup	alc663_m51va_setup
+#define alc663_mode4_setup	alc663_mode1_setup
 
 static void alc663_mode4_inithook(struct hda_codec *codec)
 {
@@ -16737,7 +17051,7 @@ static void alc663_mode5_unsol_event(struct hda_codec *codec,
 	}
 }
 
-#define alc663_mode5_setup	alc663_m51va_setup
+#define alc663_mode5_setup	alc663_mode1_setup
 
 static void alc663_mode5_inithook(struct hda_codec *codec)
 {
@@ -16758,7 +17072,7 @@ static void alc663_mode6_unsol_event(struct hda_codec *codec,
 	}
 }
 
-#define alc663_mode6_setup	alc663_m51va_setup
+#define alc663_mode6_setup	alc663_mode1_setup
 
 static void alc663_mode6_inithook(struct hda_codec *codec)
 {
@@ -16766,6 +17080,50 @@ static void alc663_mode6_inithook(struct hda_codec *codec)
 	alc_mic_automute(codec);
 }
 
+/* ***************** Mode7 ******************************/
+static void alc663_mode7_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_two_hp_m7_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc_mic_automute(codec);
+		break;
+	}
+}
+
+#define alc663_mode7_setup	alc663_mode1_setup
+
+static void alc663_mode7_inithook(struct hda_codec *codec)
+{
+	alc663_two_hp_m7_speaker_automute(codec);
+	alc_mic_automute(codec);
+}
+
+/* ***************** Mode8 ******************************/
+static void alc663_mode8_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_two_hp_m8_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc_mic_automute(codec);
+		break;
+	}
+}
+
+#define alc663_mode8_setup	alc663_m51va_setup
+
+static void alc663_mode8_inithook(struct hda_codec *codec)
+{
+	alc663_two_hp_m8_speaker_automute(codec);
+	alc_mic_automute(codec);
+}
+
 static void alc663_g71v_hp_automute(struct hda_codec *codec)
 {
 	unsigned int present;
@@ -16900,6 +17258,8 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
 	[ALC663_ASUS_MODE4] = "asus-mode4",
 	[ALC663_ASUS_MODE5] = "asus-mode5",
 	[ALC663_ASUS_MODE6] = "asus-mode6",
+	[ALC663_ASUS_MODE7] = "asus-mode7",
+	[ALC663_ASUS_MODE8] = "asus-mode8",
 	[ALC272_DELL]		= "dell",
 	[ALC272_DELL_ZM1]	= "dell-zm1",
 	[ALC272_SAMSUNG_NC10]	= "samsung-nc10",
@@ -16916,12 +17276,22 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7),
+	SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7),
+	SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8),
+	SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
 	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
 	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
 	SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
 	SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
@@ -16960,7 +17330,7 @@ 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(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
 	SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
 		      ALC662_3ST_6ch_DIG),
@@ -17205,6 +17575,36 @@ static struct alc_config_preset alc662_presets[] = {
 		.setup = alc663_mode6_setup,
 		.init_hook = alc663_mode6_inithook,
 	},
+	[ALC663_ASUS_MODE7] = {
+		.mixers = { alc663_mode7_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
+		.init_verbs = { alc662_init_verbs,
+				alc663_mode7_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
+		.hp_nid = 0x03,
+		.dac_nids = alc662_dac_nids,
+		.dig_out_nid = ALC662_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+		.channel_mode = alc662_3ST_2ch_modes,
+		.unsol_event = alc663_mode7_unsol_event,
+		.setup = alc663_mode7_setup,
+		.init_hook = alc663_mode7_inithook,
+	},
+	[ALC663_ASUS_MODE8] = {
+		.mixers = { alc663_mode8_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
+		.init_verbs = { alc662_init_verbs,
+				alc663_mode8_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
+		.hp_nid = 0x03,
+		.dac_nids = alc662_dac_nids,
+		.dig_out_nid = ALC662_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+		.channel_mode = alc662_3ST_2ch_modes,
+		.unsol_event = alc663_mode8_unsol_event,
+		.setup = alc663_mode8_setup,
+		.init_hook = alc663_mode8_inithook,
+	},
 	[ALC272_DELL] = {
 		.mixers = { alc663_m51va_mixer },
 		.cap_mixer = alc272_auto_capture_mixer,
@@ -17688,7 +18088,9 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
 	{ .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
 	{ .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 },
+	{ .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 },
 	{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
+	{ .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
 	  .patch = patch_alc861 },
 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3d59f8325848..799ba2570902 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2104,6 +2104,7 @@ static unsigned int ref9205_pin_configs[12] = {
     10280204
     1028021F
     10280228 (Dell Vostro 1500)
+    10280229 (Dell Vostro 1700)
 */
 static unsigned int dell_9205_m42_pin_configs[12] = {
 	0x0321101F, 0x03A11020, 0x400003FA, 0x90170310,
@@ -2189,6 +2190,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 		      "Dell Inspiron", STAC_9205_DELL_M44),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
 		      "Dell Vostro 1500", STAC_9205_DELL_M42),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0229,
+		      "Dell Vostro 1700", STAC_9205_DELL_M42),
 	/* Gateway */
 	SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
 	SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
@@ -3779,15 +3782,16 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		err = snd_hda_attach_beep_device(codec, nid);
 		if (err < 0)
 			return err;
-		/* IDT/STAC codecs have linear beep tone parameter */
-		codec->beep->linear_tone = 1;
-		/* if no beep switch is available, make its own one */
-		caps = query_amp_caps(codec, nid, HDA_OUTPUT);
-		if (codec->beep &&
-		    !((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT)) {
-			err = stac92xx_beep_switch_ctl(codec);
-			if (err < 0)
-				return err;
+		if (codec->beep) {
+			/* IDT/STAC codecs have linear beep tone parameter */
+			codec->beep->linear_tone = 1;
+			/* if no beep switch is available, make its own one */
+			caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+			if (!(caps & AC_AMPCAP_MUTE)) {
+				err = stac92xx_beep_switch_ctl(codec);
+				if (err < 0)
+					return err;
+			}
 		}
 	}
 #endif
@@ -4449,14 +4453,7 @@ static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
 {
 	if (!nid)
 		return 0;
-	/* NOTE: we can't use snd_hda_jack_detect() here because STAC/IDT
-	 * codecs behave wrongly when SET_PIN_SENSE is triggered, although
-	 * the pincap gives TRIG_REQ bit.
-	 */
-	if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) &
-	    AC_PINSENSE_PRESENCE)
-		return 1;
-	return 0;
+	return snd_hda_jack_detect(codec, nid);
 }
 
 static void stac92xx_line_out_detect(struct hda_codec *codec,
@@ -4733,6 +4730,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 }
 
+static int hp_blike_system(u32 subsystem_id);
+
+static void set_hp_led_gpio(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	switch (codec->vendor_id) {
+	case 0x111d7608:
+		/* GPIO 0 */
+		spec->gpio_led = 0x01;
+		break;
+	case 0x111d7600:
+	case 0x111d7601:
+	case 0x111d7602:
+	case 0x111d7603:
+		/* GPIO 3 */
+		spec->gpio_led = 0x08;
+		break;
+	}
+}
+
 /*
  * This method searches for the mute LED GPIO configuration
  * provided as OEM string in SMBIOS. The format of that string
@@ -4744,6 +4761,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
  *
  * So, HP B-series like systems may have HP_Mute_LED_0 (current models)
  * or  HP_Mute_LED_0_3 (future models) OEM SMBIOS strings
+ *
+ *
+ * The dv-series laptops don't seem to have the HP_Mute_LED* strings in
+ * SMBIOS - at least the ones I have seen do not have them - which include
+ * my own system (HP Pavilion dv6-1110ax) and my cousin's
+ * HP Pavilion dv9500t CTO.
+ * Need more information on whether it is true across the entire series.
+ * -- kunal
  */
 static int find_mute_led_gpio(struct hda_codec *codec)
 {
@@ -4754,28 +4779,27 @@ static int find_mute_led_gpio(struct hda_codec *codec)
 		while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
 								NULL, dev))) {
 			if (sscanf(dev->name, "HP_Mute_LED_%d_%d",
-			      &spec->gpio_led_polarity,
-			      &spec->gpio_led) == 2) {
+				  &spec->gpio_led_polarity,
+				  &spec->gpio_led) == 2) {
 				spec->gpio_led = 1 << spec->gpio_led;
 				return 1;
 			}
 			if (sscanf(dev->name, "HP_Mute_LED_%d",
-			      &spec->gpio_led_polarity) == 1) {
-				switch (codec->vendor_id) {
-				case 0x111d7608:
-					/* GPIO 0 */
-					spec->gpio_led = 0x01;
-					return 1;
-				case 0x111d7600:
-				case 0x111d7601:
-				case 0x111d7602:
-				case 0x111d7603:
-					/* GPIO 3 */
-					spec->gpio_led = 0x08;
-					return 1;
-				}
+				  &spec->gpio_led_polarity) == 1) {
+				set_hp_led_gpio(codec);
+				return 1;
 			}
 		}
+
+		/*
+		 * Fallback case - if we don't find the DMI strings,
+		 * we statically set the GPIO - if not a B-series system.
+		 */
+		if (!hp_blike_system(codec->subsystem_id)) {
+			set_hp_led_gpio(codec);
+			spec->gpio_led_polarity = 1;
+			return 1;
+		}
 	}
 	return 0;
 }
@@ -4958,6 +4982,7 @@ static int patch_stac9200(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	spec->num_pins = ARRAY_SIZE(stac9200_pin_nids);
 	spec->pin_nids = stac9200_pin_nids;
@@ -5020,6 +5045,7 @@ static int patch_stac925x(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	spec->num_pins = ARRAY_SIZE(stac925x_pin_nids);
 	spec->pin_nids = stac925x_pin_nids;
@@ -5104,6 +5130,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	codec->slave_dig_outs = stac92hd73xx_slave_dig_outs;
 	spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids);
@@ -5251,6 +5278,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
 	spec->digbeep_nid = 0x21;
@@ -5414,6 +5442,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	codec->patch_ops = stac92xx_patch_ops;
 	spec->num_pins = STAC92HD71BXX_NUM_PINS;
@@ -5546,6 +5575,8 @@ again:
 	spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
 	spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
 
+	snd_printdd("Found board config: %d\n", spec->board_config);
+
 	switch (spec->board_config) {
 	case STAC_HP_M4:
 		/* enable internal microphone */
@@ -5657,6 +5688,7 @@ static int patch_stac922x(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	spec->num_pins = ARRAY_SIZE(stac922x_pin_nids);
 	spec->pin_nids = stac922x_pin_nids;
@@ -5760,6 +5792,7 @@ static int patch_stac927x(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	codec->slave_dig_outs = stac927x_slave_dig_outs;
 	spec->num_pins = ARRAY_SIZE(stac927x_pin_nids);
@@ -5894,6 +5927,7 @@ static int patch_stac9205(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	spec->num_pins = ARRAY_SIZE(stac9205_pin_nids);
 	spec->pin_nids = stac9205_pin_nids;
@@ -6049,6 +6083,7 @@ static int patch_stac9872(struct hda_codec *codec)
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
+	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
 	spec->pin_nids = stac9872_pin_nids;
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 765d7bd4c3d4..9e66f6d306f8 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -703,11 +703,13 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho
 {
 	unsigned char nvol;
 
-	if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
+	if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) {
 		nvol = 0;
-	else
+	} else {
 		nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) /
 								WM_VOL_MAX;
+		nvol += 0x1b;
+	}
 
 	wm_put(ice, index, nvol);
 	wm_put_nocache(ice, index, 0x180 | nvol);
@@ -778,7 +780,7 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
 	for (ch = 0; ch < 2; ch++) {
 		unsigned int vol = ucontrol->value.integer.value[ch];
 		if (vol > WM_VOL_MAX)
-			continue;
+			vol = WM_VOL_MAX;
 		vol |= spec->master[ch] & WM_VOL_MUTE;
 		if (vol != spec->master[ch]) {
 			int dac;
@@ -834,8 +836,8 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *
 	for (i = 0; i < voices; i++) {
 		unsigned int vol = ucontrol->value.integer.value[i];
 		if (vol > WM_VOL_MAX)
-			continue;
-		vol |= spec->vol[ofs+i];
+			vol = WM_VOL_MAX;
+		vol |= spec->vol[ofs+i] & WM_VOL_MUTE;
 		if (vol != spec->vol[ofs+i]) {
 			spec->vol[ofs+i] = vol;
 			idx  = WM_DAC_ATTEN + ofs + i;
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index b5ca02e2038c..e66ef2b69b5d 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate)
 				 rptr.retwords[2] != M &&
 				 rptr.retwords[3] != N &&
 				 i++ < MAX_WRITE_RETRY);
-			if (i == MAX_WRITE_RETRY) {
+			if (i > MAX_WRITE_RETRY) {
 				snd_printdd("sent samplerate %d: %d failed\n",
 					    *intdec, rate);
 				return -EIO;
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index d057e6489643..5cfa608823f7 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -51,7 +51,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s
 			return 0; /* already enough large */
 		vfree(runtime->dma_area);
 	}
-	runtime->dma_area = vmalloc_32(size);
+	runtime->dma_area = vmalloc_32_user(size);
 	if (! runtime->dma_area)
 		return -ENOMEM;
 	runtime->dma_bytes = size;
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 69bd0acc81c8..a1bbe16b7f96 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -102,6 +102,12 @@ static int ac97_soc_probe(struct platform_device *pdev)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
+	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+	if (ret < 0) {
+		printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n");
+		goto err;
+	}
+
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0)
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index b69861d52161..3ef16bbc8c83 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -470,7 +470,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642);
 
 static int __init ak4642_modinit(void)
 {
-	int ret;
+	int ret = 0;
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	ret = i2c_add_driver(&ak4642_i2c_driver);
 #endif
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index bbc72c2ddfca..81b8c9dfe7fc 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -191,6 +191,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream,
 	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
 
 	vra |= 0x1; /* enable variable rate audio */
+	vra &= ~0x4; /* disable SPDIF output */
 
 	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
 
@@ -221,22 +222,6 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream,
 	return stac9766_ac97_write(codec, reg, runtime->rate);
 }
 
-static int ac97_digital_trigger(struct snd_pcm_substream *substream,
-				int cmd, struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	unsigned short vra;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_STOP:
-		vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
-		vra &= !0x04;
-		stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
-		break;
-	}
-	return 0;
-}
-
 static int stac9766_set_bias_level(struct snd_soc_codec *codec,
 				   enum snd_soc_bias_level level)
 {
@@ -315,7 +300,6 @@ static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
 
 static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
 	.prepare = ac97_digital_prepare,
-	.trigger = ac97_digital_trigger,
 };
 
 struct snd_soc_dai stac9766_dai[] = {
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a9dc5fb54774..da589d8664d0 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 	u16 reg;
 
 	/* Sync reg_cache with the hardware */
-	for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
+	for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
 		u16 val = tlv320aic23_read_reg_cache(codec, reg);
 		tlv320aic23_write(codec, reg, val);
 	}
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ebbf11b653a4..718ef912e758 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		iface |= 0x3 << 8;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		iface |= 0x3 << 8;	/* lg not sure which mode */
+		iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 265e68c75df8..af8cb6995a1f 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -424,23 +424,23 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	/* filter coefficient */
 	switch (params_rate(params)) {
-	case SNDRV_PCM_RATE_8000:
+	case 8000:
 		adn |= 0x5 << 1;
 		break;
-	case SNDRV_PCM_RATE_11025:
+	case 11025:
 		adn |= 0x4 << 1;
 		break;
-	case SNDRV_PCM_RATE_16000:
+	case 16000:
 		adn |= 0x3 << 1;
 		break;
-	case SNDRV_PCM_RATE_22050:
+	case 22050:
 		adn |= 0x2 << 1;
 		break;
-	case SNDRV_PCM_RATE_32000:
+	case 32000:
 		adn |= 0x1 << 1;
 		break;
-	case SNDRV_PCM_RATE_44100:
-	case SNDRV_PCM_RATE_48000:
+	case 44100:
+	case 48000:
 		break;
 	}
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce5515e3f2b0..3595bd57c4eb 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev)
 	struct i2c_client *i2c = codec->control_data;
 	int i;
 	u16 *reg_cache = codec->reg_cache;
-	u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+	u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults),
 				 GFP_KERNEL);
 
 	/* Bring the codec back up to standby first to minimise pop/clicks */
@@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev)
 		for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
 			if (tmp_cache[i] != reg_cache[i])
 				snd_soc_write(codec, i, tmp_cache[i]);
+		kfree(tmp_cache);
 	} else {
 		dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
 	}
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 3d850b97037a..31e39ffd1d8e 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -378,23 +378,23 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
 		iface |= (1 << 9);
 
 	switch (params_rate(params)) {
-	case SNDRV_PCM_RATE_8000:
+	case 8000:
 		addcntrl |= (0x5 << 1);
 		break;
-	case SNDRV_PCM_RATE_11025:
+	case 11025:
 		addcntrl |= (0x4 << 1);
 		break;
-	case SNDRV_PCM_RATE_16000:
+	case 16000:
 		addcntrl |= (0x3 << 1);
 		break;
-	case SNDRV_PCM_RATE_22050:
+	case 22050:
 		addcntrl |= (0x2 << 1);
 		break;
-	case SNDRV_PCM_RATE_32000:
+	case 32000:
 		addcntrl |= (0x1 << 1);
 		break;
-	case SNDRV_PCM_RATE_44100:
-	case SNDRV_PCM_RATE_48000:
+	case 44100:
+	case 48000:
 		break;
 	}
 	ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl);
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 81c57b5c591c..8812751da8c9 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -47,7 +47,7 @@ static const u16 wm8974_reg[WM8974_CACHEREGNUM] = {
 };
 
 #define WM8974_POWER1_BIASEN  0x08
-#define WM8974_POWER1_BUFIOEN 0x10
+#define WM8974_POWER1_BUFIOEN 0x04
 
 struct wm8974_priv {
 	struct snd_soc_codec codec;
@@ -482,23 +482,23 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	/* filter coefficient */
 	switch (params_rate(params)) {
-	case SNDRV_PCM_RATE_8000:
+	case 8000:
 		adn |= 0x5 << 1;
 		break;
-	case SNDRV_PCM_RATE_11025:
+	case 11025:
 		adn |= 0x4 << 1;
 		break;
-	case SNDRV_PCM_RATE_16000:
+	case 16000:
 		adn |= 0x3 << 1;
 		break;
-	case SNDRV_PCM_RATE_22050:
+	case 22050:
 		adn |= 0x2 << 1;
 		break;
-	case SNDRV_PCM_RATE_32000:
+	case 32000:
 		adn |= 0x1 << 1;
 		break;
-	case SNDRV_PCM_RATE_44100:
-	case SNDRV_PCM_RATE_48000:
+	case 44100:
+	case 48000:
 		break;
 	}
 
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 0ac1215dcd9b..e237bf615129 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -463,7 +463,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 {
 	u16 *cache = codec->reg_cache;
 
-	soc_ac97_ops.write(codec->ac97, reg, val);
+	if (reg < 0x7c)
+		soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
 	if (reg < (ARRAY_SIZE(wm9712_reg)))
 		cache[reg] = val;
diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c
index 0267d2d91685..07d2a248438c 100644
--- a/sound/soc/imx/mx27vis_wm8974.c
+++ b/sound/soc/imx/mx27vis_wm8974.c
@@ -180,7 +180,8 @@ static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
 	/* disable the PLL */
-	return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0);
+	return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG,
+				       0, 0);
 }
 
 /*
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 3db8a6c523f4..19283e5edfbf 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 71b2c161158d..68980c19a3bc 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+	{"PCM DAC", NULL, "APLL Enable"},
 	{"Headphone Amplifier", NULL, "PCM DAC"},
 	{"Line Out", NULL, "PCM DAC"},
 	{"Headphone Jack", NULL, "Headphone Amplifier"},
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index c071f9603a38..3c85c0f92823 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -24,7 +24,7 @@
 
 #include <linux/clk.h>
 #include <linux/platform_device.h>
-#include <linux/i2c/twl4030.h>
+#include <linux/i2c/twl.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
@@ -321,11 +321,11 @@ static int __init sdp3430_soc_init(void)
 	*(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
 
 	/* Set TWL4030 GPIO6 as EXTMUTE signal */
-	twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
+	twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
 						TWL4030_INTBR_PMBR1);
 	pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
 	pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
-	twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
+	twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
 						TWL4030_INTBR_PMBR1);
 
 	ret = platform_device_add(sdp3430_snd_device);
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index c7af09729c6e..5263ab18f827 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -42,42 +42,12 @@ static struct snd_soc_device fsi_snd_devdata = {
 	.codec_dev	= &soc_codec_dev_ak4642,
 };
 
-#define AK4642_BUS 0
-#define AK4642_ADR 0x12
-static int ak4642_add_i2c_device(void)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = AK4642_ADR;
-	strlcpy(info.type, "ak4642", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(AK4642_BUS);
-	if (!adapter) {
-		printk(KERN_DEBUG "can't get i2c adapter\n");
-		return -ENODEV;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		printk(KERN_DEBUG "can't add i2c device\n");
-		return -ENODEV;
-	}
-
-	return 0;
-}
-
 static struct platform_device *fsi_snd_device;
 
 static int __init fsi_ak4642_init(void)
 {
 	int ret = -ENOMEM;
 
-	ak4642_add_i2c_device();
-
 	fsi_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!fsi_snd_device)
 		goto out;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 9c49c11c43ce..42813b808389 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -876,7 +876,7 @@ static int fsi_probe(struct platform_device *pdev)
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	irq = platform_get_irq(pdev, 0);
-	if (!res || !irq) {
+	if (!res || (int)irq <= 0) {
 		dev_err(&pdev->dev, "Not enough FSI platform resources.\n");
 		ret = -ENODEV;
 		goto exit;
diff --git a/sound/sound_core.c b/sound/sound_core.c
index dbca7c909a31..7c2d677a2df5 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void)
 	class_destroy(sound_class);
 }
 
-module_init(init_soundcore);
+subsys_initcall(init_soundcore);
 module_exit(cleanup_soundcore);
 
 
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index b074a594c595..9edef4684978 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -752,7 +752,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s
 			return 0; /* already large enough */
 		vfree(runtime->dma_area);
 	}
-	runtime->dma_area = vmalloc(size);
+	runtime->dma_area = vmalloc_user(size);
 	if (!runtime->dma_area)
 		return -ENOMEM;
 	runtime->dma_bytes = size;
@@ -1936,7 +1936,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
 	struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
 	struct snd_usb_substream *subs = &as->substream[direction];
 
-	if (subs->interface >= 0) {
+	if (!as->chip->shutdown && subs->interface >= 0) {
 		usb_set_interface(subs->dev, subs->interface, 0);
 		subs->interface = -1;
 	}