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

45 files changed, 337 insertions, 312 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1497dce..656e474 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 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/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index b4b48af..5d94118 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -159,7 +159,7 @@ static int pxa2xx_ac97_resume(struct device *dev)
 	return ret;
 }
 
-static struct dev_pm_ops pxa2xx_ac97_pm_ops = {
+static const struct dev_pm_ops pxa2xx_ac97_pm_ops = {
 	.suspend	= pxa2xx_ac97_suspend,
 	.resume		= pxa2xx_ac97_resume,
 };
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index c15682a..475455c 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 30f4108..a27545b2 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 29ab46a1..25b0641 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 ca8068b..b01d948 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/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c
index 661205c..af888a0 100644
--- a/sound/isa/gus/gus_mem.c
+++ b/sound/isa/gus/gus_mem.c
@@ -127,7 +127,8 @@ static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc,
 	    !share_id[2] && !share_id[3])
 		return NULL;
 	for (block = alloc->first; block; block = block->next)
-		if (!memcmp(share_id, block->share_id, sizeof(share_id)))
+		if (!memcmp(share_id, block->share_id,
+				sizeof(block->share_id)))
 			return block;
 	return NULL;
 }
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/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
index fda7a94..ccc6422 100644
--- a/sound/pci/cs5535audio/Makefile
+++ b/sound/pci/cs5535audio/Makefile
@@ -4,9 +4,7 @@
 
 snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o
 snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o
-ifdef CONFIG_MGEODE_LX
 snd-cs5535audio-$(CONFIG_OLPC) += cs5535audio_olpc.o
-endif
 
 # Toplevel Module Dependency
 obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 05f56e0..91e7faf 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -389,6 +389,7 @@ probefail_out:
 
 static void __devexit snd_cs5535audio_remove(struct pci_dev *pci)
 {
+	olpc_quirks_cleanup();
 	snd_card_free(pci_get_drvdata(pci));
 	pci_set_drvdata(pci, NULL);
 }
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 7a298ac..51966d7 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -99,10 +99,11 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
 int snd_cs5535audio_resume(struct pci_dev *pci);
 #endif
 
-#if defined(CONFIG_OLPC) && defined(CONFIG_MGEODE_LX)
+#ifdef CONFIG_OLPC
 void __devinit olpc_prequirks(struct snd_card *card,
 		struct snd_ac97_template *ac97);
 int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97);
+void __devexit olpc_quirks_cleanup(void);
 void olpc_analog_input(struct snd_ac97 *ac97, int on);
 void olpc_mic_bias(struct snd_ac97 *ac97, int on);
 
@@ -128,6 +129,7 @@ static inline int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
 {
 	return 0;
 }
+static inline void olpc_quirks_cleanup(void) { }
 static inline void olpc_analog_input(struct snd_ac97 *ac97, int on) { }
 static inline void olpc_mic_bias(struct snd_ac97 *ac97, int on) { }
 static inline void olpc_capture_open(struct snd_ac97 *ac97) { }
diff --git a/sound/pci/cs5535audio/cs5535audio_olpc.c b/sound/pci/cs5535audio/cs5535audio_olpc.c
index 5c68143..50da49b 100644
--- a/sound/pci/cs5535audio/cs5535audio_olpc.c
+++ b/sound/pci/cs5535audio/cs5535audio_olpc.c
@@ -13,10 +13,13 @@
 #include <sound/info.h>
 #include <sound/control.h>
 #include <sound/ac97_codec.h>
+#include <linux/gpio.h>
 
 #include <asm/olpc.h>
 #include "cs5535audio.h"
 
+#define DRV_NAME "cs5535audio-olpc"
+
 /*
  * OLPC has an additional feature on top of the regular AD1888 codec features.
  * It has an Analog Input mode that is switched into (after disabling the
@@ -38,10 +41,7 @@ void olpc_analog_input(struct snd_ac97 *ac97, int on)
 	}
 
 	/* set Analog Input through GPIO */
-	if (on)
-		geode_gpio_set(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
-	else
-		geode_gpio_clear(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
+	gpio_set_value(OLPC_GPIO_MIC_AC, on);
 }
 
 /*
@@ -73,8 +73,7 @@ static int olpc_dc_info(struct snd_kcontrol *kctl,
 
 static int olpc_dc_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
 {
-	v->value.integer.value[0] = geode_gpio_isset(OLPC_GPIO_MIC_AC,
-			GPIO_OUTPUT_VAL);
+	v->value.integer.value[0] = gpio_get_value(OLPC_GPIO_MIC_AC);
 	return 0;
 }
 
@@ -153,6 +152,12 @@ int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
 	if (!machine_is_olpc())
 		return 0;
 
+	if (gpio_request(OLPC_GPIO_MIC_AC, DRV_NAME)) {
+		printk(KERN_ERR DRV_NAME ": unable to allocate MIC GPIO\n");
+		return -EIO;
+	}
+	gpio_direction_output(OLPC_GPIO_MIC_AC, 0);
+
 	/* drop the original AD1888 HPF control */
 	memset(&elem, 0, sizeof(elem));
 	elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
@@ -169,11 +174,18 @@ int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
 	for (i = 0; i < ARRAY_SIZE(olpc_cs5535audio_ctls); i++) {
 		err = snd_ctl_add(card, snd_ctl_new1(&olpc_cs5535audio_ctls[i],
 				ac97->private_data));
-		if (err < 0)
+		if (err < 0) {
+			gpio_free(OLPC_GPIO_MIC_AC);
 			return err;
+		}
 	}
 
 	/* turn off the mic by default */
 	olpc_mic_bias(ac97, 0);
 	return 0;
 }
+
+void __devexit olpc_quirks_cleanup(void)
+{
+	gpio_free(OLPC_GPIO_MIC_AC);
+}
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 5fe34a8..e4581a42 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 9cfdb77..f98b47c 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 1d541b7..0a770a2 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_hwdep.c b/sound/pci/hda/hda_hwdep.c
index d243286..40ccb41 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -24,6 +24,7 @@
 #include <linux/compat.h>
 #include <linux/mutex.h>
 #include <linux/ctype.h>
+#include <linux/string.h>
 #include <linux/firmware.h>
 #include <sound/core.h>
 #include "hda_codec.h"
@@ -428,8 +429,7 @@ static int parse_hints(struct hda_codec *codec, const char *buf)
 	char *key, *val;
 	struct hda_hint *hint;
 
-	while (isspace(*buf))
-		buf++;
+	buf = skip_spaces(buf);
 	if (!*buf || *buf == '#' || *buf == '\n')
 		return 0;
 	if (*buf == '=')
@@ -444,8 +444,7 @@ static int parse_hints(struct hda_codec *codec, const char *buf)
 		return -EINVAL;
 	}
 	*val++ = 0;
-	while (isspace(*val))
-		val++;
+	val = skip_spaces(val);
 	remove_trail_spaces(key);
 	remove_trail_spaces(val);
 	hint = get_hint(codec, key);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9b56f93..ec9c348 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;
@@ -1441,10 +1442,13 @@ 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 +1457,15 @@ 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 +1514,7 @@ 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;
@@ -2322,6 +2331,7 @@ 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 */
 	{}
 };
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1a36137..69a941c 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 4b200da..fe0423c 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_realtek.c b/sound/pci/hda/patch_realtek.c
index 2d3f4f8..c746505 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -337,6 +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;
@@ -388,6 +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
 };
 
@@ -900,6 +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
 
@@ -1665,9 +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
@@ -1780,6 +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;
@@ -1810,6 +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
  *
@@ -3603,12 +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
@@ -3625,6 +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
 };
@@ -9381,10 +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),
@@ -9401,6 +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,
@@ -10684,6 +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 = {
@@ -11726,7 +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,
@@ -12863,7 +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;
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3d59f83..2291a83 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,
@@ -4958,6 +4955,7 @@ 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 +5018,7 @@ 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 +5103,7 @@ 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 +5251,7 @@ 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 +5415,7 @@ 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;
@@ -5657,6 +5659,7 @@ 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 +5763,7 @@ 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 +5898,7 @@ 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 +6054,7 @@ 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/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 7717e01..edaa729 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -143,7 +143,8 @@ static int snd_pdacf_probe(struct pcmcia_device *link)
 	link->io.NumPorts1 = 16;
 
 	link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_FORCED_PULSE;
-	// link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING|IRQ_FIRST_SHARED;
+	/* FIXME: This driver should be updated to allow for dynamic IRQ sharing */
+	/* link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING | IRQ_FORCED_PULSE; */
 
 	link->irq.Handler = pdacf_interrupt;
 	link->conf.Attributes = CONF_ENABLE_IRQ;
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/ac97.c b/sound/soc/codecs/ac97.c
index 69bd0ac..a1bbe16 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 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/twl4030.c b/sound/soc/codecs/twl4030.c
index 5f1681f..2a27f7b 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -26,7 +26,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.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/pcm_params.h>
@@ -175,7 +175,7 @@ static int twl4030_write(struct snd_soc_codec *codec,
 {
 	twl4030_write_reg_cache(codec, reg, value);
 	if (likely(reg < TWL4030_REG_SW_SHADOW))
-		return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
+		return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
 					    reg);
 	else
 		return 0;
@@ -261,7 +261,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 	do {
 		/* this takes a little while, so don't slam i2c */
 		udelay(2000);
-		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+		twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
 				    TWL4030_REG_ANAMICL);
 	} while ((i++ < 100) &&
 		 ((byte & TWL4030_CNCL_OFFSET_START) ==
@@ -542,7 +542,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w,		\
 		break;							\
 	case SND_SOC_DAPM_POST_PMD:					\
 		reg_val = twl4030_read_reg_cache(w->codec, reg);	\
-		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,	\
+		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,	\
 					reg_val & (~mask),		\
 					reg);				\
 		break;							\
@@ -679,7 +679,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 		mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
 			twl4030->sysclk) + 1);
 		/* Bypass the reg_cache to mute the headset */
-		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
 					hs_gain & (~0x0f),
 					TWL4030_REG_HS_GAIN_SET);
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f82125d..ebbf11b 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1340,9 +1340,10 @@ static int wm8350_resume(struct platform_device *pdev)
 	return 0;
 }
 
-static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
+static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
 {
 	struct wm8350_data *priv = data;
+	struct wm8350 *wm8350 = priv->codec.control_data;
 	u16 reg;
 	int report;
 	int mask;
@@ -1365,7 +1366,7 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
 
 	if (!jack->jack) {
 		dev_warn(wm8350->dev, "Jack interrupt called with no jack\n");
-		return;
+		return IRQ_NONE;
 	}
 
 	/* Debounce */
@@ -1378,6 +1379,8 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
 		report = 0;
 
 	snd_soc_jack_report(jack->jack, report, jack->report);
+
+	return IRQ_HANDLED;
 }
 
 /**
@@ -1421,9 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 	wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
 
 	/* Sync status */
-	wm8350_hp_jack_handler(wm8350, irq, priv);
-
-	wm8350_unmask_irq(wm8350, irq);
+	wm8350_hp_jack_handler(irq, priv);
 
 	return 0;
 }
@@ -1482,12 +1483,16 @@ static int wm8350_probe(struct platform_device *pdev)
 	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
 			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
 
-	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
-	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+	/* Make sure jack detect is disabled to start off with */
+	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
+			  WM8350_JDL_ENA | WM8350_JDR_ENA);
+
 	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
-			    wm8350_hp_jack_handler, priv);
+			    wm8350_hp_jack_handler, 0, "Left jack detect",
+			    priv);
 	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
-			    wm8350_hp_jack_handler, priv);
+			    wm8350_hp_jack_handler, 0, "Right jack detect",
+			    priv);
 
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
@@ -1516,8 +1521,6 @@ static int wm8350_remove(struct platform_device *pdev)
 			  WM8350_JDL_ENA | WM8350_JDR_ENA);
 	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
 
-	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
-	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
 	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
 	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 265e68c..af8cb69 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/wm8900.c b/sound/soc/codecs/wm8900.c
index c9438dd..dbc368c 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -199,7 +199,7 @@ static void wm8900_reset(struct snd_soc_codec *codec)
 	snd_soc_write(codec, WM8900_REG_RESET, 0);
 
 	memcpy(codec->reg_cache, wm8900_reg_defaults,
-	       sizeof(codec->reg_cache));
+	       sizeof(wm8900_reg_defaults));
 }
 
 static int wm8900_hp_event(struct snd_soc_dapm_widget *w,
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 3d850b9..31e39ff 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 81c57b5..8812751 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 0ac1215..e237bf6 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 0267d2d..07d2a24 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/sdp3430.c b/sound/soc/omap/sdp3430.c
index c071f96..3c85c0f 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/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
index d441c3b..4984754 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -312,7 +312,7 @@ int simtec_audio_resume(struct device *dev)
 	return 0;
 }
 
-struct dev_pm_ops simtec_audio_pmops = {
+const struct dev_pm_ops simtec_audio_pmops = {
 	.resume	= simtec_audio_resume,
 };
 EXPORT_SYMBOL_GPL(simtec_audio_pmops);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h
index 2714203..e18faee 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec.h
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.h
@@ -15,7 +15,7 @@ extern int simtec_audio_core_probe(struct platform_device *pdev,
 extern int simtec_audio_remove(struct platform_device *pdev);
 
 #ifdef CONFIG_PM
-extern struct dev_pm_ops simtec_audio_pmops;
+extern const struct dev_pm_ops simtec_audio_pmops;
 #define simtec_audio_pm &simtec_audio_pmops
 #else
 #define simtec_audio_pm NULL
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index c7af097..5263ab1 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 9c49c11..42813b8 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/soc/soc-core.c b/sound/soc/soc-core.c
index ef8f282..0a6440c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1236,7 +1236,7 @@ static int soc_poweroff(struct device *dev)
 	return 0;
 }
 
-static struct dev_pm_ops soc_pm_ops = {
+static const struct dev_pm_ops soc_pm_ops = {
 	.suspend = soc_suspend,
 	.resume = soc_resume,
 	.poweroff = soc_poweroff,
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;