[GIT PULL] sound fixes

[Takashi Iwai <tiwai@suse.de>]

Linus,

please pull sound fixes for v2.6.33-rc2 from:

  git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus

containing the following fixes.  Most of them are small individual
fixes.  A big chunk is for supporting a few new Realtek codec chips,
which are more or less compatible with older ones.


Thanks!

Takashi

===

Clemens Ladisch (1):
      sound: sgio2audio/pdaudiocf/usb-audio: initialize PCM buffer

Daniel T Chen (1):
      ALSA: hda: Set Front Mic to input vref 50% for Lenovo 3000 Y410

Einar Rünkaru (2):
      ALSA: hda - Fixed internal mic initialization for Dell Vostro 1015
      ALSA: hda - Make use of beep device found in Dell Vostro 1015n

Guennadi Liakhovetski (1):
      ASoC: wm8974: fix a wrong bit definition

Hector Martin (3):
      ALSA: HDA: simplify Aspire 8930G verb array
      ALSA: HDA: remove useless mixers on Aspire 8930G
      ALSA: HDA: add powersaving hook for Realtek

Jaroslav Kysela (1):
      ALSA: hda/realtek: Remove extra .capsrc_nids initialization for ALC889_INTEL

Jon Smirl (1):
      ASoC: Fix disable of SPDIF on STAC9766 codec

Julia Lawall (1):
      ALSA: Use kzalloc for allocating only one thing

Kailang Yang (1):
      ALSA: hda - More ALC663 fixes and support of compatible chips

Krzysztof Helt (2):
      ALSA: fix incorrect rounding direction in snd_interval_ratnum()
      ALSA: sbawe: fix memory detection

Kuninori Morimoto (1):
      ASoC: ak4642: Add default return value in ak4642_modinit

Roel Kluin (1):
      sound/oss/pss: Fix test of unsigned in pss_reset_dsp() and pss_download_boot()

Russell King (5):
      ALSA: AACI: simplify codec rate information
      ALSA: AACI: cleanup aaci_pcm_hw_params
      ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params
      ALSA: AACI: add double-rate support
      ALSA: AACI: switch to per-pcm locking

Takashi Iwai (3):
      ALSA: hda - Fix missing capsrc_nids for ALC88x
      ALSA: hda - Fix quirk for Maxdata obook4-1
      ALSA: aaci - Fix a typo

---
 sound/arm/aaci.c                       |  177 +++++----------
 sound/arm/aaci.h                       |    2 +-
 sound/core/pcm_lib.c                   |    4 +-
 sound/isa/msnd/msnd_midi.c             |    2 +-
 sound/isa/sb/emu8000.c                 |    6 +-
 sound/mips/sgio2audio.c                |    2 +-
 sound/oss/pss.c                        |    6 +-
 sound/pci/hda/patch_conexant.c         |   43 +++-
 sound/pci/hda/patch_realtek.c          |  387 +++++++++++++++++++++++++++++---
 sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c |    2 +-
 sound/soc/codecs/ak4642.c              |    2 +-
 sound/soc/codecs/stac9766.c            |   18 +--
 sound/soc/codecs/wm8974.c              |    2 +-
 sound/usb/usbaudio.c                   |    2 +-
 14 files changed, 467 insertions(+), 188 deletions(-)

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1497dce..c569986 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)
@@ -507,18 +450,22 @@ 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 +474,7 @@ 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	=
@@ -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;
 }
@@ -721,18 +664,10 @@ static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream,
 	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 +675,7 @@ 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 +690,7 @@ 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 +707,11 @@ 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 +740,7 @@ 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 +823,12 @@ 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 +941,6 @@ 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 +967,7 @@ 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 +1027,14 @@ 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 924f69c..6a4a2ee 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/pcm_lib.c b/sound/core/pcm_lib.c
index 30f4108..a27545b 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/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c
index cb9aa4c..4be562b 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 96678d5..751762f 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -393,8 +393,6 @@ size_dram(struct snd_emu8000 *emu)
 
 	while (size < EMU8000_MAX_DRAM) {
 
-		size += 512 * 1024;  /* increment 512kbytes */
-
 		/* Write a unique data on the test address.
 		 * if the address is out of range, the data is written on
 		 * 0x200000(=EMU8000_DRAM_OFFSET).  Then the id word is
@@ -414,7 +412,9 @@ 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 */
+
+		size += 512 * 1024;  /* increment 512kbytes */
 
 		snd_emu8000_read_wait(emu);
 
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 8691f4c..f1d9d16 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/pss.c b/sound/oss/pss.c
index 83f5ee2..e19dd5d 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/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a09c03c..c578c28 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 aeed4cc..c746505 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
 
@@ -1663,9 +1670,6 @@ 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 +1782,25 @@ 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 +1831,16 @@ 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)
+{
+	snd_hda_codec_write(codec, 0x14, 0,
+			    AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+	snd_hda_codec_write(codec, 0x15, 0,
+			    AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+}
+#endif
+
 /*
  * ALC880 3-stack model
  *
@@ -3601,12 +3634,29 @@ 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 +3673,7 @@ 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
 };
@@ -8919,7 +8970,7 @@ 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),
@@ -9282,6 +9333,7 @@ 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,
@@ -9378,10 +9430,11 @@ 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 +9451,9 @@ 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 +9484,7 @@ 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 +9546,7 @@ 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 +9726,7 @@ 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,
@@ -10678,6 +10737,13 @@ 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 = {
@@ -11720,7 +11786,8 @@ 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,
@@ -12857,7 +12924,7 @@ 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 +13299,12 @@ 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 +13529,8 @@ 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 +13539,41 @@ 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 +13603,7 @@ 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 +13617,7 @@ 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,
@@ -16160,6 +16252,52 @@ 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 +16585,45 @@ 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 +16803,54 @@ 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 +16870,7 @@ 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 +16882,17 @@ 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 +16909,7 @@ 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 +16930,7 @@ 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 +16951,7 @@ 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 +16972,7 @@ 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 +16993,7 @@ 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 +17001,50 @@ 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 +17179,8 @@ 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 +17197,22 @@ 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),
@@ -17205,6 +17496,36 @@ 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 +18009,9 @@ 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/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index d057e64..5cfa608 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/ak4642.c b/sound/soc/codecs/ak4642.c
index b69861d..3ef16bb 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 bbc72c2..81b8c9d 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/wm8974.c b/sound/soc/codecs/wm8974.c
index 81c57b5..a808675 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;
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index b074a59..4963def 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;