diff --git a/[refs] b/[refs]
index 20d189247cbe..96ab5e20bf86 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: 0f101fa6bccdc797ddcee40fd4c55e879155a40f
+refs/heads/master: c4dc5071859bf666a5a9d6565f16c51a261a88b7
diff --git a/trunk/include/sound/core.h b/trunk/include/sound/core.h
index 7e5589472681..e5eec5f73502 100644
--- a/trunk/include/sound/core.h
+++ b/trunk/include/sound/core.h
@@ -388,13 +388,9 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
 
 #else /* !CONFIG_SND_DEBUG */
 
-#define snd_printd(fmt, args...)	do { } while (0)
-#define snd_BUG()			do { } while (0)
-static inline int __snd_bug_on(void)
-{
-	return 0;
-}
-#define snd_BUG_ON(cond)		__snd_bug_on()  /* always false */
+#define snd_printd(fmt, args...)	/* nothing */
+#define snd_BUG()			/* nothing */
+#define snd_BUG_ON(cond)	({/*(void)(cond);*/ 0;})  /* always false */
 
 #endif /* CONFIG_SND_DEBUG */
 
diff --git a/trunk/include/sound/soc-dapm.h b/trunk/include/sound/soc-dapm.h
index ca699a3017f3..c1b26fcc0b5c 100644
--- a/trunk/include/sound/soc-dapm.h
+++ b/trunk/include/sound/soc-dapm.h
@@ -240,7 +240,6 @@ int snd_soc_dapm_sys_add(struct device *dev);
 /* dapm audio pin control and status */
 int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_sync(struct snd_soc_codec *codec);
 
diff --git a/trunk/sound/aoa/soundbus/core.c b/trunk/sound/aoa/soundbus/core.c
index fa8ab2815a98..f84f3e505788 100644
--- a/trunk/sound/aoa/soundbus/core.c
+++ b/trunk/sound/aoa/soundbus/core.c
@@ -176,7 +176,7 @@ int soundbus_add_one(struct soundbus_dev *dev)
 		return -EINVAL;
 	}
 
-	dev_set_name(&dev->ofdev.dev, "soundbus:%x", ++devcount);
+	snprintf(dev->ofdev.dev.bus_id, BUS_ID_SIZE, "soundbus:%x", ++devcount);
 	dev->ofdev.dev.bus = &soundbus_bus_type;
 	return of_device_register(&dev->ofdev);
 }
diff --git a/trunk/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/trunk/sound/aoa/soundbus/i2sbus/i2sbus-core.c
index b4590df07466..e6beb92c6933 100644
--- a/trunk/sound/aoa/soundbus/i2sbus/i2sbus-core.c
+++ b/trunk/sound/aoa/soundbus/i2sbus/i2sbus-core.c
@@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	struct i2sbus_dev *dev;
 	struct device_node *child = NULL, *sound = NULL;
 	struct resource *r;
-	int i, layout = 0, rlen, ok = force;
+	int i, layout = 0, rlen;
 	static const char *rnames[] = { "i2sbus: %s (control)",
 					"i2sbus: %s (tx)",
 					"i2sbus: %s (rx)" };
@@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 			layout = *layout_id;
 			snprintf(dev->sound.modalias, 32,
 				 "sound-layout-%d", layout);
-			ok = 1;
+			force = 1;
 		}
 	}
 	/* for the time being, until we can handle non-layout-id
@@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	 * When there are two i2s busses and only one has a layout-id,
 	 * then this depends on the order, but that isn't important
 	 * either as the second one in that case is just a modem. */
-	if (!ok) {
+	if (!force) {
 		kfree(dev);
 		return -ENODEV;
 	}
diff --git a/trunk/sound/arm/pxa2xx-ac97-lib.c b/trunk/sound/arm/pxa2xx-ac97-lib.c
index 34c1d94f921e..99026dfb81ea 100644
--- a/trunk/sound/arm/pxa2xx-ac97-lib.c
+++ b/trunk/sound/arm/pxa2xx-ac97-lib.c
@@ -50,7 +50,7 @@ unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 	mutex_lock(&car_mutex);
 
 	/* set up primary or secondary codec space */
-	if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS)
+	if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS)
 		reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
 	else
 		reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
@@ -90,7 +90,7 @@ void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	mutex_lock(&car_mutex);
 
 	/* set up primary or secondary codec space */
-	if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS)
+	if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS)
 		reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
 	else
 		reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
@@ -200,7 +200,7 @@ static inline void pxa_ac97_cold_pxa3xx(void)
 bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
 {
 #ifdef CONFIG_PXA25x
-	if (cpu_is_pxa25x())
+	if (cpu_is_pxa21x() || cpu_is_pxa25x())
 		pxa_ac97_warm_pxa25x();
 	else
 #endif
@@ -230,7 +230,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset);
 bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
 {
 #ifdef CONFIG_PXA25x
-	if (cpu_is_pxa25x())
+	if (cpu_is_pxa21x() || cpu_is_pxa25x())
 		pxa_ac97_cold_pxa25x();
 	else
 #endif
@@ -301,7 +301,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend);
 
 int pxa2xx_ac97_hw_resume(void)
 {
-	if (cpu_is_pxa25x() || cpu_is_pxa27x()) {
+	if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) {
 		pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
 		pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
 		pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
@@ -325,7 +325,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
 	if (ret < 0)
 		goto err;
 
-	if (cpu_is_pxa25x() || cpu_is_pxa27x()) {
+	if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) {
 		pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
 		pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
 		pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
diff --git a/trunk/sound/arm/pxa2xx-pcm-lib.c b/trunk/sound/arm/pxa2xx-pcm-lib.c
index 75a0d746fb60..1c93eb77cb99 100644
--- a/trunk/sound/arm/pxa2xx-pcm-lib.c
+++ b/trunk/sound/arm/pxa2xx-pcm-lib.c
@@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 		goto out;
 
 	ret = -ENOMEM;
-	rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
+	rtd = kmalloc(sizeof(*rtd), GFP_KERNEL);
 	if (!rtd)
 		goto out;
 	rtd->dma_desc_array =
diff --git a/trunk/sound/core/control.c b/trunk/sound/core/control.c
index b0bf42691047..6d71f9a7ccbb 100644
--- a/trunk/sound/core/control.c
+++ b/trunk/sound/core/control.c
@@ -225,13 +225,8 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol,
 	kctl.id.iface = ncontrol->iface;
 	kctl.id.device = ncontrol->device;
 	kctl.id.subdevice = ncontrol->subdevice;
-	if (ncontrol->name) {
+	if (ncontrol->name)
 		strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name));
-		if (strcmp(ncontrol->name, kctl.id.name) != 0)
-			snd_printk(KERN_WARNING
-				   "Control name '%s' truncated to '%s'\n",
-				   ncontrol->name, kctl.id.name);
-	}
 	kctl.id.index = ncontrol->index;
 	kctl.count = ncontrol->count ? ncontrol->count : 1;
 	access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
diff --git a/trunk/sound/core/jack.c b/trunk/sound/core/jack.c
index bd2d9e6b55e9..8133a2b173a5 100644
--- a/trunk/sound/core/jack.c
+++ b/trunk/sound/core/jack.c
@@ -147,9 +147,6 @@ EXPORT_SYMBOL(snd_jack_set_parent);
  */
 void snd_jack_report(struct snd_jack *jack, int status)
 {
-	if (!jack)
-		return;
-
 	if (jack->type & SND_JACK_HEADPHONE)
 		input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT,
 				    status & SND_JACK_HEADPHONE);
diff --git a/trunk/sound/core/pcm_lib.c b/trunk/sound/core/pcm_lib.c
index 921691080f35..6ea5cfb83998 100644
--- a/trunk/sound/core/pcm_lib.c
+++ b/trunk/sound/core/pcm_lib.c
@@ -908,12 +908,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond,
 EXPORT_SYMBOL(snd_pcm_hw_rule_add);
 
 /**
- * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint
+ * snd_pcm_hw_constraint_mask
  * @runtime: PCM runtime instance
  * @var: hw_params variable to apply the mask
  * @mask: the bitmap mask
  *
- * Apply the constraint of the given bitmap mask to a 32-bit mask parameter.
+ * Apply the constraint of the given bitmap mask to a mask parameter.
  */
 int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
 			       u_int32_t mask)
@@ -928,12 +928,12 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param
 }
 
 /**
- * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint
+ * snd_pcm_hw_constraint_mask64
  * @runtime: PCM runtime instance
  * @var: hw_params variable to apply the mask
  * @mask: the 64bit bitmap mask
  *
- * Apply the constraint of the given bitmap mask to a 64-bit mask parameter.
+ * Apply the constraint of the given bitmap mask to a mask parameter.
  */
 int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
 				 u_int64_t mask)
@@ -949,7 +949,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par
 }
 
 /**
- * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval
+ * snd_pcm_hw_constraint_integer
  * @runtime: PCM runtime instance
  * @var: hw_params variable to apply the integer constraint
  *
@@ -964,7 +964,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa
 EXPORT_SYMBOL(snd_pcm_hw_constraint_integer);
 
 /**
- * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval
+ * snd_pcm_hw_constraint_minmax
  * @runtime: PCM runtime instance
  * @var: hw_params variable to apply the range
  * @min: the minimal value
@@ -995,7 +995,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params,
 
 
 /**
- * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter
+ * snd_pcm_hw_constraint_list
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the list constraint
@@ -1031,7 +1031,7 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
 }
 
 /**
- * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter
+ * snd_pcm_hw_constraint_ratnums
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the ratnums constraint
@@ -1064,7 +1064,7 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
 }
 
 /**
- * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter
+ * snd_pcm_hw_constraint_ratdens
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the ratdens constraint
@@ -1095,7 +1095,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params,
 }
 
 /**
- * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule
+ * snd_pcm_hw_constraint_msbits
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @width: sample bits width
@@ -1123,7 +1123,7 @@ static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params,
 }
 
 /**
- * snd_pcm_hw_constraint_step - add a hw constraint step rule
+ * snd_pcm_hw_constraint_step
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the step constraint
@@ -1154,7 +1154,7 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm
 }		
 
 /**
- * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule
+ * snd_pcm_hw_constraint_pow2
  * @runtime: PCM runtime instance
  * @cond: condition bits
  * @var: hw_params variable to apply the power-of-2 constraint
@@ -1202,13 +1202,13 @@ void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params)
 EXPORT_SYMBOL(_snd_pcm_hw_params_any);
 
 /**
- * snd_pcm_hw_param_value - return @params field @var value
+ * snd_pcm_hw_param_value
  * @params: the hw_params instance
  * @var: parameter to retrieve
- * @dir: pointer to the direction (-1,0,1) or %NULL
+ * @dir: pointer to the direction (-1,0,1) or NULL
  *
- * Return the value for field @var if it's fixed in configuration space
- * defined by @params. Return -%EINVAL otherwise.
+ * Return the value for field PAR if it's fixed in configuration space 
+ *  defined by PARAMS. Return -EINVAL otherwise
  */
 int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params,
 			   snd_pcm_hw_param_t var, int *dir)
@@ -1271,13 +1271,13 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params,
 
 
 /**
- * snd_pcm_hw_param_first - refine config space and return minimum value
+ * snd_pcm_hw_param_first
  * @pcm: PCM instance
  * @params: the hw_params instance
  * @var: parameter to retrieve
- * @dir: pointer to the direction (-1,0,1) or %NULL
+ * @dir: pointer to the direction (-1,0,1) or NULL
  *
- * Inside configuration space defined by @params remove from @var all
+ * Inside configuration space defined by PARAMS remove from PAR all 
  * values > minimum. Reduce configuration space accordingly.
  * Return the minimum.
  */
@@ -1317,13 +1317,13 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params,
 
 
 /**
- * snd_pcm_hw_param_last - refine config space and return maximum value
+ * snd_pcm_hw_param_last
  * @pcm: PCM instance
  * @params: the hw_params instance
  * @var: parameter to retrieve
- * @dir: pointer to the direction (-1,0,1) or %NULL
+ * @dir: pointer to the direction (-1,0,1) or NULL
  *
- * Inside configuration space defined by @params remove from @var all
+ * Inside configuration space defined by PARAMS remove from PAR all 
  * values < maximum. Reduce configuration space accordingly.
  * Return the maximum.
  */
@@ -1345,11 +1345,11 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm,
 EXPORT_SYMBOL(snd_pcm_hw_param_last);
 
 /**
- * snd_pcm_hw_param_choose - choose a configuration defined by @params
+ * snd_pcm_hw_param_choose
  * @pcm: PCM instance
  * @params: the hw_params instance
  *
- * Choose one configuration from configuration space defined by @params.
+ * Choose one configuration from configuration space defined by PARAMS
  * The configuration chosen is that obtained fixing in this order:
  * first access, first format, first subformat, min channels,
  * min rate, min period time, max buffer size, min tick time
diff --git a/trunk/sound/core/pcm_misc.c b/trunk/sound/core/pcm_misc.c
index ea2bf82c9373..89b7f549bebd 100644
--- a/trunk/sound/core/pcm_misc.c
+++ b/trunk/sound/core/pcm_misc.c
@@ -319,7 +319,6 @@ EXPORT_SYMBOL(snd_pcm_format_physical_width);
 /**
  * snd_pcm_format_size - return the byte size of samples on the given format
  * @format: the format to check
- * @samples: sampling rate
  *
  * Returns the byte size of the given samples for the format, or a
  * negative error code if unknown format.
diff --git a/trunk/sound/core/pcm_native.c b/trunk/sound/core/pcm_native.c
index aef18682c035..e61e12506ded 100644
--- a/trunk/sound/core/pcm_native.c
+++ b/trunk/sound/core/pcm_native.c
@@ -875,8 +875,10 @@ static struct action_ops snd_pcm_action_start = {
 };
 
 /**
- * snd_pcm_start - start all linked streams
+ * snd_pcm_start
  * @substream: the PCM substream instance
+ *
+ * Start all linked streams.
  */
 int snd_pcm_start(struct snd_pcm_substream *substream)
 {
@@ -924,11 +926,12 @@ static struct action_ops snd_pcm_action_stop = {
 };
 
 /**
- * snd_pcm_stop - try to stop all running streams in the substream group
+ * snd_pcm_stop
  * @substream: the PCM substream instance
  * @state: PCM state after stopping the stream
  *
- * The state of each stream is then changed to the given state unconditionally.
+ * Try to stop all running streams in the substream group.
+ * The state of each stream is changed to the given value after that unconditionally.
  */
 int snd_pcm_stop(struct snd_pcm_substream *substream, int state)
 {
@@ -938,10 +941,11 @@ int snd_pcm_stop(struct snd_pcm_substream *substream, int state)
 EXPORT_SYMBOL(snd_pcm_stop);
 
 /**
- * snd_pcm_drain_done - stop the DMA only when the given stream is playback
+ * snd_pcm_drain_done
  * @substream: the PCM substream
  *
- * After stopping, the state is changed to SETUP.
+ * Stop the DMA only when the given stream is playback.
+ * The state is changed to SETUP.
  * Unlike snd_pcm_stop(), this affects only the given stream.
  */
 int snd_pcm_drain_done(struct snd_pcm_substream *substream)
@@ -1061,9 +1065,10 @@ static struct action_ops snd_pcm_action_suspend = {
 };
 
 /**
- * snd_pcm_suspend - trigger SUSPEND to all linked streams
+ * snd_pcm_suspend
  * @substream: the PCM substream
  *
+ * Trigger SUSPEND to all linked streams.
  * After this call, all streams are changed to SUSPENDED state.
  */
 int snd_pcm_suspend(struct snd_pcm_substream *substream)
@@ -1083,9 +1088,10 @@ int snd_pcm_suspend(struct snd_pcm_substream *substream)
 EXPORT_SYMBOL(snd_pcm_suspend);
 
 /**
- * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm
+ * snd_pcm_suspend_all
  * @pcm: the PCM instance
  *
+ * Trigger SUSPEND to all substreams in the given pcm.
  * After this call, all streams are changed to SUSPENDED state.
  */
 int snd_pcm_suspend_all(struct snd_pcm *pcm)
@@ -1307,9 +1313,11 @@ static struct action_ops snd_pcm_action_prepare = {
 };
 
 /**
- * snd_pcm_prepare - prepare the PCM substream to be triggerable
+ * snd_pcm_prepare
  * @substream: the PCM substream instance
  * @file: file to refer f_flags
+ *
+ * Prepare the PCM substream to be triggerable.
  */
 static int snd_pcm_prepare(struct snd_pcm_substream *substream,
 			   struct file *file)
diff --git a/trunk/sound/core/rawmidi.c b/trunk/sound/core/rawmidi.c
index 39672f68ce5d..c4995c9f5730 100644
--- a/trunk/sound/core/rawmidi.c
+++ b/trunk/sound/core/rawmidi.c
@@ -148,8 +148,6 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream)
 
 static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up)
 {
-	if (!substream->opened)
-		return;
 	if (up) {
 		tasklet_hi_schedule(&substream->runtime->tasklet);
 	} else {
@@ -160,8 +158,6 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs
 
 static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
 {
-	if (!substream->opened)
-		return;
 	substream->ops->trigger(substream, up);
 	if (!up && substream->runtime->event)
 		tasklet_kill(&substream->runtime->tasklet);
@@ -861,8 +857,6 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
 	int result = 0, count1;
 	struct snd_rawmidi_runtime *runtime = substream->runtime;
 
-	if (!substream->opened)
-		return -EBADFD;
 	if (runtime->buffer == NULL) {
 		snd_printd("snd_rawmidi_receive: input is not active!!!\n");
 		return -EINVAL;
@@ -1132,8 +1126,6 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count)
 int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
 			 unsigned char *buffer, int count)
 {
-	if (!substream->opened)
-		return -EBADFD;
 	count = snd_rawmidi_transmit_peek(substream, buffer, count);
 	if (count < 0)
 		return count;
diff --git a/trunk/sound/drivers/dummy.c b/trunk/sound/drivers/dummy.c
index 73be7e14a603..e5e749f3e0ef 100644
--- a/trunk/sound/drivers/dummy.c
+++ b/trunk/sound/drivers/dummy.c
@@ -51,7 +51,7 @@ static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime)
 	if (err < 0)
 		return err;
 	err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX);
-	if (err < 0)
+	if (err) < 0)
 		return err;
 	return 0;
 }
diff --git a/trunk/sound/drivers/ml403-ac97cr.c b/trunk/sound/drivers/ml403-ac97cr.c
index 7783843ca9ae..ecdbeb6d3603 100644
--- a/trunk/sound/drivers/ml403-ac97cr.c
+++ b/trunk/sound/drivers/ml403-ac97cr.c
@@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
 	/* get irq */
 	irq = platform_get_irq(pfdev, 0);
 	if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
-			dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
+			pfdev->dev.bus_id, (void *)ml403_ac97cr)) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
 			   "unable to grab IRQ %d\n",
 			   irq);
@@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
 		   ml403_ac97cr->irq);
 	irq = platform_get_irq(pfdev, 1);
 	if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
-			dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
+			pfdev->dev.bus_id, (void *)ml403_ac97cr)) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
 			   "unable to grab IRQ %d\n",
 			   irq);
diff --git a/trunk/sound/drivers/pcsp/pcsp_input.c b/trunk/sound/drivers/pcsp/pcsp_input.c
index 0444cdeb4bec..cd9b83e7f7d1 100644
--- a/trunk/sound/drivers/pcsp/pcsp_input.c
+++ b/trunk/sound/drivers/pcsp/pcsp_input.c
@@ -24,13 +24,13 @@ static void pcspkr_do_sound(unsigned int count)
 	spin_lock_irqsave(&i8253_lock, flags);
 
 	if (count) {
+		/* enable counter 2 */
+		outb_p(inb_p(0x61) | 3, 0x61);
 		/* set command for counter 2, 2 byte write */
 		outb_p(0xB6, 0x43);
 		/* select desired HZ */
 		outb_p(count & 0xff, 0x42);
 		outb((count >> 8) & 0xff, 0x42);
-		/* enable counter 2 */
-		outb_p(inb_p(0x61) | 3, 0x61);
 	} else {
 		/* disable counter 2 */
 		outb(inb_p(0x61) & 0xFC, 0x61);
diff --git a/trunk/sound/isa/Kconfig b/trunk/sound/isa/Kconfig
index ce0aa044e274..660beb41f767 100644
--- a/trunk/sound/isa/Kconfig
+++ b/trunk/sound/isa/Kconfig
@@ -211,7 +211,7 @@ config SND_GUSCLASSIC
 
 config SND_GUSEXTREME
 	tristate "Gravis UltraSound Extreme"
-	select SND_OPL3_LIB
+	select SND_HWDEP
 	select SND_MPU401_UART
 	select SND_PCM
 	help
diff --git a/trunk/sound/isa/ad1848/ad1848.c b/trunk/sound/isa/ad1848/ad1848.c
index 223a6c038819..b68d20edc20f 100644
--- a/trunk/sound/isa/ad1848/ad1848.c
+++ b/trunk/sound/isa/ad1848/ad1848.c
@@ -70,15 +70,15 @@ static int __devinit snd_ad1848_match(struct device *dev, unsigned int n)
 		return 0;
 
 	if (port[n] == SNDRV_AUTO_PORT) {
-		dev_err(dev, "please specify port\n");
+		snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
 		return 0;
 	}
 	if (irq[n] == SNDRV_AUTO_IRQ) {
-		dev_err(dev, "please specify irq\n");
+		snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id);
 		return 0;	
 	}
 	if (dma1[n] == SNDRV_AUTO_DMA) {
-		dev_err(dev, "please specify dma1\n");
+		snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id);
 		return 0;
 	}
 	return 1;
diff --git a/trunk/sound/isa/adlib.c b/trunk/sound/isa/adlib.c
index 374b7177e111..efa8c80d05b6 100644
--- a/trunk/sound/isa/adlib.c
+++ b/trunk/sound/isa/adlib.c
@@ -36,7 +36,7 @@ static int __devinit snd_adlib_match(struct device *dev, unsigned int n)
 		return 0;
 
 	if (port[n] == SNDRV_AUTO_PORT) {
-		dev_err(dev, "please specify port\n");
+		snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
 		return 0;
 	}
 	return 1;
@@ -55,13 +55,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
 
 	card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
 	if (!card) {
-		dev_err(dev, "could not create card\n");
+		snd_printk(KERN_ERR "%s: could not create card\n", dev->bus_id);
 		return -EINVAL;
 	}
 
 	card->private_data = request_region(port[n], 4, CRD_NAME);
 	if (!card->private_data) {
-		dev_err(dev, "could not grab ports\n");
+		snd_printk(KERN_ERR "%s: could not grab ports\n", dev->bus_id);
 		error = -EBUSY;
 		goto out;
 	}
@@ -73,13 +73,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
 
 	error = snd_opl3_create(card, port[n], port[n] + 2, OPL3_HW_AUTO, 1, &opl3);
 	if (error < 0) {
-		dev_err(dev, "could not create OPL\n");
+		snd_printk(KERN_ERR "%s: could not create OPL\n", dev->bus_id);
 		goto out;
 	}
 
 	error = snd_opl3_hwdep_new(opl3, 0, 0, NULL);
 	if (error < 0) {
-		dev_err(dev, "could not create FM\n");
+		snd_printk(KERN_ERR "%s: could not create FM\n", dev->bus_id);
 		goto out;
 	}
 
@@ -87,7 +87,7 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
 
 	error = snd_card_register(card);
 	if (error < 0) {
-		dev_err(dev, "could not register card\n");
+		snd_printk(KERN_ERR "%s: could not register card\n", dev->bus_id);
 		goto out;
 	}
 
diff --git a/trunk/sound/isa/cs423x/cs4231.c b/trunk/sound/isa/cs423x/cs4231.c
index f019d449e2d6..ddd289120aa8 100644
--- a/trunk/sound/isa/cs423x/cs4231.c
+++ b/trunk/sound/isa/cs423x/cs4231.c
@@ -74,15 +74,15 @@ static int __devinit snd_cs4231_match(struct device *dev, unsigned int n)
 		return 0;
 
 	if (port[n] == SNDRV_AUTO_PORT) {
-		dev_err(dev, "please specify port\n");
+		snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
 		return 0;
 	}
 	if (irq[n] == SNDRV_AUTO_IRQ) {
-		dev_err(dev, "please specify irq\n");
+		snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id);
 		return 0;
 	}
 	if (dma1[n] == SNDRV_AUTO_DMA) {
-		dev_err(dev, "please specify dma1\n");
+		snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id);
 		return 0;
 	}
 	return 1;
@@ -133,7 +133,7 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n)
 					mpu_port[n], 0, mpu_irq[n],
 					mpu_irq[n] >= 0 ? IRQF_DISABLED : 0,
 					NULL) < 0)
-			dev_warn(dev, "MPU401 not detected\n");
+			printk(KERN_WARNING "%s: MPU401 not detected\n", dev->bus_id);
 	}
 
 	snd_card_set_dev(card, dev);
diff --git a/trunk/sound/isa/cs423x/cs4236.c b/trunk/sound/isa/cs423x/cs4236.c
index 019c9401663e..91f9c15d3e30 100644
--- a/trunk/sound/isa/cs423x/cs4236.c
+++ b/trunk/sound/isa/cs423x/cs4236.c
@@ -488,19 +488,19 @@ static int __devinit snd_cs423x_isa_match(struct device *pdev,
 		return 0;
 
 	if (port[dev] == SNDRV_AUTO_PORT) {
-		dev_err(pdev, "please specify port\n");
+		snd_printk(KERN_ERR "%s: please specify port\n", pdev->bus_id);
 		return 0;
 	}
 	if (cport[dev] == SNDRV_AUTO_PORT) {
-		dev_err(pdev, "please specify cport\n");
+		snd_printk(KERN_ERR "%s: please specify cport\n", pdev->bus_id);
 		return 0;
 	}
 	if (irq[dev] == SNDRV_AUTO_IRQ) {
-		dev_err(pdev, "please specify irq\n");
+		snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id);
 		return 0;
 	}
 	if (dma1[dev] == SNDRV_AUTO_DMA) {
-		dev_err(pdev, "please specify dma1\n");
+		snd_printk(KERN_ERR "%s: please specify dma1\n", pdev->bus_id);
 		return 0;
 	}
 	return 1;
diff --git a/trunk/sound/isa/es1688/es1688.c b/trunk/sound/isa/es1688/es1688.c
index b46377139cf8..f88639ea64b2 100644
--- a/trunk/sound/isa/es1688/es1688.c
+++ b/trunk/sound/isa/es1688/es1688.c
@@ -88,14 +88,16 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card,
 	if (irq[n] == SNDRV_AUTO_IRQ) {
 		irq[n] = snd_legacy_find_free_irq(possible_irqs);
 		if (irq[n] < 0) {
-			dev_err(dev, "unable to find a free IRQ\n");
+			snd_printk(KERN_ERR "%s: unable to find a free IRQ\n",
+				dev->bus_id);
 			return -EBUSY;
 		}
 	}
 	if (dma8[n] == SNDRV_AUTO_DMA) {
 		dma8[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma8[n] < 0) {
-			dev_err(dev, "unable to find a free DMA\n");
+			snd_printk(KERN_ERR "%s: unable to find a free DMA\n",
+				dev->bus_id);
 			return -EBUSY;
 		}
 	}
@@ -145,7 +147,8 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n)
 
 	if (snd_opl3_create(card, chip->port, chip->port + 2,
 			OPL3_HW_OPL3, 0, &opl3) < 0)
-		dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port);
+		printk(KERN_WARNING "%s: opl3 not detected at 0x%lx\n",
+			dev->bus_id, chip->port);
 	else {
 		error =	snd_opl3_hwdep_new(opl3, 0, 1, NULL);
 		if (error < 0)
diff --git a/trunk/sound/isa/gus/gusclassic.c b/trunk/sound/isa/gus/gusclassic.c
index 426532a4d730..8f914b37bf89 100644
--- a/trunk/sound/isa/gus/gusclassic.c
+++ b/trunk/sound/isa/gus/gusclassic.c
@@ -90,21 +90,24 @@ static int __devinit snd_gusclassic_create(struct snd_card *card,
 	if (irq[n] == SNDRV_AUTO_IRQ) {
 		irq[n] = snd_legacy_find_free_irq(possible_irqs);
 		if (irq[n] < 0) {
-			dev_err(dev, "unable to find a free IRQ\n");
+			snd_printk(KERN_ERR "%s: unable to find a free IRQ\n",
+				dev->bus_id);
 			return -EBUSY;
 		}
 	}
 	if (dma1[n] == SNDRV_AUTO_DMA) {
 		dma1[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma1[n] < 0) {
-			dev_err(dev, "unable to find a free DMA1\n");
+			snd_printk(KERN_ERR "%s: unable to find a free DMA1\n",
+				dev->bus_id);
 			return -EBUSY;
 		}
 	}
 	if (dma2[n] == SNDRV_AUTO_DMA) {
 		dma2[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma2[n] < 0) {
-			dev_err(dev, "unable to find a free DMA2\n");
+			snd_printk(KERN_ERR "%s: unable to find a free DMA2\n",
+				dev->bus_id);
 			return -EBUSY;
 		}
 	}
@@ -171,8 +174,8 @@ static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n)
 
 	error = -ENODEV;
 	if (gus->max_flag || gus->ess_flag) {
-		dev_err(dev, "GUS Classic or ACE soundcard was "
-			"not detected at 0x%lx\n", gus->gf1.port);
+		snd_printk(KERN_ERR "%s: GUS Classic or ACE soundcard was "
+			"not detected at 0x%lx\n", dev->bus_id, gus->gf1.port);
 		goto out;
 	}
 
diff --git a/trunk/sound/isa/gus/gusextreme.c b/trunk/sound/isa/gus/gusextreme.c
index 7ad4c3b41a84..da13185eb0a0 100644
--- a/trunk/sound/isa/gus/gusextreme.c
+++ b/trunk/sound/isa/gus/gusextreme.c
@@ -106,14 +106,16 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card,
 	if (irq[n] == SNDRV_AUTO_IRQ) {
 		irq[n] = snd_legacy_find_free_irq(possible_irqs);
 		if (irq[n] < 0) {
-			dev_err(dev, "unable to find a free IRQ for ES1688\n");
+			snd_printk(KERN_ERR "%s: unable to find a free IRQ "
+				"for ES1688\n", dev->bus_id);
 			return -EBUSY;
 		}
 	}
 	if (dma8[n] == SNDRV_AUTO_DMA) {
 		dma8[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma8[n] < 0) {
-			dev_err(dev, "unable to find a free DMA for ES1688\n");
+			snd_printk(KERN_ERR "%s: unable to find a free DMA "
+				"for ES1688\n", dev->bus_id);
 			return -EBUSY;
 		}
 	}
@@ -141,14 +143,16 @@ static int __devinit snd_gusextreme_gus_card_create(struct snd_card *card,
 	if (gf1_irq[n] == SNDRV_AUTO_IRQ) {
 		gf1_irq[n] = snd_legacy_find_free_irq(possible_irqs);
 		if (gf1_irq[n] < 0) {
-			dev_err(dev, "unable to find a free IRQ for GF1\n");
+			snd_printk(KERN_ERR "%s: unable to find a free IRQ "
+				"for GF1\n", dev->bus_id);
 			return -EBUSY;
 		}
 	}
 	if (dma1[n] == SNDRV_AUTO_DMA) {
 		dma1[n] = snd_legacy_find_free_dma(possible_dmas);
 		if (dma1[n] < 0) {
-			dev_err(dev, "unable to find a free DMA for GF1\n");
+			snd_printk(KERN_ERR "%s: unable to find a free DMA "
+				"for GF1\n", dev->bus_id);
 			return -EBUSY;
 		}
 	}
@@ -274,8 +278,8 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
 
 	error = -ENODEV;
 	if (!gus->ess_flag) {
-		dev_err(dev, "GUS Extreme soundcard was not "
-			"detected at 0x%lx\n", gus->gf1.port);
+		snd_printk(KERN_ERR "%s: GUS Extreme soundcard was not "
+			"detected at 0x%lx\n", dev->bus_id, gus->gf1.port);
 		goto out;
 	}
 	gus->codec_flag = 1;
@@ -306,7 +310,8 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
 
 	if (snd_opl3_create(card, es1688->port, es1688->port + 2,
 			OPL3_HW_OPL3, 0, &opl3) < 0)
-		dev_warn(dev, "opl3 not detected at 0x%lx\n", es1688->port);
+		printk(KERN_ERR "%s: opl3 not detected at 0x%lx\n",
+			dev->bus_id, es1688->port);
 	else {
 		error = snd_opl3_hwdep_new(opl3, 0, 2, NULL);
 		if (error < 0)
diff --git a/trunk/sound/isa/sb/sb8.c b/trunk/sound/isa/sb/sb8.c
index 667eccc676a4..336a34277907 100644
--- a/trunk/sound/isa/sb/sb8.c
+++ b/trunk/sound/isa/sb/sb8.c
@@ -85,11 +85,11 @@ static int __devinit snd_sb8_match(struct device *pdev, unsigned int dev)
 	if (!enable[dev])
 		return 0;
 	if (irq[dev] == SNDRV_AUTO_IRQ) {
-		dev_err(pdev, "please specify irq\n");
+		snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id);
 		return 0;
 	}
 	if (dma8[dev] == SNDRV_AUTO_DMA) {
-		dev_err(pdev, "please specify dma8\n");
+		snd_printk(KERN_ERR "%s: please specify dma8\n", pdev->bus_id);
 		return 0;
 	}
 	return 1;
diff --git a/trunk/sound/oss/ac97_codec.c b/trunk/sound/oss/ac97_codec.c
index 456a1b4d7832..b63839e8f9bd 100644
--- a/trunk/sound/oss/ac97_codec.c
+++ b/trunk/sound/oss/ac97_codec.c
@@ -30,7 +30,7 @@
  **************************************************************************
  *
  * History
- * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
+ * May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *	Removed non existant WM9700
  *	Added support for WM9705, WM9708, WM9709, WM9710, WM9711
  *	WM9712 and WM9717
diff --git a/trunk/sound/oss/kahlua.c b/trunk/sound/oss/kahlua.c
index c180598f1710..eb9bc365530d 100644
--- a/trunk/sound/oss/kahlua.c
+++ b/trunk/sound/oss/kahlua.c
@@ -1,7 +1,7 @@
 /*
  *	Initialisation code for Cyrix/NatSemi VSA1 softaudio
  *
- *	(C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
+ *	(C) Copyright 2003 Red Hat Inc <alan@redhat.com>
  *
  * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems.
  * The older version (VSA1) provides fairly good soundblaster emulation
diff --git a/trunk/sound/pci/ac97/ac97_codec.c b/trunk/sound/pci/ac97/ac97_codec.c
index bd510eceff1f..6704acbca8c0 100644
--- a/trunk/sound/pci/ac97/ac97_codec.c
+++ b/trunk/sound/pci/ac97/ac97_codec.c
@@ -1927,9 +1927,9 @@ static int snd_ac97_dev_register(struct snd_device *device)
 	ac97->dev.bus = &ac97_bus_type;
 	ac97->dev.parent = ac97->bus->card->dev;
 	ac97->dev.release = ac97_device_release;
-	dev_set_name(&ac97->dev, "%d-%d:%s",
-		     ac97->bus->card->number, ac97->num,
-		     snd_ac97_get_short_name(ac97));
+	snprintf(ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
+		 ac97->bus->card->number, ac97->num,
+		 snd_ac97_get_short_name(ac97));
 	if ((err = device_register(&ac97->dev)) < 0) {
 		snd_printk(KERN_ERR "Can't register ac97 bus\n");
 		ac97->dev.bus = NULL;
diff --git a/trunk/sound/pci/ac97/ac97_patch.c b/trunk/sound/pci/ac97/ac97_patch.c
index 6e831aff1bd0..6ce3cbe98a6a 100644
--- a/trunk/sound/pci/ac97/ac97_patch.c
+++ b/trunk/sound/pci/ac97/ac97_patch.c
@@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97)
 }
 
 /*
- * May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
+ * May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *  removed broken wolfson00 patch.
  *  added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717.
  */
diff --git a/trunk/sound/pci/ca0106/ca0106_main.c b/trunk/sound/pci/ca0106/ca0106_main.c
index 88fbf285d2b7..a7d89662acf6 100644
--- a/trunk/sound/pci/ca0106/ca0106_main.c
+++ b/trunk/sound/pci/ca0106/ca0106_main.c
@@ -759,6 +759,7 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream)
 			       SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
 			       SPCS_GENERATIONSTATUS | 0x00001200 |
 			       0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT );
+	}
 #endif
 
 	return 0;
diff --git a/trunk/sound/pci/cs5530.c b/trunk/sound/pci/cs5530.c
index 4ba2db2f4250..7ff8b68e997e 100644
--- a/trunk/sound/pci/cs5530.c
+++ b/trunk/sound/pci/cs5530.c
@@ -2,7 +2,7 @@
  * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio
  *
  * 	(C) Copyright 2007 Ash Willis <ashwillis@programmer.net>
- *	(C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
+ *	(C) Copyright 2003 Red Hat Inc <alan@redhat.com>
  *
  * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did
  * mess with it a bit. The chip seems to have to have trouble with full duplex
diff --git a/trunk/sound/pci/emu10k1/emu10k1_main.c b/trunk/sound/pci/emu10k1/emu10k1_main.c
index de5ee8f097f6..2f283ea6ad9a 100644
--- a/trunk/sound/pci/emu10k1/emu10k1_main.c
+++ b/trunk/sound/pci/emu10k1/emu10k1_main.c
@@ -1464,7 +1464,6 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
-	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102,
 	 .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", 
@@ -1474,7 +1473,6 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
-	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102,
 	 .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", 
@@ -1484,7 +1482,6 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
-	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	/* Audigy 2 */
 	/* Tested by James@superbug.co.uk 3rd July 2005 */
diff --git a/trunk/sound/pci/hda/hda_codec.c b/trunk/sound/pci/hda/hda_codec.c
index 6447754ae56e..ba1ab737b55f 100644
--- a/trunk/sound/pci/hda/hda_codec.c
+++ b/trunk/sound/pci/hda/hda_codec.c
@@ -64,6 +64,7 @@ static struct hda_vendor_id hda_vendor_ids[] = {
 	{ 0x14f1, "Conexant" },
 	{ 0x17e8, "Chrontel" },
 	{ 0x1854, "LG" },
+	{ 0x1aec, "Wolfson Microelectronics" },
 	{ 0x434d, "C-Media" },
 	{ 0x8384, "SigmaTel" },
 	{} /* terminator */
diff --git a/trunk/sound/pci/hda/hda_intel.c b/trunk/sound/pci/hda/hda_intel.c
index 9f316c1b2790..c04ac38e2b2e 100644
--- a/trunk/sound/pci/hda/hda_intel.c
+++ b/trunk/sound/pci/hda/hda_intel.c
@@ -45,6 +45,7 @@
 #include <linux/slab.h>
 #include <linux/pci.h>
 #include <linux/mutex.h>
+#include <linux/reboot.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include "hda_codec.h"
@@ -397,6 +398,9 @@ struct azx {
 
 	/* for pending irqs */
 	struct work_struct irq_pending_work;
+
+	/* reboot notifier (for mysterious hangup problem at power-down) */
+	struct notifier_block reboot_notifier;
 };
 
 /* driver types */
@@ -1978,6 +1982,28 @@ static int azx_resume(struct pci_dev *pci)
 #endif /* CONFIG_PM */
 
 
+/*
+ * reboot notifier for hang-up problem at power-down
+ */
+static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf)
+{
+	struct azx *chip = container_of(nb, struct azx, reboot_notifier);
+	azx_stop_chip(chip);
+	return NOTIFY_OK;
+}
+
+static void azx_notifier_register(struct azx *chip)
+{
+	chip->reboot_notifier.notifier_call = azx_halt;
+	register_reboot_notifier(&chip->reboot_notifier);
+}
+
+static void azx_notifier_unregister(struct azx *chip)
+{
+	if (chip->reboot_notifier.notifier_call)
+		unregister_reboot_notifier(&chip->reboot_notifier);
+}
+
 /*
  * destructor
  */
@@ -1985,6 +2011,8 @@ static int azx_free(struct azx *chip)
 {
 	int i;
 
+	azx_notifier_unregister(chip);
+
 	if (chip->initialized) {
 		azx_clear_irq_pending(chip);
 		for (i = 0; i < chip->num_streams; i++)
@@ -2348,6 +2376,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	pci_set_drvdata(pci, card);
 	chip->running = 1;
 	power_down_all_codecs(chip);
+	azx_notifier_register(chip);
 
 	dev++;
 	return err;
diff --git a/trunk/sound/pci/hda/hda_proc.c b/trunk/sound/pci/hda/hda_proc.c
index 743d77922bce..c39af986bff1 100644
--- a/trunk/sound/pci/hda/hda_proc.c
+++ b/trunk/sound/pci/hda/hda_proc.c
@@ -483,6 +483,8 @@ static void print_gpio(struct snd_info_buffer *buffer,
 		    (gpio & AC_GPIO_UNSOLICITED) ? 1 : 0,
 		    (gpio & AC_GPIO_WAKE) ? 1 : 0);
 	max = gpio & AC_GPIO_IO_COUNT;
+	if (!max || max > 8)
+		return;
 	enable = snd_hda_codec_read(codec, nid, 0,
 				    AC_VERB_GET_GPIO_MASK, 0);
 	direction = snd_hda_codec_read(codec, nid, 0,
diff --git a/trunk/sound/pci/hda/patch_analog.c b/trunk/sound/pci/hda/patch_analog.c
index 2b00c4afdf97..d3fd432cb3ea 100644
--- a/trunk/sound/pci/hda/patch_analog.c
+++ b/trunk/sound/pci/hda/patch_analog.c
@@ -3860,6 +3860,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
 
 static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
+	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
 	{}
 };
diff --git a/trunk/sound/pci/hda/patch_nvhdmi.c b/trunk/sound/pci/hda/patch_nvhdmi.c
index 1a65775d28e1..2eed2c8b98da 100644
--- a/trunk/sound/pci/hda/patch_nvhdmi.c
+++ b/trunk/sound/pci/hda/patch_nvhdmi.c
@@ -116,6 +116,7 @@ static int nvhdmi_build_pcms(struct hda_codec *codec)
 	codec->pcm_info = info;
 
 	info->name = "NVIDIA HDMI";
+	info->pcm_type = HDA_PCM_TYPE_HDMI;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback;
 
 	return 0;
diff --git a/trunk/sound/pci/hda/patch_realtek.c b/trunk/sound/pci/hda/patch_realtek.c
index 0b6e682c46d0..a4666c96a44f 100644
--- a/trunk/sound/pci/hda/patch_realtek.c
+++ b/trunk/sound/pci/hda/patch_realtek.c
@@ -307,6 +307,13 @@ struct alc_spec {
 	/* for PLL fix */
 	hda_nid_t pll_nid;
 	unsigned int pll_coef_idx, pll_coef_bit;
+
+#ifdef SND_HDA_NEEDS_RESUME
+#define ALC_MAX_PINS	16
+	unsigned int num_pins;
+	hda_nid_t pin_nids[ALC_MAX_PINS];
+	unsigned int pin_cfgs[ALC_MAX_PINS];
+#endif
 };
 
 /*
@@ -822,6 +829,31 @@ static void alc_sku_automute(struct hda_codec *codec)
 			    spec->jack_present ? 0 : PIN_OUT);
 }
 
+#if 0 /* it's broken in some acses -- temporarily disabled */
+static void alc_mic_automute(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int present;
+	unsigned int mic_nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
+	unsigned int fmic_nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
+	unsigned int mix_nid = spec->capsrc_nids[0];
+	unsigned int capsrc_idx_mic, capsrc_idx_fmic;
+
+	capsrc_idx_mic = mic_nid - 0x18;
+	capsrc_idx_fmic = fmic_nid - 0x18;
+	present = snd_hda_codec_read(codec, mic_nid, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+		    0x7000 | (capsrc_idx_mic << 8) | (present ? 0 : 0x80));
+	snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+		    0x7000 | (capsrc_idx_fmic << 8) | (present ? 0x80 : 0));
+	snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, capsrc_idx_fmic,
+			 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+#else
+#define alc_mic_automute(codec) /* NOP */
+#endif /* disabled */
+
 /* unsolicited event for HP jack sensing */
 static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
 {
@@ -829,10 +861,17 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
 		res >>= 28;
 	else
 		res >>= 26;
-	if (res != ALC880_HP_EVENT)
-		return;
+	if (res == ALC880_HP_EVENT)
+		alc_sku_automute(codec);
+
+	if (res == ALC880_MIC_EVENT)
+		alc_mic_automute(codec);
+}
 
+static void alc_inithook(struct hda_codec *codec)
+{
 	alc_sku_automute(codec);
+	alc_mic_automute(codec);
 }
 
 /* additional initialization for ALC888 variants */
@@ -1018,10 +1057,19 @@ static void alc_subsystem_id(struct hda_codec *codec,
 		else
 			return;
 	}
-
-	snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0,
-			    AC_VERB_SET_UNSOLICITED_ENABLE,
-			    AC_USRSP_EN | ALC880_HP_EVENT);
+	if (spec->autocfg.hp_pins[0])
+		snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0,
+			AC_VERB_SET_UNSOLICITED_ENABLE,
+			AC_USRSP_EN | ALC880_HP_EVENT);
+
+#if 0 /* it's broken in some acses -- temporarily disabled */
+	if (spec->autocfg.input_pins[AUTO_PIN_MIC] &&
+		spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC])
+		snd_hda_codec_write(codec,
+			spec->autocfg.input_pins[AUTO_PIN_MIC], 0,
+			AC_VERB_SET_UNSOLICITED_ENABLE,
+			AC_USRSP_EN | ALC880_MIC_EVENT);
+#endif /* disabled */
 
 	spec->unsol_event = alc_sku_unsol_event;
 }
@@ -2743,6 +2791,64 @@ static void alc_free(struct hda_codec *codec)
 	codec->spec = NULL; /* to be sure */
 }
 
+#ifdef SND_HDA_NEEDS_RESUME
+static void store_pin_configs(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid, end_nid;
+
+	end_nid = codec->start_nid + codec->num_nodes;
+	for (nid = codec->start_nid; nid < end_nid; nid++) {
+		unsigned int wid_caps = get_wcaps(codec, nid);
+		unsigned int wid_type =
+			(wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		if (wid_type != AC_WID_PIN)
+			continue;
+		if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids))
+			break;
+		spec->pin_nids[spec->num_pins] = nid;
+		spec->pin_cfgs[spec->num_pins] =
+			snd_hda_codec_read(codec, nid, 0,
+					   AC_VERB_GET_CONFIG_DEFAULT, 0);
+		spec->num_pins++;
+	}
+}
+
+static void resume_pin_configs(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < spec->num_pins; i++) {
+		hda_nid_t pin_nid = spec->pin_nids[i];
+		unsigned int pin_config = spec->pin_cfgs[i];
+		snd_hda_codec_write(codec, pin_nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
+				    pin_config & 0x000000ff);
+		snd_hda_codec_write(codec, pin_nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
+				    (pin_config & 0x0000ff00) >> 8);
+		snd_hda_codec_write(codec, pin_nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
+				    (pin_config & 0x00ff0000) >> 16);
+		snd_hda_codec_write(codec, pin_nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+				    pin_config >> 24);
+	}
+}
+
+static int alc_resume(struct hda_codec *codec)
+{
+	resume_pin_configs(codec);
+	codec->patch_ops.init(codec);
+	snd_hda_codec_resume_amp(codec);
+	snd_hda_codec_resume_cache(codec);
+	return 0;
+}
+#else
+#define store_pin_configs(codec)
+#endif
+
 /*
  */
 static struct hda_codec_ops alc_patch_ops = {
@@ -2751,6 +2857,9 @@ static struct hda_codec_ops alc_patch_ops = {
 	.init = alc_init,
 	.free = alc_free,
 	.unsol_event = alc_unsol_event,
+#ifdef SND_HDA_NEEDS_RESUME
+	.resume = alc_resume,
+#endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	.check_power_status = alc_check_power_status,
 #endif
@@ -3797,6 +3906,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -3808,7 +3918,7 @@ static void alc880_auto_init(struct hda_codec *codec)
 	alc880_auto_init_extra_out(codec);
 	alc880_auto_init_analog_input(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 /*
@@ -4961,7 +5071,7 @@ static struct hda_verb alc260_test_init_verbs[] = {
  */
 
 static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
-					const char *pfx)
+					const char *pfx, int *vol_bits)
 {
 	hda_nid_t nid_vol;
 	unsigned long vol_val, sw_val;
@@ -4983,10 +5093,14 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
 	} else
 		return 0; /* N/A */
 
-	snprintf(name, sizeof(name), "%s Playback Volume", pfx);
-	err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
-	if (err < 0)
-		return err;
+	if (!(*vol_bits & (1 << nid_vol))) {
+		/* first control for the volume widget */
+		snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+		err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
+		if (err < 0)
+			return err;
+		*vol_bits |= (1 << nid_vol);
+	}
 	snprintf(name, sizeof(name), "%s Playback Switch", pfx);
 	err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val);
 	if (err < 0)
@@ -5000,6 +5114,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 {
 	hda_nid_t nid;
 	int err;
+	int vols = 0;
 
 	spec->multiout.num_dacs = 1;
 	spec->multiout.dac_nids = spec->private_dac_nids;
@@ -5007,21 +5122,22 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 
 	nid = cfg->line_out_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Front");
+		err = alc260_add_playback_controls(spec, nid, "Front", &vols);
 		if (err < 0)
 			return err;
 	}
 
 	nid = cfg->speaker_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Speaker");
+		err = alc260_add_playback_controls(spec, nid, "Speaker", &vols);
 		if (err < 0)
 			return err;
 	}
 
 	nid = cfg->hp_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Headphone");
+		err = alc260_add_playback_controls(spec, nid, "Headphone",
+						   &vols);
 		if (err < 0)
 			return err;
 	}
@@ -5209,6 +5325,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 	}
 	spec->num_mixers++;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -5219,7 +5336,7 @@ static void alc260_auto_init(struct hda_codec *codec)
 	alc260_auto_init_multi_out(codec);
 	alc260_auto_init_analog_input(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -6629,7 +6746,7 @@ static void alc882_auto_init(struct hda_codec *codec)
 	alc882_auto_init_analog_input(codec);
 	alc882_auto_init_input_src(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */
@@ -8297,6 +8414,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
 static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
@@ -8306,8 +8424,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1043, 0x8317, "Asus M90V", ALC888_ASUS_M90V),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
 	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
@@ -8758,7 +8876,7 @@ static void alc883_auto_init(struct hda_codec *codec)
 	alc883_auto_init_analog_input(codec);
 	alc883_auto_init_input_src(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 static int patch_alc883(struct hda_codec *codec)
@@ -8802,8 +8920,13 @@ static int patch_alc883(struct hda_codec *codec)
 
 	switch (codec->vendor_id) {
 	case 0x10ec0888:
-		spec->stream_name_analog = "ALC888 Analog";
-		spec->stream_name_digital = "ALC888 Digital";
+		if (codec->revision_id == 0x100101) {
+			spec->stream_name_analog = "ALC1200 Analog";
+			spec->stream_name_digital = "ALC1200 Digital";
+		} else {
+			spec->stream_name_analog = "ALC888 Analog";
+			spec->stream_name_digital = "ALC888 Digital";
+		}
 		break;
 	case 0x10ec0889:
 		spec->stream_name_analog = "ALC889 Analog";
@@ -10267,6 +10390,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -10285,7 +10409,7 @@ static void alc262_auto_init(struct hda_codec *codec)
 	alc262_auto_init_analog_input(codec);
 	alc262_auto_init_input_src(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 /*
@@ -10343,7 +10467,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
 		      ALC262_TOSHIBA_RX1),
-	SND_PCI_QUIRK(0x1179, 0x0268, "Toshiba S06", ALC262_TOSHIBA_S06),
+	SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
 	SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
@@ -11401,6 +11525,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -11417,7 +11542,7 @@ static void alc268_auto_init(struct hda_codec *codec)
 	alc268_auto_init_mono_speaker_out(codec);
 	alc268_auto_init_analog_input(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 /*
@@ -12120,8 +12245,26 @@ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
 	return 0;
 }
 
-#define alc269_auto_create_analog_input_ctls \
-	alc880_auto_create_analog_input_ctls
+static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
+						const struct auto_pin_cfg *cfg)
+{
+	int err;
+
+	err = alc880_auto_create_analog_input_ctls(spec, cfg);
+	if (err < 0)
+		return err;
+	/* digital-mic input pin is excluded in alc880_auto_create..()
+	 * because it's under 0x18
+	 */
+	if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
+	    cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
+		struct hda_input_mux *imux = &spec->private_imux;
+		imux->items[imux->num_items].label = "Int Mic";
+		imux->items[imux->num_items].index = 0x05;
+		imux->num_items++;
+	}
+	return 0;
+}
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 #define alc269_loopbacks	alc880_loopbacks
@@ -12184,6 +12327,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	spec->mixers[spec->num_mixers] = alc269_capture_mixer;
 	spec->num_mixers++;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -12200,7 +12344,7 @@ static void alc269_auto_init(struct hda_codec *codec)
 	alc269_auto_init_hp_out(codec);
 	alc269_auto_init_analog_input(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 /*
@@ -13270,6 +13414,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	spec->mixers[spec->num_mixers] = alc861_capture_mixer;
 	spec->num_mixers++;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -13281,7 +13426,7 @@ static void alc861_auto_init(struct hda_codec *codec)
 	alc861_auto_init_hp_out(codec);
 	alc861_auto_init_analog_input(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -14381,6 +14526,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -14393,7 +14539,7 @@ static void alc861vd_auto_init(struct hda_codec *codec)
 	alc861vd_auto_init_analog_input(codec);
 	alc861vd_auto_init_input_src(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 static int patch_alc861vd(struct hda_codec *codec)
@@ -15667,7 +15813,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
 
 static struct snd_pci_quirk alc662_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
-	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS M51VA", ALC663_ASUS_G50V),
+	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
 	SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
 	SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
@@ -15680,6 +15826,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
@@ -16211,6 +16358,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 
 	spec->mixers[spec->num_mixers] = alc662_capture_mixer;
 	spec->num_mixers++;
+
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -16223,7 +16372,7 @@ static void alc662_auto_init(struct hda_codec *codec)
 	alc662_auto_init_analog_input(codec);
 	alc662_auto_init_input_src(codec);
 	if (spec->unsol_event)
-		alc_sku_automute(codec);
+		alc_inithook(codec);
 }
 
 static int patch_alc662(struct hda_codec *codec)
@@ -16268,6 +16417,9 @@ static int patch_alc662(struct hda_codec *codec)
 	if (codec->vendor_id == 0x10ec0663) {
 		spec->stream_name_analog = "ALC663 Analog";
 		spec->stream_name_digital = "ALC663 Digital";
+	} else if (codec->vendor_id == 0x10ec0272) {
+		spec->stream_name_analog = "ALC272 Analog";
+		spec->stream_name_digital = "ALC272 Digital";
 	} else {
 		spec->stream_name_analog = "ALC662 Analog";
 		spec->stream_name_digital = "ALC662 Digital";
@@ -16305,6 +16457,7 @@ 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 = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
 	  .patch = patch_alc861 },
 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
@@ -16323,7 +16476,10 @@ struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
 	  .patch = patch_alc882 }, /* should be patch_alc883() in future */
 	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
+	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 },
 	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
+	{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
+	  .patch = patch_alc883 },
 	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 },
 	{} /* terminator */
 };
diff --git a/trunk/sound/pci/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c
index c59065513118..e6085915d86d 100644
--- a/trunk/sound/pci/hda/patch_sigmatel.c
+++ b/trunk/sound/pci/hda/patch_sigmatel.c
@@ -69,6 +69,7 @@ enum {
 enum {
 	STAC_92HD73XX_REF,
 	STAC_DELL_M6,
+	STAC_DELL_EQ,
 	STAC_92HD73XX_MODELS
 };
 
@@ -566,10 +567,8 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
 			nid = codec->slave_dig_outs[smux_idx - 1];
 		if (spec->cur_smux[smux_idx] == smux->num_items - 1)
 			val = AMP_OUT_MUTE;
-		if (smux_idx == 0)
-			nid = spec->multiout.dig_out_nid;
 		else
-			nid = codec->slave_dig_outs[smux_idx - 1];
+			val = AMP_OUT_UNMUTE;
 		/* un/mute SPDIF out */
 		snd_hda_codec_write_cache(codec, nid, 0,
 			AC_VERB_SET_AMP_GAIN_MUTE, val);
@@ -775,9 +774,7 @@ static struct hda_verb dell_eq_core_init[] = {
 };
 
 static struct hda_verb dell_m6_core_init[] = {
-	/* set master volume to max value without distortion
-	 * and direct control */
-	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
+	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* setup audio connections */
 	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
@@ -1282,7 +1279,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 			return err;
 		spec->multiout.share_spdif = 1;
 	}
-	if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) {
+	if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
 		if (err < 0)
 			return err;
@@ -1602,11 +1599,13 @@ static unsigned int dell_m6_pin_configs[13] = {
 static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
 	[STAC_92HD73XX_REF]	= ref92hd73xx_pin_configs,
 	[STAC_DELL_M6]	= dell_m6_pin_configs,
+	[STAC_DELL_EQ]	= dell_m6_pin_configs,
 };
 
 static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
 	[STAC_92HD73XX_REF] = "ref",
 	[STAC_DELL_M6] = "dell-m6",
+	[STAC_DELL_EQ] = "dell-eq",
 };
 
 static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
@@ -2816,7 +2815,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 	static const char *chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
-	hda_nid_t nid;
+	hda_nid_t nid = 0;
 	int i, err;
 
 	struct sigmatel_spec *spec = codec->spec;
@@ -4133,12 +4132,17 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 			sizeof(stac92hd73xx_dmux));
 
 	switch (spec->board_config) {
-	case STAC_DELL_M6:
+	case STAC_DELL_EQ:
 		spec->init = dell_eq_core_init;
+		/* fallthru */
+	case STAC_DELL_M6:
 		spec->num_smuxes = 0;
 		spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER];
 		spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
 		spec->num_amps = 1;
+
+		if (!spec->init)
+			spec->init = dell_m6_core_init;
 		switch (codec->subsystem_id) {
 		case 0x1028025e: /* Analog Mics */
 		case 0x1028025f:
@@ -4148,8 +4152,6 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 			break;
 		case 0x10280271: /* Digital Mics */
 		case 0x10280272:
-			spec->init = dell_m6_core_init;
-			/* fall-through */
 		case 0x10280254:
 		case 0x10280255:
 			stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
diff --git a/trunk/sound/pci/ice1712/ice1712.c b/trunk/sound/pci/ice1712/ice1712.c
index 58d7cda03de5..5b442383fcda 100644
--- a/trunk/sound/pci/ice1712/ice1712.c
+++ b/trunk/sound/pci/ice1712/ice1712.c
@@ -2688,13 +2688,12 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 		return err;
 	}
 
-	if (ice_has_con_ac97(ice)) {
+	if (ice_has_con_ac97(ice))
 		err = snd_ice1712_pcm(ice, pcm_dev++, NULL);
 		if (err < 0) {
 			snd_card_free(card);
 			return err;
 		}
-	}
 
 	err = snd_ice1712_ac97_mixer(ice);
 	if (err < 0) {
@@ -2716,13 +2715,12 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 		}
 	}
 
-	if (ice_has_con_ac97(ice)) {
+	if (ice_has_con_ac97(ice))
 		err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL);
 		if (err < 0) {
 			snd_card_free(card);
 			return err;
 		}
-	}
 
 	if (!c->no_mpu401) {
 		err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
diff --git a/trunk/sound/pci/intel8x0.c b/trunk/sound/pci/intel8x0.c
index 19d3391e229f..c88d1eace1c4 100644
--- a/trunk/sound/pci/intel8x0.c
+++ b/trunk/sound/pci/intel8x0.c
@@ -2702,7 +2702,6 @@ static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000),
 	SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100),
 	SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000),
-	SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),
 	SND_PCI_QUIRK(0x1043, 0x80f3, "AD1985", 48000),
 	{ }	/* terminator */
 };
diff --git a/trunk/sound/pci/rme9652/hdsp.c b/trunk/sound/pci/rme9652/hdsp.c
index 736246f98acc..d723543beadd 100644
--- a/trunk/sound/pci/rme9652/hdsp.c
+++ b/trunk/sound/pci/rme9652/hdsp.c
@@ -4548,20 +4548,11 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 {
 	struct hdsp *hdsp = (struct hdsp *)hw->private_data;	
 	void __user *argp = (void __user *)arg;
-	int err;
 
 	switch (cmd) {
 	case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: {
 		struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg;
 
-		err = hdsp_check_for_iobox(hdsp);
-		if (err < 0)
-			return err;
-
-		err = hdsp_check_for_firmware(hdsp, 1);
-		if (err < 0)
-			return err;
-
 		if (!(hdsp->state & HDSP_FirmwareLoaded)) {
 			snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n");
 			return -EINVAL;
@@ -4581,14 +4572,10 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 		unsigned long flags;
 		int i;
 		
-		err = hdsp_check_for_iobox(hdsp);
-		if (err < 0)
-			return err;
-
-		err = hdsp_check_for_firmware(hdsp, 1);
-		if (err < 0)
-			return err;
-
+		if (!(hdsp->state & HDSP_FirmwareLoaded)) {
+			snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n");	
+			return -EINVAL;
+		}
 		spin_lock_irqsave(&hdsp->lock, flags);
 		info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp);
 		info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp);
@@ -5058,10 +5045,6 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 		/* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */
 		ssleep(2);
 
-		err = hdsp_check_for_iobox(hdsp);
-		if (err < 0)
-			return err;
-
 		if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
 #ifdef HDSP_FW_LOADER
 			if ((err = hdsp_request_fw_loader(hdsp)) < 0)
@@ -5074,7 +5057,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 				/* init is complete, we return */
 				return 0;
 #endif
-			/* we defer initialization */
+			/* no iobox connected, we defer initialization */
 			snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n");
 			if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0)
 				return err;
diff --git a/trunk/sound/soc/at91/Kconfig b/trunk/sound/soc/at91/Kconfig
index 85a883299c2e..905186502e00 100644
--- a/trunk/sound/soc/at91/Kconfig
+++ b/trunk/sound/soc/at91/Kconfig
@@ -8,3 +8,20 @@ config SND_AT91_SOC
 
 config SND_AT91_SOC_SSC
 	tristate
+
+config SND_AT91_SOC_ETI_B1_WM8731
+	tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards"
+	depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
+	select SND_AT91_SOC_SSC
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on WM8731-based
+	  Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.
+
+config SND_AT91_SOC_ETI_SLAVE
+	bool "Run codec in slave Mode on Endrelia boards"
+	depends on SND_AT91_SOC_ETI_B1_WM8731
+	default n
+	help
+	  Say Y if you want to run with the AT91 SSC generating the BCLK
+	  and LRC signals on Endrelia boards.
diff --git a/trunk/sound/soc/at91/Makefile b/trunk/sound/soc/at91/Makefile
index b817f11df286..f23da17cc328 100644
--- a/trunk/sound/soc/at91/Makefile
+++ b/trunk/sound/soc/at91/Makefile
@@ -4,3 +4,8 @@ snd-soc-at91-ssc-objs := at91-ssc.o
 
 obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
 obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
+
+# AT91 Machine Support
+snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
+
+obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o
diff --git a/trunk/sound/soc/at91/at91-ssc.c b/trunk/sound/soc/at91/at91-ssc.c
index 1b61cc461261..a5b1a79ebffb 100644
--- a/trunk/sound/soc/at91/at91-ssc.c
+++ b/trunk/sound/soc/at91/at91-ssc.c
@@ -5,7 +5,7 @@
  *         Endrelia Technologies Inc.
  *
  * Based on pxa2xx Platform drivers by
- * Liam Girdwood <lrg@slimlogic.co.uk>
+ * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/trunk/sound/soc/at91/eti_b1_wm8731.c b/trunk/sound/soc/at91/eti_b1_wm8731.c
new file mode 100644
index 000000000000..684781e4088b
--- /dev/null
+++ b/trunk/sound/soc/at91/eti_b1_wm8731.c
@@ -0,0 +1,349 @@
+/*
+ * eti_b1_wm8731  --  SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
+ *
+ * Author:	Frank Mandarino <fmandarino@endrelia.com>
+ *		Endrelia Technologies Inc.
+ * Created:	Mar 29, 2006
+ *
+ * Based on corgi.c by:
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+
+#include "../codecs/wm8731.h"
+#include "at91-pcm.h"
+#include "at91-ssc.h"
+
+#if 0
+#define	DBG(x...)	printk(KERN_INFO "eti_b1_wm8731: " x)
+#else
+#define	DBG(x...)
+#endif
+
+static struct clk *pck1_clk;
+static struct clk *pllb_clk;
+
+
+static int eti_b1_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* cpu clock is the AT91 master clock sent to the SSC */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
+		60000000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* codec system clock is supplied by PCK1, set to 12MHz */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+		12000000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* Start PCK1 clock. */
+	clk_enable(pck1_clk);
+	DBG("pck1 started\n");
+
+	return 0;
+}
+
+static void eti_b1_shutdown(struct snd_pcm_substream *substream)
+{
+	/* Stop PCK1 clock. */
+	clk_disable(pck1_clk);
+	DBG("pck1 stopped\n");
+}
+
+static int eti_b1_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
+	unsigned int rate;
+	int cmr_div, period;
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/*
+	 * The SSC clock dividers depend on the sample rate.  The CMR.DIV
+	 * field divides the system master clock MCK to drive the SSC TK
+	 * signal which provides the codec BCLK.  The TCMR.PERIOD and
+	 * RCMR.PERIOD fields further divide the BCLK signal to drive
+	 * the SSC TF and RF signals which provide the codec DACLRC and
+	 * ADCLRC clocks.
+	 *
+	 * The dividers were determined through trial and error, where a
+	 * CMR.DIV value is chosen such that the resulting BCLK value is
+	 * divisible, or almost divisible, by (2 * sample rate), and then
+	 * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
+	 */
+	rate = params_rate(params);
+
+	switch (rate) {
+	case 8000:
+		cmr_div = 25;	/* BCLK = 60MHz/(2*25) = 1.2MHz */
+		period = 74;	/* LRC = BCLK/(2*(74+1)) = 8000Hz */
+		break;
+	case 32000:
+		cmr_div = 7;	/* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
+		period = 66;	/* LRC = BCLK/(2*(66+1)) = 31982.942Hz */
+		break;
+	case 48000:
+		cmr_div = 13;	/* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */
+		period = 23;	/* LRC = BCLK/(2*(23+1)) = 48076.923Hz */
+		break;
+	default:
+		printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate);
+		return -EINVAL;
+	}
+
+	/* set the MCK divider for BCLK */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
+	if (ret < 0)
+		return ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* set the BCLK divider for DACLRC */
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
+						AT91SSC_TCMR_PERIOD, period);
+	} else {
+		/* set the BCLK divider for ADCLRC */
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
+						AT91SSC_RCMR_PERIOD, period);
+	}
+	if (ret < 0)
+		return ret;
+
+#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
+	/*
+	 * Codec in Master Mode.
+	 */
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
+
+	return 0;
+}
+
+static struct snd_soc_ops eti_b1_ops = {
+	.startup = eti_b1_startup,
+	.hw_params = eti_b1_hw_params,
+	.shutdown = eti_b1_shutdown,
+};
+
+
+static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+
+	/* speaker connected to LHPOUT */
+	{"Ext Spk", NULL, "LHPOUT"},
+
+	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
+	{"MICIN", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Int Mic"},
+};
+
+/*
+ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
+ */
+static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
+{
+	DBG("eti_b1_wm8731_init() called\n");
+
+	/* Add specific widgets */
+	snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets,
+				  ARRAY_SIZE(eti_b1_dapm_widgets));
+
+	/* Set up specific audio path interconnects */
+	snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon));
+
+	/* not connected */
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(codec, "Int Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link eti_b1_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731 PCM",
+	.cpu_dai = &at91_ssc_dai[1],
+	.codec_dai = &wm8731_dai,
+	.init = eti_b1_wm8731_init,
+	.ops = &eti_b1_ops,
+};
+
+static struct snd_soc_machine snd_soc_machine_eti_b1 = {
+	.name = "ETI_B1_WM8731",
+	.dai_link = &eti_b1_dai,
+	.num_links = 1,
+};
+
+static struct wm8731_setup_data eti_b1_wm8731_setup = {
+	.i2c_bus = 0,
+	.i2c_address = 0x1a,
+};
+
+static struct snd_soc_device eti_b1_snd_devdata = {
+	.machine = &snd_soc_machine_eti_b1,
+	.platform = &at91_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &eti_b1_wm8731_setup,
+};
+
+static struct platform_device *eti_b1_snd_device;
+
+static int __init eti_b1_init(void)
+{
+	int ret;
+	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
+
+	if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
+		DBG("SSC1 memory region is busy\n");
+		return -EBUSY;
+	}
+
+	ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K);
+	if (!ssc->base) {
+		DBG("SSC1 memory ioremap failed\n");
+		ret = -ENOMEM;
+		goto fail_release_mem;
+	}
+
+	ssc->pid = AT91RM9200_ID_SSC1;
+
+	eti_b1_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!eti_b1_snd_device) {
+		DBG("platform device allocation failed\n");
+		ret = -ENOMEM;
+		goto fail_io_unmap;
+	}
+
+	platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata);
+	eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev;
+
+	ret = platform_device_add(eti_b1_snd_device);
+	if (ret) {
+		DBG("platform device add failed\n");
+		platform_device_put(eti_b1_snd_device);
+		goto fail_io_unmap;
+	}
+
+	at91_set_A_periph(AT91_PIN_PB6, 0);	/* TF1 */
+	at91_set_A_periph(AT91_PIN_PB7, 0);	/* TK1 */
+	at91_set_A_periph(AT91_PIN_PB8, 0);	/* TD1 */
+	at91_set_A_periph(AT91_PIN_PB9, 0);	/* RD1 */
+/*	at91_set_A_periph(AT91_PIN_PB10, 0);*/	/* RK1 */
+	at91_set_A_periph(AT91_PIN_PB11, 0);	/* RF1 */
+
+	/*
+	 * Set PCK1 parent to PLLB and its rate to 12 Mhz.
+	 */
+	pllb_clk = clk_get(NULL, "pllb");
+	pck1_clk = clk_get(NULL, "pck1");
+
+	clk_set_parent(pck1_clk, pllb_clk);
+	clk_set_rate(pck1_clk, 12000000);
+
+	DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk));
+
+	/* assign the GPIO pin to PCK1 */
+	at91_set_B_periph(AT91_PIN_PA24, 0);
+
+#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
+	printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n");
+#else
+	printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n");
+#endif
+	return ret;
+
+fail_io_unmap:
+	iounmap(ssc->base);
+fail_release_mem:
+	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
+	return ret;
+}
+
+static void __exit eti_b1_exit(void)
+{
+	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
+
+	clk_put(pck1_clk);
+	clk_put(pllb_clk);
+
+	platform_device_unregister(eti_b1_snd_device);
+
+	iounmap(ssc->base);
+	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
+}
+
+module_init(eti_b1_init);
+module_exit(eti_b1_exit);
+
+/* Module information */
+MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
+MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731");
+MODULE_LICENSE("GPL");
diff --git a/trunk/sound/soc/blackfin/Kconfig b/trunk/sound/soc/blackfin/Kconfig
index dc006206f622..f98331d099e7 100644
--- a/trunk/sound/soc/blackfin/Kconfig
+++ b/trunk/sound/soc/blackfin/Kconfig
@@ -17,22 +17,6 @@ config SND_BF5XX_SOC_SSM2602
 	help
 	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
 
-config SND_BF5XX_SOC_AD73311
-	tristate "SoC AD73311 Audio support for Blackfin"
-	depends on SND_BF5XX_I2S
-	select SND_BF5XX_SOC_I2S
-	select SND_SOC_AD73311
-	help
-	  Say Y if you want to add support for AD73311 codec on Blackfin.
-
-config SND_BFIN_AD73311_SE
-	int "PF pin for AD73311L Chip Select"
-	depends on SND_BF5XX_SOC_AD73311
-	default 4
-	help
-	  Enter the GPIO used to control AD73311's SE pin. Acceptable
-	  values are 0 to 7
-
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
 	depends on BLACKFIN && SND_SOC
diff --git a/trunk/sound/soc/blackfin/Makefile b/trunk/sound/soc/blackfin/Makefile
index 97bb37a6359c..9ea8bd9e0ba3 100644
--- a/trunk/sound/soc/blackfin/Makefile
+++ b/trunk/sound/soc/blackfin/Makefile
@@ -14,8 +14,7 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
 # Blackfin Machine Support
 snd-ad1980-objs := bf5xx-ad1980.o
 snd-ssm2602-objs := bf5xx-ssm2602.o
-snd-ad73311-objs := bf5xx-ad73311.o
+
 
 obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
 obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
-obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
diff --git a/trunk/sound/soc/blackfin/bf5xx-ac97-pcm.c b/trunk/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 25e50d2ea1ec..51f4907c4831 100644
--- a/trunk/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/trunk/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -56,7 +56,6 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
 		sport->tx_pos += runtime->period_size;
 		if (sport->tx_pos >= runtime->buffer_size)
 			sport->tx_pos %= runtime->buffer_size;
-		sport->tx_delay_pos = sport->tx_pos;
 	} else {
 		bf5xx_ac97_to_pcm(
 			(struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos,
@@ -73,15 +72,7 @@ static void bf5xx_dma_irq(void *data)
 	struct snd_pcm_substream *pcm = data;
 #if defined(CONFIG_SND_MMAP_SUPPORT)
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	struct sport_device *sport = runtime->private_data;
 	bf5xx_mmap_copy(pcm, runtime->period_size);
-	if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (sport->once == 0) {
-			snd_pcm_period_elapsed(pcm);
-			bf5xx_mmap_copy(pcm, runtime->period_size);
-			sport->once = 1;
-		}
-	}
 #endif
 	snd_pcm_period_elapsed(pcm);
 }
@@ -123,10 +114,6 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
 
 static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-	memset(runtime->dma_area, 0, runtime->buffer_size);
 	snd_pcm_lib_free_pages(substream);
 	return 0;
 }
@@ -140,11 +127,16 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
 	 * SPORT working in TMD mode(include AC97).
 	 */
 #if defined(CONFIG_SND_MMAP_SUPPORT)
+	size_t size = bf5xx_pcm_hardware.buffer_bytes_max
+			* sizeof(struct ac97_frame) / 4;
+	/*clean up intermediate buffer*/
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		memset(sport->tx_dma_buf, 0, size);
 		sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
 		sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods,
 			runtime->period_size * sizeof(struct ac97_frame));
 	} else {
+		memset(sport->rx_dma_buf, 0, size);
 		sport_set_rx_callback(sport, bf5xx_dma_irq, substream);
 		sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods,
 			runtime->period_size * sizeof(struct ac97_frame));
@@ -172,12 +164,8 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	pr_debug("%s enter\n", __func__);
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			bf5xx_mmap_copy(substream, runtime->period_size);
-			snd_pcm_period_elapsed(substream);
-			sport->tx_delay_pos = 0;
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			sport_tx_start(sport);
-		}
 		else
 			sport_rx_start(sport);
 		break;
@@ -210,7 +198,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
 
 #if defined(CONFIG_SND_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		curr = sport->tx_delay_pos;
+		curr = sport->tx_pos;
 	else
 		curr = sport->rx_pos;
 #else
@@ -249,21 +237,6 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
 	return ret;
 }
 
-static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct sport_device *sport = runtime->private_data;
-
-	pr_debug("%s enter\n", __func__);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		sport->once = 0;
-		memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
-	} else
-		memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
-
-	return 0;
-}
-
 #ifdef CONFIG_SND_MMAP_SUPPORT
 static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma)
@@ -299,7 +272,6 @@ static	int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 
 struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
 	.open		= bf5xx_pcm_open,
-	.close		= bf5xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
 	.hw_free	= bf5xx_pcm_hw_free,
diff --git a/trunk/sound/soc/blackfin/bf5xx-ac97.c b/trunk/sound/soc/blackfin/bf5xx-ac97.c
index 5e5aafb6485f..c782e311fd56 100644
--- a/trunk/sound/soc/blackfin/bf5xx-ac97.c
+++ b/trunk/sound/soc/blackfin/bf5xx-ac97.c
@@ -128,6 +128,7 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data)
 	int nextfrag = sport_tx_curr_frag(sport);
 	struct ac97_frame *nextwrite;
 
+	sport_incfrag(sport, &nextfrag, 1);
 	sport_incfrag(sport, &nextfrag, 1);
 
 	nextwrite = (struct ac97_frame *)(sport->tx_buf + \
diff --git a/trunk/sound/soc/blackfin/bf5xx-ad73311.c b/trunk/sound/soc/blackfin/bf5xx-ad73311.c
deleted file mode 100644
index 622c9b909532..000000000000
--- a/trunk/sound/soc/blackfin/bf5xx-ad73311.c
+++ /dev/null
@@ -1,240 +0,0 @@
-/*
- * File:         sound/soc/blackfin/bf5xx-ad73311.c
- * Author:       Cliff Cai <Cliff.Cai@analog.com>
- *
- * Created:      Thur Sep 25 2008
- * Description:  Board driver for ad73311 sound chip
- *
- * Modified:
- *               Copyright 2008 Analog Devices Inc.
- *
- * Bugs:         Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, see the file COPYING, or write
- * to the Free Software Foundation, Inc.,
- * 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <linux/delay.h>
-#include <linux/gpio.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <sound/pcm_params.h>
-
-#include <asm/blackfin.h>
-#include <asm/cacheflush.h>
-#include <asm/irq.h>
-#include <asm/dma.h>
-#include <asm/portmux.h>
-
-#include "../codecs/ad73311.h"
-#include "bf5xx-sport.h"
-#include "bf5xx-i2s-pcm.h"
-#include "bf5xx-i2s.h"
-
-#if CONFIG_SND_BF5XX_SPORT_NUM == 0
-#define bfin_write_SPORT_TCR1	bfin_write_SPORT0_TCR1
-#define bfin_read_SPORT_TCR1	bfin_read_SPORT0_TCR1
-#define bfin_write_SPORT_TCR2	bfin_write_SPORT0_TCR2
-#define bfin_write_SPORT_TX16	bfin_write_SPORT0_TX16
-#define bfin_read_SPORT_STAT	bfin_read_SPORT0_STAT
-#else
-#define bfin_write_SPORT_TCR1	bfin_write_SPORT1_TCR1
-#define bfin_read_SPORT_TCR1	bfin_read_SPORT1_TCR1
-#define bfin_write_SPORT_TCR2	bfin_write_SPORT1_TCR2
-#define bfin_write_SPORT_TX16	bfin_write_SPORT1_TX16
-#define bfin_read_SPORT_STAT	bfin_read_SPORT1_STAT
-#endif
-
-#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
-
-static struct snd_soc_machine bf5xx_ad73311;
-
-static int snd_ad73311_startup(void)
-{
-	pr_debug("%s enter\n", __func__);
-
-	/* Pull up SE pin on AD73311L */
-	gpio_set_value(GPIO_SE, 1);
-	return 0;
-}
-
-static int snd_ad73311_configure(void)
-{
-	unsigned short ctrl_regs[6];
-	unsigned short status = 0;
-	int count = 0;
-
-	/* DMCLK = MCLK = 16.384 MHz
-	 * SCLK = DMCLK/8 = 2.048 MHz
-	 * Sample Rate = DMCLK/2048  = 8 KHz
-	 */
-	ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \
-			REGB_SCDIV(0) | REGB_DIRATE(0);
-	ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \
-			REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ;
-	ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \
-			REGD_IGS(2);
-	ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f);
-	ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ;
-	ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA;
-
-	local_irq_disable();
-	snd_ad73311_startup();
-	udelay(1);
-
-	bfin_write_SPORT_TCR1(TFSR);
-	bfin_write_SPORT_TCR2(0xF);
-	SSYNC();
-
-	/* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to
-	 * FIFO before enable SPORT to transfer the data
-	 */
-	for (count = 0; count < 6; count++)
-		bfin_write_SPORT_TX16(ctrl_regs[count]);
-	SSYNC();
-	bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN);
-	SSYNC();
-
-	/* When TUVF is set, the data is already send out */
-	while (!(status & TUVF) && count++ < 10000) {
-		udelay(1);
-		status = bfin_read_SPORT_STAT();
-		SSYNC();
-	}
-	bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN);
-	SSYNC();
-	local_irq_enable();
-
-	if (count == 10000) {
-		printk(KERN_ERR "ad73311: failed to configure codec\n");
-		return -1;
-	}
-	return 0;
-}
-
-static int bf5xx_probe(struct platform_device *pdev)
-{
-	int err;
-	if (gpio_request(GPIO_SE, "AD73311_SE")) {
-		printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE);
-		return -EBUSY;
-	}
-
-	gpio_direction_output(GPIO_SE, 0);
-
-	err = snd_ad73311_configure();
-	if (err < 0)
-		return -EFAULT;
-
-	return 0;
-}
-
-static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-
-	pr_debug("%s enter\n", __func__);
-	cpu_dai->private_data = sport_handle;
-	return 0;
-}
-
-static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	int ret = 0;
-
-	pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
-		params_format(params));
-
-	/* set cpu DAI configuration */
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-
-static struct snd_soc_ops bf5xx_ad73311_ops = {
-	.startup = bf5xx_ad73311_startup,
-	.hw_params = bf5xx_ad73311_hw_params,
-};
-
-static struct snd_soc_dai_link bf5xx_ad73311_dai = {
-	.name = "ad73311",
-	.stream_name = "AD73311",
-	.cpu_dai = &bf5xx_i2s_dai,
-	.codec_dai = &ad73311_dai,
-	.ops = &bf5xx_ad73311_ops,
-};
-
-static struct snd_soc_machine bf5xx_ad73311 = {
-	.name = "bf5xx_ad73311",
-	.probe = bf5xx_probe,
-	.dai_link = &bf5xx_ad73311_dai,
-	.num_links = 1,
-};
-
-static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
-	.machine = &bf5xx_ad73311,
-	.platform = &bf5xx_i2s_soc_platform,
-	.codec_dev = &soc_codec_dev_ad73311,
-};
-
-static struct platform_device *bf52x_ad73311_snd_device;
-
-static int __init bf5xx_ad73311_init(void)
-{
-	int ret;
-
-	pr_debug("%s enter\n", __func__);
-	bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!bf52x_ad73311_snd_device)
-		return -ENOMEM;
-
-	platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata);
-	bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev;
-	ret = platform_device_add(bf52x_ad73311_snd_device);
-
-	if (ret)
-		platform_device_put(bf52x_ad73311_snd_device);
-
-	return ret;
-}
-
-static void __exit bf5xx_ad73311_exit(void)
-{
-	pr_debug("%s enter\n", __func__);
-	platform_device_unregister(bf52x_ad73311_snd_device);
-}
-
-module_init(bf5xx_ad73311_init);
-module_exit(bf5xx_ad73311_exit);
-
-/* Module information */
-MODULE_AUTHOR("Cliff Cai");
-MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin");
-MODULE_LICENSE("GPL");
-
diff --git a/trunk/sound/soc/blackfin/bf5xx-i2s.c b/trunk/sound/soc/blackfin/bf5xx-i2s.c
index 827587f08180..43a4092eeb89 100644
--- a/trunk/sound/soc/blackfin/bf5xx-i2s.c
+++ b/trunk/sound/soc/blackfin/bf5xx-i2s.c
@@ -70,13 +70,6 @@ static struct sport_param sport_params[2] = {
 	}
 };
 
-static u16 sport_req[][7] = {
-		{ P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
-		  P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0},
-		{ P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS,
-		  P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0},
-};
-
 static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		unsigned int fmt)
 {
@@ -85,14 +78,6 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	/* interface format:support I2S,slave mode */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		bf5xx_i2s.tcr1 |= TFSR | TCKFE;
-		bf5xx_i2s.rcr1 |= RFSR | RCKFE;
-		bf5xx_i2s.tcr2 |= TSFSE;
-		bf5xx_i2s.rcr2 |= RSFSE;
-		break;
-	case SND_SOC_DAIFMT_DSP_A:
-		bf5xx_i2s.tcr1 |= TFSR;
-		bf5xx_i2s.rcr1 |= RFSR;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
 		ret = -EINVAL;
@@ -142,17 +127,14 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bf5xx_i2s.tcr2 |= 15;
 		bf5xx_i2s.rcr2 |= 15;
-		sport_handle->wdsize = 2;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		bf5xx_i2s.tcr2 |= 23;
 		bf5xx_i2s.rcr2 |= 23;
-		sport_handle->wdsize = 3;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
 		bf5xx_i2s.tcr2 |= 31;
 		bf5xx_i2s.rcr2 |= 31;
-		sport_handle->wdsize = 4;
 		break;
 	}
 
@@ -163,17 +145,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
 		 * need to configure both of them at the time when the first
 		 * stream is opened.
 		 *
-		 * CPU DAI:slave mode.
+		 * CPU DAI format:I2S, slave mode.
 		 */
-		ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1,
-				      bf5xx_i2s.rcr2, 0, 0);
+		ret = sport_config_rx(sport_handle, RFSR | RCKFE,
+				      RSFSE|bf5xx_i2s.rcr2, 0, 0);
 		if (ret) {
 			pr_err("SPORT is busy!\n");
 			return -EBUSY;
 		}
 
-		ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1,
-				      bf5xx_i2s.tcr2, 0, 0);
+		ret = sport_config_tx(sport_handle, TFSR | TCKFE,
+				      TSFSE|bf5xx_i2s.tcr2, 0, 0);
 		if (ret) {
 			pr_err("SPORT is busy!\n");
 			return -EBUSY;
@@ -192,6 +174,13 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream)
 static int bf5xx_i2s_probe(struct platform_device *pdev,
 			   struct snd_soc_dai *dai)
 {
+	u16 sport_req[][7] = {
+		{ P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
+		  P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0},
+		{ P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS,
+		  P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0},
+	};
+
 	pr_debug("%s enter\n", __func__);
 	if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
 		pr_err("Requesting Peripherals failed\n");
@@ -209,13 +198,6 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void bf5xx_i2s_remove(struct platform_device *pdev,
-			   struct snd_soc_dai *dai)
-{
-	pr_debug("%s enter\n", __func__);
-	peripheral_free_list(&sport_req[sport_num][0]);
-}
-
 #ifdef CONFIG_PM
 static int bf5xx_i2s_suspend(struct platform_device *dev,
 			     struct snd_soc_dai *dai)
@@ -281,16 +263,15 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 	.id = 0,
 	.type = SND_SOC_DAI_I2S,
 	.probe = bf5xx_i2s_probe,
-	.remove = bf5xx_i2s_remove,
 	.suspend = bf5xx_i2s_suspend,
 	.resume = bf5xx_i2s_resume,
 	.playback = {
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = BF5XX_I2S_RATES,
 		.formats = BF5XX_I2S_FORMATS,},
 	.capture = {
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = BF5XX_I2S_RATES,
 		.formats = BF5XX_I2S_FORMATS,},
diff --git a/trunk/sound/soc/blackfin/bf5xx-sport.h b/trunk/sound/soc/blackfin/bf5xx-sport.h
index fcadcc081f7f..4c163454bbf8 100644
--- a/trunk/sound/soc/blackfin/bf5xx-sport.h
+++ b/trunk/sound/soc/blackfin/bf5xx-sport.h
@@ -123,8 +123,6 @@ struct sport_device {
 	int rx_pos;
 	unsigned int tx_buffer_size;
 	unsigned int rx_buffer_size;
-	int tx_delay_pos;
-	int once;
 #endif
 	void *private_data;
 };
diff --git a/trunk/sound/soc/codecs/Kconfig b/trunk/sound/soc/codecs/Kconfig
index 4975d8573e4f..e0b9869df0f1 100644
--- a/trunk/sound/soc/codecs/Kconfig
+++ b/trunk/sound/soc/codecs/Kconfig
@@ -3,11 +3,9 @@ config SND_SOC_ALL_CODECS
 	depends on I2C
 	select SPI
 	select SPI_MASTER
-	select SND_SOC_AD73311
 	select SND_SOC_AK4535
 	select SND_SOC_CS4270
 	select SND_SOC_SSM2602
-	select SND_SOC_TLV320AIC23
 	select SND_SOC_TLV320AIC26
 	select SND_SOC_TLV320AIC3X
 	select SND_SOC_UDA1380
@@ -36,9 +34,6 @@ config SND_SOC_AC97_CODEC
 config SND_SOC_AD1980
 	tristate
 
-config SND_SOC_AD73311
-	tristate
-
 config SND_SOC_AK4535
 	tristate
 
@@ -63,13 +58,9 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_SSM2602
 	tristate
 
-config SND_SOC_TLV320AIC23
-	tristate
-	depends on I2C
-
 config SND_SOC_TLV320AIC26
 	tristate "TI TLV320AIC26 Codec support"
-	depends on SPI
+	depends on SND_SOC && SPI
 
 config SND_SOC_TLV320AIC3X
 	tristate
diff --git a/trunk/sound/soc/codecs/Makefile b/trunk/sound/soc/codecs/Makefile
index 90f0a585fc70..f977978a3409 100644
--- a/trunk/sound/soc/codecs/Makefile
+++ b/trunk/sound/soc/codecs/Makefile
@@ -1,10 +1,8 @@
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1980-objs := ad1980.o
-snd-soc-ad73311-objs := ad73311.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-ssm2602-objs := ssm2602.o
-snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-uda1380-objs := uda1380.o
@@ -22,11 +20,9 @@ snd-soc-wm9713-objs := wm9713.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
-obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
-obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
diff --git a/trunk/sound/soc/codecs/ac97.c b/trunk/sound/soc/codecs/ac97.c
index bd1ebdc6c86c..61fd96ca7bc7 100644
--- a/trunk/sound/soc/codecs/ac97.c
+++ b/trunk/sound/soc/codecs/ac97.c
@@ -2,7 +2,8 @@
  * ac97.c  --  ALSA Soc AC97 codec support
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/trunk/sound/soc/codecs/ad1980.c b/trunk/sound/soc/codecs/ad1980.c
index 1397b8e06c0b..4e09c1f2c063 100644
--- a/trunk/sound/soc/codecs/ad1980.c
+++ b/trunk/sound/soc/codecs/ad1980.c
@@ -13,6 +13,7 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
+#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
diff --git a/trunk/sound/soc/codecs/ad73311.c b/trunk/sound/soc/codecs/ad73311.c
deleted file mode 100644
index 37af8607b00a..000000000000
--- a/trunk/sound/soc/codecs/ad73311.c
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * ad73311.c  --  ALSA Soc AD73311 codec support
- *
- * Copyright:	Analog Device Inc.
- * Author:	Cliff Cai <cliff.cai@analog.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- *  Revision history
- *    25th Sep 2008   Initial version.
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/version.h>
-#include <linux/kernel.h>
-#include <linux/device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/ac97_codec.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include "ad73311.h"
-
-struct snd_soc_dai ad73311_dai = {
-	.name = "AD73311",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
-};
-EXPORT_SYMBOL_GPL(ad73311_dai);
-
-static int ad73311_soc_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec;
-	int ret = 0;
-
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-	mutex_init(&codec->mutex);
-	codec->name = "AD73311";
-	codec->owner = THIS_MODULE;
-	codec->dai = &ad73311_dai;
-	codec->num_dai = 1;
-	socdev->codec = codec;
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-
-	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-	if (ret < 0) {
-		printk(KERN_ERR "ad73311: failed to create pcms\n");
-		goto pcm_err;
-	}
-
-	ret = snd_soc_register_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "ad73311: failed to register card\n");
-		goto register_err;
-	}
-
-	return ret;
-
-register_err:
-	snd_soc_free_pcms(socdev);
-pcm_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
-	return ret;
-}
-
-static int ad73311_soc_remove(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
-
-	if (codec == NULL)
-		return 0;
-	snd_soc_free_pcms(socdev);
-	kfree(codec);
-	return 0;
-}
-
-struct snd_soc_codec_device soc_codec_dev_ad73311 = {
-	.probe = 	ad73311_soc_probe,
-	.remove = 	ad73311_soc_remove,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
-
-MODULE_DESCRIPTION("ASoC ad73311 driver");
-MODULE_AUTHOR("Cliff Cai ");
-MODULE_LICENSE("GPL");
diff --git a/trunk/sound/soc/codecs/ad73311.h b/trunk/sound/soc/codecs/ad73311.h
deleted file mode 100644
index 507ce0c30edf..000000000000
--- a/trunk/sound/soc/codecs/ad73311.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- * File:         sound/soc/codec/ad73311.h
- * Based on:
- * Author:       Cliff Cai <cliff.cai@analog.com>
- *
- * Created:      Thur Sep 25, 2008
- * Description:  definitions for AD73311 registers
- *
- *
- * Modified:
- *               Copyright 2006 Analog Devices Inc.
- *
- * Bugs:         Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, see the file COPYING, or write
- * to the Free Software Foundation, Inc.,
- * 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
- */
-
-#ifndef __AD73311_H__
-#define __AD73311_H__
-
-#define AD_CONTROL	0x8000
-#define AD_DATA		0x0000
-#define AD_READ		0x4000
-#define AD_WRITE	0x0000
-
-/* Control register A */
-#define CTRL_REG_A	(0 << 8)
-
-#define REGA_MODE_PRO	0x00
-#define REGA_MODE_DATA	0x01
-#define REGA_MODE_MIXED	0x03
-#define REGA_DLB		0x04
-#define REGA_SLB		0x08
-#define REGA_DEVC(x)		((x & 0x7) << 4)
-#define REGA_RESET		0x80
-
-/* Control register B */
-#define CTRL_REG_B	(1 << 8)
-
-#define REGB_DIRATE(x)	(x & 0x3)
-#define REGB_SCDIV(x)	((x & 0x3) << 2)
-#define REGB_MCDIV(x)	((x & 0x7) << 4)
-#define REGB_CEE		(1 << 7)
-
-/* Control register C */
-#define CTRL_REG_C	(2 << 8)
-
-#define REGC_PUDEV		(1 << 0)
-#define REGC_PUADC		(1 << 3)
-#define REGC_PUDAC		(1 << 4)
-#define REGC_PUREF		(1 << 5)
-#define REGC_REFUSE		(1 << 6)
-
-/* Control register D */
-#define CTRL_REG_D	(3 << 8)
-
-#define REGD_IGS(x)		(x & 0x7)
-#define REGD_RMOD		(1 << 3)
-#define REGD_OGS(x)		((x & 0x7) << 4)
-#define REGD_MUTE		(x << 7)
-
-/* Control register E */
-#define CTRL_REG_E	(4 << 8)
-
-#define REGE_DA(x)		(x & 0x1f)
-#define REGE_IBYP		(1 << 5)
-
-/* Control register F */
-#define CTRL_REG_F	(5 << 8)
-
-#define REGF_SEEN		(1 << 5)
-#define REGF_INV		(1 << 6)
-#define REGF_ALB		(1 << 7)
-
-extern struct snd_soc_dai ad73311_dai;
-extern struct snd_soc_codec_device soc_codec_dev_ad73311;
-#endif
diff --git a/trunk/sound/soc/codecs/ak4535.c b/trunk/sound/soc/codecs/ak4535.c
index 2a89b5888e11..088cf9927720 100644
--- a/trunk/sound/soc/codecs/ak4535.c
+++ b/trunk/sound/soc/codecs/ak4535.c
@@ -28,6 +28,7 @@
 
 #include "ak4535.h"
 
+#define AUDIO_NAME "ak4535"
 #define AK4535_VERSION "0.3"
 
 struct snd_soc_codec_device soc_codec_dev_ak4535;
diff --git a/trunk/sound/soc/codecs/ssm2602.c b/trunk/sound/soc/codecs/ssm2602.c
index 44ef0dacd564..940ce1c3522e 100644
--- a/trunk/sound/soc/codecs/ssm2602.c
+++ b/trunk/sound/soc/codecs/ssm2602.c
@@ -42,6 +42,7 @@
 
 #include "ssm2602.h"
 
+#define AUDIO_NAME "ssm2602"
 #define SSM2602_VERSION "0.1"
 
 struct snd_soc_codec_device soc_codec_dev_ssm2602;
diff --git a/trunk/sound/soc/codecs/tlv320aic23.c b/trunk/sound/soc/codecs/tlv320aic23.c
deleted file mode 100644
index bac7815e00fb..000000000000
--- a/trunk/sound/soc/codecs/tlv320aic23.c
+++ /dev/null
@@ -1,714 +0,0 @@
-/*
- * ALSA SoC TLV320AIC23 codec driver
- *
- * Author:      Arun KS, <arunks@mistralsolutions.com>
- * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd.,
- *
- * Based on sound/soc/codecs/wm8731.c by Richard Purdie
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * Notes:
- *  The AIC23 is a driver for a low power stereo audio
- *  codec tlv320aic23
- *
- *  The machine layer should disable unsupported inputs/outputs by
- *  snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
-#include <linux/delay.h>
-#include <linux/pm.h>
-#include <linux/i2c.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <sound/tlv.h>
-#include <sound/initval.h>
-
-#include "tlv320aic23.h"
-
-#define AIC23_VERSION "0.1"
-
-struct tlv320aic23_srate_reg_info {
-	u32 sample_rate;
-	u8 control;		/* SR3, SR2, SR1, SR0 and BOSR */
-	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
-};
-
-/*
- * AIC23 register cache
- */
-static const u16 tlv320aic23_reg[] = {
-	0x0097, 0x0097, 0x00F9, 0x00F9,	/* 0 */
-	0x001A, 0x0004, 0x0007, 0x0001,	/* 4 */
-	0x0020, 0x0000, 0x0000, 0x0000,	/* 8 */
-	0x0000, 0x0000, 0x0000, 0x0000,	/* 12 */
-};
-
-/*
- * read tlv320aic23 register cache
- */
-static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
-						      *codec, unsigned int reg)
-{
-	u16 *cache = codec->reg_cache;
-	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
-		return -1;
-	return cache[reg];
-}
-
-/*
- * write tlv320aic23 register cache
- */
-static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
-					       u8 reg, u16 value)
-{
-	u16 *cache = codec->reg_cache;
-	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
-		return;
-	cache[reg] = value;
-}
-
-/*
- * write to the tlv320aic23 register space
- */
-static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
-			     unsigned int value)
-{
-
-	u8 data;
-
-	/* TLV320AIC23 has 7 bit address and 9 bits of data
-	 * so we need to switch one data bit into reg and rest
-	 * of data into val
-	 */
-
-	if ((reg < 0 || reg > 9) && (reg != 15)) {
-		printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
-		return -1;
-	}
-
-	data = (reg << 1) | (value >> 8 & 0x01);
-
-	tlv320aic23_write_reg_cache(codec, reg, value);
-
-	if (codec->hw_write(codec->control_data, data,
-			    (value & 0xff)) == 0)
-		return 0;
-
-	printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
-	       value, reg);
-
-	return -EIO;
-}
-
-static const char *rec_src_text[] = { "Line", "Mic" };
-static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
-
-static const struct soc_enum rec_src_enum =
-	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
-
-static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
-SOC_DAPM_ENUM("Input Select", rec_src_enum);
-
-static const struct soc_enum tlv320aic23_rec_src =
-	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
-static const struct soc_enum tlv320aic23_deemph =
-	SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
-
-static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
-static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
-static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
-
-static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	u16 val, reg;
-
-	val = (ucontrol->value.integer.value[0] & 0x07);
-
-	/* linear conversion to userspace
-	* 000	=	-6db
-	* 001	=	-9db
-	* 010	=	-12db
-	* 011	=	-18db (Min)
-	* 100	=	0db (Max)
-	*/
-	val = (val >= 4) ? 4  : (3 - val);
-
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0);
-	tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
-
-	return 0;
-}
-
-static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	u16 val;
-
-	val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0);
-	val = val >> 6;
-	val = (val >= 4) ? 4  : (3 -  val);
-	ucontrol->value.integer.value[0] = val;
-	return 0;
-
-}
-
-#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
-		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
-	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\
-	.put = snd_soc_tlv320aic23_put_volsw, \
-	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
-
-static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
-	SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
-			 TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
-	SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
-	SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
-		     TLV320AIC23_RINVOL, 7, 1, 0),
-	SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
-			 TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
-	SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
-	SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
-	SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG,
-				  6, 4, 0, sidetone_vol_tlv),
-	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
-};
-
-/* add non dapm controls */
-static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
-{
-
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&tlv320aic23_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-
-}
-
-/* PGA Mixer controls for Line and Mic switch */
-static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
-	SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
-	SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
-};
-
-static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
-	SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
-	SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
-	SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
-			 &tlv320aic23_rec_src_mux_controls),
-	SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
-			   &tlv320aic23_output_mixer_controls[0],
-			   ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
-	SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
-	SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
-
-	SND_SOC_DAPM_OUTPUT("LHPOUT"),
-	SND_SOC_DAPM_OUTPUT("RHPOUT"),
-	SND_SOC_DAPM_OUTPUT("LOUT"),
-	SND_SOC_DAPM_OUTPUT("ROUT"),
-
-	SND_SOC_DAPM_INPUT("LLINEIN"),
-	SND_SOC_DAPM_INPUT("RLINEIN"),
-
-	SND_SOC_DAPM_INPUT("MICIN"),
-};
-
-static const struct snd_soc_dapm_route intercon[] = {
-	/* Output Mixer */
-	{"Output Mixer", "Line Bypass Switch", "Line Input"},
-	{"Output Mixer", "Playback Switch", "DAC"},
-	{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
-
-	/* Outputs */
-	{"RHPOUT", NULL, "Output Mixer"},
-	{"LHPOUT", NULL, "Output Mixer"},
-	{"LOUT", NULL, "Output Mixer"},
-	{"ROUT", NULL, "Output Mixer"},
-
-	/* Inputs */
-	{"Line Input", "NULL", "LLINEIN"},
-	{"Line Input", "NULL", "RLINEIN"},
-
-	{"Mic Input", "NULL", "MICIN"},
-
-	/* input mux */
-	{"Capture Source", "Line", "Line Input"},
-	{"Capture Source", "Mic", "Mic Input"},
-	{"ADC", NULL, "Capture Source"},
-
-};
-
-/* tlv320aic23 related */
-static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
-	{4000, 0x06, 1},	/*  4000 */
-	{8000, 0x06, 0},	/*  8000 */
-	{16000, 0x0C, 1},	/* 16000 */
-	{22050, 0x11, 1},	/* 22050 */
-	{24000, 0x00, 1},	/* 24000 */
-	{32000, 0x0C, 0},	/* 32000 */
-	{44100, 0x11, 0},	/* 44100 */
-	{48000, 0x00, 0},	/* 48000 */
-	{88200, 0x1F, 0},	/* 88200 */
-	{96000, 0x0E, 0},	/* 96000 */
-};
-
-static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
-{
-	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
-				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
-
-	/* set up audio path interconnects */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
-
-	snd_soc_dapm_new_widgets(codec);
-	return 0;
-}
-
-static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	u16 iface_reg, data;
-	u8 count = 0;
-
-	iface_reg =
-	    tlv320aic23_read_reg_cache(codec,
-				       TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
-
-	/* Search for the right sample rate */
-	/* Verify what happens if the rate is not supported
-	 * now it goes to 96Khz */
-	while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
-	       (count < ARRAY_SIZE(srate_reg_info))) {
-		count++;
-	}
-
-	data =  (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
-		(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
-		TLV320AIC23_USB_CLK_ON;
-
-	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
-
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
-		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
-		iface_reg |= (0x01 << 2);
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-		iface_reg |= (0x02 << 2);
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-		iface_reg |= (0x03 << 2);
-		break;
-	}
-	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
-
-	return 0;
-}
-
-static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-
-	/* set active */
-	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
-
-	return 0;
-}
-
-static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-
-	/* deactivate */
-	if (!codec->active) {
-		udelay(50);
-		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
-	}
-}
-
-static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	u16 reg;
-
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
-	if (mute)
-		reg |= TLV320AIC23_DACM_MUTE;
-
-	else
-		reg &= ~TLV320AIC23_DACM_MUTE;
-
-	tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
-
-	return 0;
-}
-
-static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
-				   unsigned int fmt)
-{
-	struct snd_soc_codec *codec = codec_dai->codec;
-	u16 iface_reg;
-
-	iface_reg =
-	    tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
-
-	/* set master/slave audio interface */
-	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBM_CFM:
-		iface_reg |= TLV320AIC23_MS_MASTER;
-		break;
-	case SND_SOC_DAIFMT_CBS_CFS:
-		break;
-	default:
-		return -EINVAL;
-
-	}
-
-	/* interface format */
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_I2S:
-		iface_reg |= TLV320AIC23_FOR_I2S;
-		break;
-	case SND_SOC_DAIFMT_DSP_A:
-		iface_reg |= TLV320AIC23_FOR_DSP;
-		break;
-	case SND_SOC_DAIFMT_RIGHT_J:
-		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		iface_reg |= TLV320AIC23_FOR_LJUST;
-		break;
-	default:
-		return -EINVAL;
-
-	}
-
-	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
-
-	return 0;
-}
-
-static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
-				      int clk_id, unsigned int freq, int dir)
-{
-	struct snd_soc_codec *codec = codec_dai->codec;
-
-	switch (freq) {
-	case 12000000:
-		return 0;
-	}
-	return -EINVAL;
-}
-
-static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
-				      enum snd_soc_bias_level level)
-{
-	u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
-
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		/* vref/mid, osc on, dac unmute */
-		tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
-		break;
-	case SND_SOC_BIAS_PREPARE:
-		break;
-	case SND_SOC_BIAS_STANDBY:
-		/* everything off except vref/vmid, */
-		tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
-		break;
-	case SND_SOC_BIAS_OFF:
-		/* everything off, dac mute, inactive */
-		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
-		tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
-		break;
-	}
-	codec->bias_level = level;
-	return 0;
-}
-
-#define AIC23_RATES	SNDRV_PCM_RATE_8000_96000
-#define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
-			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
-
-struct snd_soc_dai tlv320aic23_dai = {
-	.name = "tlv320aic23",
-	.playback = {
-		     .stream_name = "Playback",
-		     .channels_min = 2,
-		     .channels_max = 2,
-		     .rates = AIC23_RATES,
-		     .formats = AIC23_FORMATS,},
-	.capture = {
-		    .stream_name = "Capture",
-		    .channels_min = 2,
-		    .channels_max = 2,
-		    .rates = AIC23_RATES,
-		    .formats = AIC23_FORMATS,},
-	.ops = {
-		.prepare = tlv320aic23_pcm_prepare,
-		.hw_params = tlv320aic23_hw_params,
-		.shutdown = tlv320aic23_shutdown,
-		},
-	.dai_ops = {
-		    .digital_mute = tlv320aic23_mute,
-		    .set_fmt = tlv320aic23_set_dai_fmt,
-		    .set_sysclk = tlv320aic23_set_dai_sysclk,
-		    }
-};
-EXPORT_SYMBOL_GPL(tlv320aic23_dai);
-
-static int tlv320aic23_suspend(struct platform_device *pdev,
-			       pm_message_t state)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
-
-	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
-	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int tlv320aic23_resume(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
-	int i;
-	u16 reg;
-
-	/* Sync reg_cache with the hardware */
-	for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
-		u16 val = tlv320aic23_read_reg_cache(codec, reg);
-		tlv320aic23_write(codec, reg, val);
-	}
-
-	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
-
-	return 0;
-}
-
-/*
- * initialise the AIC23 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int tlv320aic23_init(struct snd_soc_device *socdev)
-{
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret = 0;
-	u16 reg;
-
-	codec->name = "tlv320aic23";
-	codec->owner = THIS_MODULE;
-	codec->read = tlv320aic23_read_reg_cache;
-	codec->write = tlv320aic23_write;
-	codec->set_bias_level = tlv320aic23_set_bias_level;
-	codec->dai = &tlv320aic23_dai;
-	codec->num_dai = 1;
-	codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg);
-	codec->reg_cache =
-	    kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL);
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
-
-	/* Reset codec */
-	tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
-
-	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-	if (ret < 0) {
-		printk(KERN_ERR "tlv320aic23: failed to create pcms\n");
-		goto pcm_err;
-	}
-
-	/* power on device */
-	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
-
-	/* Unmute input */
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
-	tlv320aic23_write(codec, TLV320AIC23_LINVOL,
-			  (reg & (~TLV320AIC23_LIM_MUTED)) |
-			  (TLV320AIC23_LRS_ENABLED));
-
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
-	tlv320aic23_write(codec, TLV320AIC23_RINVOL,
-			  (reg & (~TLV320AIC23_LIM_MUTED)) |
-			  TLV320AIC23_LRS_ENABLED);
-
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
-	tlv320aic23_write(codec, TLV320AIC23_ANLG,
-			 (reg) & (~TLV320AIC23_BYPASS_ON) &
-			 (~TLV320AIC23_MICM_MUTED));
-
-	/* Default output volume */
-	tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
-			  TLV320AIC23_DEFAULT_OUT_VOL &
-			  TLV320AIC23_OUT_VOL_MASK);
-	tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
-			  TLV320AIC23_DEFAULT_OUT_VOL &
-			  TLV320AIC23_OUT_VOL_MASK);
-
-	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
-
-	tlv320aic23_add_controls(codec);
-	tlv320aic23_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "tlv320aic23: failed to register card\n");
-		goto card_err;
-	}
-
-	return ret;
-
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-pcm_err:
-	kfree(codec->reg_cache);
-	return ret;
-}
-static struct snd_soc_device *tlv320aic23_socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-/*
- * If the i2c layer weren't so broken, we could pass this kind of data
- * around
- */
-static int tlv320aic23_codec_probe(struct i2c_client *i2c,
-				   const struct i2c_device_id *i2c_id)
-{
-	struct snd_soc_device *socdev = tlv320aic23_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
-		return -EINVAL;
-
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
-
-	ret = tlv320aic23_init(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n");
-		goto err;
-	}
-	return ret;
-
-err:
-	kfree(codec);
-	kfree(i2c);
-	return ret;
-}
-static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
-{
-	put_device(&i2c->dev);
-	return 0;
-}
-
-static const struct i2c_device_id tlv320aic23_id[] = {
-	{"tlv320aic23", 0},
-	{}
-};
-
-MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
-
-static struct i2c_driver tlv320aic23_i2c_driver = {
-	.driver = {
-		   .name = "tlv320aic23",
-		   },
-	.probe = tlv320aic23_codec_probe,
-	.remove = __exit_p(tlv320aic23_i2c_remove),
-	.id_table = tlv320aic23_id,
-};
-
-#endif
-
-static int tlv320aic23_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec;
-	int ret = 0;
-
-	printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
-
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-
-	tlv320aic23_socdev = socdev;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data;
-	codec->hw_read = NULL;
-	ret = i2c_add_driver(&tlv320aic23_i2c_driver);
-	if (ret != 0)
-		printk(KERN_ERR "can't add i2c driver");
-#endif
-	return ret;
-}
-
-static int tlv320aic23_remove(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
-
-	if (codec->control_data)
-		tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_del_driver(&tlv320aic23_i2c_driver);
-#endif
-	kfree(codec->reg_cache);
-	kfree(codec);
-
-	return 0;
-}
-struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
-	.probe = tlv320aic23_probe,
-	.remove = tlv320aic23_remove,
-	.suspend = tlv320aic23_suspend,
-	.resume = tlv320aic23_resume,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
-
-MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
-MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
-MODULE_LICENSE("GPL");
diff --git a/trunk/sound/soc/codecs/tlv320aic23.h b/trunk/sound/soc/codecs/tlv320aic23.h
deleted file mode 100644
index 79d1faf8e570..000000000000
--- a/trunk/sound/soc/codecs/tlv320aic23.h
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * ALSA SoC TLV320AIC23 codec driver
- *
- * Author:      Arun KS, <arunks@mistralsolutions.com>
- * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _TLV320AIC23_H
-#define _TLV320AIC23_H
-
-/* Codec TLV320AIC23 */
-#define TLV320AIC23_LINVOL		0x00
-#define TLV320AIC23_RINVOL		0x01
-#define TLV320AIC23_LCHNVOL		0x02
-#define TLV320AIC23_RCHNVOL		0x03
-#define TLV320AIC23_ANLG		0x04
-#define TLV320AIC23_DIGT		0x05
-#define TLV320AIC23_PWR			0x06
-#define TLV320AIC23_DIGT_FMT		0x07
-#define TLV320AIC23_SRATE		0x08
-#define TLV320AIC23_ACTIVE		0x09
-#define TLV320AIC23_RESET		0x0F
-
-/* Left (right) line input volume control register */
-#define TLV320AIC23_LRS_ENABLED		0x0100
-#define TLV320AIC23_LIM_MUTED		0x0080
-#define TLV320AIC23_LIV_DEFAULT		0x0017
-#define TLV320AIC23_LIV_MAX		0x001f
-#define TLV320AIC23_LIV_MIN		0x0000
-
-/* Left (right) channel headphone volume control register */
-#define TLV320AIC23_LZC_ON		0x0080
-#define TLV320AIC23_LHV_DEFAULT		0x0079
-#define TLV320AIC23_LHV_MAX		0x007f
-#define TLV320AIC23_LHV_MIN		0x0000
-
-/* Analog audio path control register */
-#define TLV320AIC23_STA_REG(x)		((x)<<6)
-#define TLV320AIC23_STE_ENABLED		0x0020
-#define TLV320AIC23_DAC_SELECTED	0x0010
-#define TLV320AIC23_BYPASS_ON		0x0008
-#define TLV320AIC23_INSEL_MIC		0x0004
-#define TLV320AIC23_MICM_MUTED		0x0002
-#define TLV320AIC23_MICB_20DB		0x0001
-
-/* Digital audio path control register */
-#define TLV320AIC23_DACM_MUTE		0x0008
-#define TLV320AIC23_DEEMP_32K		0x0002
-#define TLV320AIC23_DEEMP_44K		0x0004
-#define TLV320AIC23_DEEMP_48K		0x0006
-#define TLV320AIC23_ADCHP_ON		0x0001
-
-/* Power control down register */
-#define TLV320AIC23_DEVICE_PWR_OFF  	0x0080
-#define TLV320AIC23_CLK_OFF		0x0040
-#define TLV320AIC23_OSC_OFF		0x0020
-#define TLV320AIC23_OUT_OFF		0x0010
-#define TLV320AIC23_DAC_OFF		0x0008
-#define TLV320AIC23_ADC_OFF		0x0004
-#define TLV320AIC23_MIC_OFF		0x0002
-#define TLV320AIC23_LINE_OFF		0x0001
-
-/* Digital audio interface register */
-#define TLV320AIC23_MS_MASTER		0x0040
-#define TLV320AIC23_LRSWAP_ON		0x0020
-#define TLV320AIC23_LRP_ON		0x0010
-#define TLV320AIC23_IWL_16		0x0000
-#define TLV320AIC23_IWL_20		0x0004
-#define TLV320AIC23_IWL_24		0x0008
-#define TLV320AIC23_IWL_32		0x000C
-#define TLV320AIC23_FOR_I2S		0x0002
-#define TLV320AIC23_FOR_DSP		0x0003
-#define TLV320AIC23_FOR_LJUST		0x0001
-
-/* Sample rate control register */
-#define TLV320AIC23_CLKOUT_HALF		0x0080
-#define TLV320AIC23_CLKIN_HALF		0x0040
-#define TLV320AIC23_BOSR_384fs		0x0002	/* BOSR_272fs in USB mode */
-#define TLV320AIC23_USB_CLK_ON		0x0001
-#define TLV320AIC23_SR_MASK             0xf
-#define TLV320AIC23_CLKOUT_SHIFT        7
-#define TLV320AIC23_CLKIN_SHIFT         6
-#define TLV320AIC23_SR_SHIFT            2
-#define TLV320AIC23_BOSR_SHIFT          1
-
-/* Digital interface register */
-#define TLV320AIC23_ACT_ON		0x0001
-
-/*
- * AUDIO related MACROS
- */
-
-#define TLV320AIC23_DEFAULT_OUT_VOL	0x70
-#define TLV320AIC23_DEFAULT_IN_VOLUME	0x10
-
-#define TLV320AIC23_OUT_VOL_MIN		TLV320AIC23_LHV_MIN
-#define TLV320AIC23_OUT_VOL_MAX		TLV320AIC23_LHV_MAX
-#define TLV320AIC23_OUT_VO_RANGE	(TLV320AIC23_OUT_VOL_MAX - \
-					TLV320AIC23_OUT_VOL_MIN)
-#define TLV320AIC23_OUT_VOL_MASK	TLV320AIC23_OUT_VOL_MAX
-
-#define TLV320AIC23_IN_VOL_MIN		TLV320AIC23_LIV_MIN
-#define TLV320AIC23_IN_VOL_MAX		TLV320AIC23_LIV_MAX
-#define TLV320AIC23_IN_VOL_RANGE	(TLV320AIC23_IN_VOL_MAX - \
-					TLV320AIC23_IN_VOL_MIN)
-#define TLV320AIC23_IN_VOL_MASK		TLV320AIC23_IN_VOL_MAX
-
-#define TLV320AIC23_SIDETONE_MASK	0x1c0
-#define TLV320AIC23_SIDETONE_0		0x100
-#define TLV320AIC23_SIDETONE_6		0x000
-#define TLV320AIC23_SIDETONE_9		0x040
-#define TLV320AIC23_SIDETONE_12		0x080
-#define TLV320AIC23_SIDETONE_18		0x0c0
-
-extern struct snd_soc_dai tlv320aic23_dai;
-extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23;
-
-#endif /* _TLV320AIC23_H */
diff --git a/trunk/sound/soc/codecs/tlv320aic3x.c b/trunk/sound/soc/codecs/tlv320aic3x.c
index 05336ed7e493..566a427c928f 100644
--- a/trunk/sound/soc/codecs/tlv320aic3x.c
+++ b/trunk/sound/soc/codecs/tlv320aic3x.c
@@ -48,6 +48,7 @@
 
 #include "tlv320aic3x.h"
 
+#define AUDIO_NAME "aic3x"
 #define AIC3X_VERSION "0.2"
 
 /* codec private data */
@@ -990,7 +991,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 struct snd_soc_dai aic3x_dai = {
-	.name = "tlv320aic3x",
+	.name = "aic3x",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
@@ -1054,7 +1055,7 @@ static int aic3x_init(struct snd_soc_device *socdev)
 	struct aic3x_setup_data *setup = socdev->codec_data;
 	int reg, ret = 0;
 
-	codec->name = "tlv320aic3x";
+	codec->name = "aic3x";
 	codec->owner = THIS_MODULE;
 	codec->read = aic3x_read_reg_cache;
 	codec->write = aic3x_write;
diff --git a/trunk/sound/soc/codecs/uda1380.c b/trunk/sound/soc/codecs/uda1380.c
index a69ee72a7af5..d206d7f892b6 100644
--- a/trunk/sound/soc/codecs/uda1380.c
+++ b/trunk/sound/soc/codecs/uda1380.c
@@ -36,6 +36,7 @@
 #include "uda1380.h"
 
 #define UDA1380_VERSION "0.6"
+#define AUDIO_NAME "uda1380"
 
 /*
  * uda1380 register cache
diff --git a/trunk/sound/soc/codecs/wm8510.c b/trunk/sound/soc/codecs/wm8510.c
index d8ca2da8d634..9a37c8d95ed2 100644
--- a/trunk/sound/soc/codecs/wm8510.c
+++ b/trunk/sound/soc/codecs/wm8510.c
@@ -3,7 +3,7 @@
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
  *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
@@ -18,7 +18,6 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
-#include <linux/spi/spi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -28,6 +27,7 @@
 
 #include "wm8510.h"
 
+#define AUDIO_NAME "wm8510"
 #define WM8510_VERSION "0.6"
 
 struct snd_soc_codec_device soc_codec_dev_wm8510;
@@ -55,9 +55,6 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
 	0x0001,
 };
 
-#define WM8510_POWER1_BIASEN  0x08
-#define WM8510_POWER1_BUFIOEN 0x10
-
 /*
  * read wm8510 register cache
  */
@@ -227,9 +224,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
 SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
 SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
 
-SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0,
-		   &wm8510_micpga_controls[0],
-		   ARRAY_SIZE(wm8510_micpga_controls)),
+SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0,
+		 &wm8510_micpga_controls[0],
+		 ARRAY_SIZE(wm8510_micpga_controls)),
 SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0,
 	&wm8510_boost_controls[0],
 	ARRAY_SIZE(wm8510_boost_controls)),
@@ -529,35 +526,23 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute)
 static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
-	u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-	case SND_SOC_BIAS_PREPARE:
-		power1 |= 0x1;  /* VMID 50k */
-		wm8510_write(codec, WM8510_POWER1, power1);
+		wm8510_write(codec, WM8510_POWER1, 0x1ff);
+		wm8510_write(codec, WM8510_POWER2, 0x1ff);
+		wm8510_write(codec, WM8510_POWER3, 0x1ff);
 		break;
-
+	case SND_SOC_BIAS_PREPARE:
 	case SND_SOC_BIAS_STANDBY:
-		power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
-
-		if (codec->bias_level == SND_SOC_BIAS_OFF) {
-			/* Initial cap charge at VMID 5k */
-			wm8510_write(codec, WM8510_POWER1, power1 | 0x3);
-			mdelay(100);
-		}
-
-		power1 |= 0x2;  /* VMID 500k */
-		wm8510_write(codec, WM8510_POWER1, power1);
 		break;
-
 	case SND_SOC_BIAS_OFF:
-		wm8510_write(codec, WM8510_POWER1, 0);
-		wm8510_write(codec, WM8510_POWER2, 0);
-		wm8510_write(codec, WM8510_POWER3, 0);
+		/* everything off, dac mute, inactive */
+		wm8510_write(codec, WM8510_POWER1, 0x0);
+		wm8510_write(codec, WM8510_POWER2, 0x0);
+		wm8510_write(codec, WM8510_POWER3, 0x0);
 		break;
 	}
-
 	codec->bias_level = level;
 	return 0;
 }
@@ -655,7 +640,6 @@ static int wm8510_init(struct snd_soc_device *socdev)
 	}
 
 	/* power on device */
-	codec->bias_level = SND_SOC_BIAS_OFF;
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm8510_add_controls(codec);
 	wm8510_add_widgets(codec);
@@ -763,62 +747,6 @@ static int wm8510_add_i2c_device(struct platform_device *pdev,
 }
 #endif
 
-#if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8510_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_device *socdev = wm8510_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	codec->control_data = spi;
-
-	ret = wm8510_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8510\n");
-
-	return ret;
-}
-
-static int __devexit wm8510_spi_remove(struct spi_device *spi)
-{
-	return 0;
-}
-
-static struct spi_driver wm8510_spi_driver = {
-	.driver = {
-		.name	= "wm8510",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8510_spi_probe,
-	.remove		= __devexit_p(wm8510_spi_remove),
-};
-
-static int wm8510_spi_write(struct spi_device *spi, const char *data, int len)
-{
-	struct spi_transfer t;
-	struct spi_message m;
-	u8 msg[2];
-
-	if (len <= 0)
-		return 0;
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-
-	spi_message_init(&m);
-	memset(&t, 0, (sizeof t));
-
-	t.tx_buf = &msg[0];
-	t.len = len;
-
-	spi_message_add_tail(&t, &m);
-	spi_sync(spi, &m);
-
-	return len;
-}
-#endif /* CONFIG_SPI_MASTER */
-
 static int wm8510_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -844,14 +772,8 @@ static int wm8510_probe(struct platform_device *pdev)
 		codec->hw_write = (hw_write_t)i2c_master_send;
 		ret = wm8510_add_i2c_device(pdev, setup);
 	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		codec->hw_write = (hw_write_t)wm8510_spi_write;
-		ret = spi_register_driver(&wm8510_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
+#else
+	/* Add other interfaces here */
 #endif
 
 	if (ret != 0)
@@ -873,9 +795,6 @@ static int wm8510_remove(struct platform_device *pdev)
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8510_i2c_driver);
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8510_spi_driver);
 #endif
 	kfree(codec);
 
diff --git a/trunk/sound/soc/codecs/wm8510.h b/trunk/sound/soc/codecs/wm8510.h
index bdefcf5c69ff..c53683960456 100644
--- a/trunk/sound/soc/codecs/wm8510.h
+++ b/trunk/sound/soc/codecs/wm8510.h
@@ -94,7 +94,6 @@
 #define WM8510_MCLKDIV_12	(7 << 5)
 
 struct wm8510_setup_data {
-	int spi;
 	int i2c_bus;
 	unsigned short i2c_address;
 };
diff --git a/trunk/sound/soc/codecs/wm8580.c b/trunk/sound/soc/codecs/wm8580.c
index 627ebfb4209b..df1ffbe305bf 100644
--- a/trunk/sound/soc/codecs/wm8580.c
+++ b/trunk/sound/soc/codecs/wm8580.c
@@ -18,6 +18,7 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
+#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
@@ -35,6 +36,7 @@
 
 #include "wm8580.h"
 
+#define AUDIO_NAME "wm8580"
 #define WM8580_VERSION "0.1"
 
 struct pll_state {
diff --git a/trunk/sound/soc/codecs/wm8731.c b/trunk/sound/soc/codecs/wm8731.c
index 7f8a7e36b33e..7b64d9a7ff76 100644
--- a/trunk/sound/soc/codecs/wm8731.c
+++ b/trunk/sound/soc/codecs/wm8731.c
@@ -29,6 +29,7 @@
 
 #include "wm8731.h"
 
+#define AUDIO_NAME "wm8731"
 #define WM8731_VERSION "0.13"
 
 struct snd_soc_codec_device soc_codec_dev_wm8731;
diff --git a/trunk/sound/soc/codecs/wm8750.c b/trunk/sound/soc/codecs/wm8750.c
index 9b7296ee5b08..4892e398a598 100644
--- a/trunk/sound/soc/codecs/wm8750.c
+++ b/trunk/sound/soc/codecs/wm8750.c
@@ -29,6 +29,7 @@
 
 #include "wm8750.h"
 
+#define AUDIO_NAME "WM8750"
 #define WM8750_VERSION "0.12"
 
 /* codec private data */
diff --git a/trunk/sound/soc/codecs/wm8753.c b/trunk/sound/soc/codecs/wm8753.c
index d426eaa22185..8c4df44f3345 100644
--- a/trunk/sound/soc/codecs/wm8753.c
+++ b/trunk/sound/soc/codecs/wm8753.c
@@ -2,7 +2,8 @@
  * wm8753.c  --  WM8753 ALSA Soc Audio driver
  *
  * Copyright 2003 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -39,7 +40,6 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
-#include <linux/spi/spi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -51,6 +51,7 @@
 
 #include "wm8753.h"
 
+#define AUDIO_NAME "wm8753"
 #define WM8753_VERSION "0.16"
 
 static int caps_charge = 2000;
@@ -1718,63 +1719,6 @@ static int wm8753_add_i2c_device(struct platform_device *pdev,
 }
 #endif
 
-#if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8753_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	codec->control_data = spi;
-
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8753\n");
-
-	return ret;
-}
-
-static int __devexit wm8753_spi_remove(struct spi_device *spi)
-{
-	return 0;
-}
-
-static struct spi_driver wm8753_spi_driver = {
-	.driver = {
-		.name	= "wm8753",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8753_spi_probe,
-	.remove		= __devexit_p(wm8753_spi_remove),
-};
-
-static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
-{
-	struct spi_transfer t;
-	struct spi_message m;
-	u8 msg[2];
-
-	if (len <= 0)
-		return 0;
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-
-	spi_message_init(&m);
-	memset(&t, 0, (sizeof t));
-
-	t.tx_buf = &msg[0];
-	t.len = len;
-
-	spi_message_add_tail(&t, &m);
-	spi_sync(spi, &m);
-
-	return len;
-}
-#endif
-
-
 static int wm8753_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -1809,14 +1753,8 @@ static int wm8753_probe(struct platform_device *pdev)
 		codec->hw_write = (hw_write_t)i2c_master_send;
 		ret = wm8753_add_i2c_device(pdev, setup);
 	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		codec->hw_write = (hw_write_t)wm8753_spi_write;
-		ret = spi_register_driver(&wm8753_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
+#else
+		/* Add other interfaces here */
 #endif
 
 	if (ret != 0) {
@@ -1859,9 +1797,6 @@ static int wm8753_remove(struct platform_device *pdev)
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8753_i2c_driver);
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8753_spi_driver);
 #endif
 	kfree(codec->private_data);
 	kfree(codec);
diff --git a/trunk/sound/soc/codecs/wm8753.h b/trunk/sound/soc/codecs/wm8753.h
index f55704ce931b..7defde069f1d 100644
--- a/trunk/sound/soc/codecs/wm8753.h
+++ b/trunk/sound/soc/codecs/wm8753.h
@@ -2,7 +2,8 @@
  * wm8753.h  --  audio driver for WM8753
  *
  * Copyright 2003 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -78,7 +79,6 @@
 #define WM8753_ADCTL2		0x3f
 
 struct wm8753_setup_data {
-	int spi;
 	int i2c_bus;
 	unsigned short i2c_address;
 };
diff --git a/trunk/sound/soc/codecs/wm8900.c b/trunk/sound/soc/codecs/wm8900.c
index 3b326c9b5586..0b8c6d38b48f 100644
--- a/trunk/sound/soc/codecs/wm8900.c
+++ b/trunk/sound/soc/codecs/wm8900.c
@@ -18,6 +18,7 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
+#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/trunk/sound/soc/codecs/wm8903.c b/trunk/sound/soc/codecs/wm8903.c
index ce40d7877605..a3f54ec4226e 100644
--- a/trunk/sound/soc/codecs/wm8903.c
+++ b/trunk/sound/soc/codecs/wm8903.c
@@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = {
 
 /* Input PGAs - No TLV since the scale depends on PGA mode */
 SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0,
-	   7, 1, 1),
+	   7, 1, 0),
 SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0,
 	   0, 31, 0),
 SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1,
 	   6, 1, 0),
 
 SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0,
-	   7, 1, 1),
+	   7, 1, 0),
 SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0,
 	   0, 31, 0),
 SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
diff --git a/trunk/sound/soc/codecs/wm8971.c b/trunk/sound/soc/codecs/wm8971.c
index f41a578ddd4f..974a4cd0f3fd 100644
--- a/trunk/sound/soc/codecs/wm8971.c
+++ b/trunk/sound/soc/codecs/wm8971.c
@@ -29,6 +29,7 @@
 
 #include "wm8971.h"
 
+#define AUDIO_NAME "wm8971"
 #define WM8971_VERSION "0.9"
 
 #define	WM8971_REG_COUNT		43
diff --git a/trunk/sound/soc/codecs/wm8990.c b/trunk/sound/soc/codecs/wm8990.c
index 572d22b0880b..63410d7b5efb 100644
--- a/trunk/sound/soc/codecs/wm8990.c
+++ b/trunk/sound/soc/codecs/wm8990.c
@@ -30,6 +30,7 @@
 
 #include "wm8990.h"
 
+#define AUDIO_NAME "wm8990"
 #define WM8990_VERSION "0.2"
 
 /* codec private data */
diff --git a/trunk/sound/soc/codecs/wm9712.c b/trunk/sound/soc/codecs/wm9712.c
index ffb471e420e2..2f1c91b1d556 100644
--- a/trunk/sound/soc/codecs/wm9712.c
+++ b/trunk/sound/soc/codecs/wm9712.c
@@ -2,7 +2,8 @@
  * wm9712.c  --  ALSA Soc WM9712 codec support
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/trunk/sound/soc/codecs/wm9713.c b/trunk/sound/soc/codecs/wm9713.c
index aba402b3c999..441d0580db1f 100644
--- a/trunk/sound/soc/codecs/wm9713.c
+++ b/trunk/sound/soc/codecs/wm9713.c
@@ -2,7 +2,8 @@
  * wm9713.c  --  ALSA Soc WM9713 codec support
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/trunk/sound/soc/omap/Kconfig b/trunk/sound/soc/omap/Kconfig
index 8b7766b998d7..aea27e70043c 100644
--- a/trunk/sound/soc/omap/Kconfig
+++ b/trunk/sound/soc/omap/Kconfig
@@ -13,11 +13,3 @@ config SND_OMAP_SOC_N810
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
-
-config SND_OMAP_SOC_OSK5912
-	tristate "SoC Audio support for omap osk5912"
-	depends on SND_OMAP_SOC && MACH_OMAP_OSK
-	select SND_OMAP_SOC_MCBSP
-	select SND_SOC_TLV320AIC23
-	help
-	  Say Y if you want to add support for SoC audio on osk5912.
diff --git a/trunk/sound/soc/omap/Makefile b/trunk/sound/soc/omap/Makefile
index e09d1f297f64..d8d8d58075e3 100644
--- a/trunk/sound/soc/omap/Makefile
+++ b/trunk/sound/soc/omap/Makefile
@@ -7,7 +7,5 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
-snd-soc-osk5912-objs := osk5912.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
-obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
diff --git a/trunk/sound/soc/omap/n810.c b/trunk/sound/soc/omap/n810.c
index fae3ad36e0bf..d166b6b2a60d 100644
--- a/trunk/sound/soc/omap/n810.c
+++ b/trunk/sound/soc/omap/n810.c
@@ -247,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
 	int i, err;
 
 	/* Not connected */
-	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
-	snd_soc_dapm_nc_pin(codec, "HPLCOM");
-	snd_soc_dapm_nc_pin(codec, "HPRCOM");
+	snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
+	snd_soc_dapm_disable_pin(codec, "HPLCOM");
+	snd_soc_dapm_disable_pin(codec, "HPRCOM");
 
 	/* Add N810 specific controls */
 	for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
diff --git a/trunk/sound/soc/omap/omap-mcbsp.c b/trunk/sound/soc/omap/omap-mcbsp.c
index 0a063a98a661..35310e16d7f3 100644
--- a/trunk/sound/soc/omap/omap-mcbsp.c
+++ b/trunk/sound/soc/omap/omap-mcbsp.c
@@ -59,7 +59,12 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
  * Stream DMA parameters. DMA request line and port address are set runtime
  * since they are different between OMAP1 and later OMAPs
  */
-static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2];
+static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = {
+{
+	{ .name		= "I2S PCM Stereo out", },
+	{ .name		= "I2S PCM Stereo in", },
+},
+};
 
 #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
 static const int omap1_dma_reqs[][2] = {
@@ -79,22 +84,11 @@ static const unsigned long omap1_mcbsp_port[][2] = {
 static const int omap1_dma_reqs[][2] = {};
 static const unsigned long omap1_mcbsp_port[][2] = {};
 #endif
-
-#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX)
-static const int omap24xx_dma_reqs[][2] = {
+#if defined(CONFIG_ARCH_OMAP2420)
+static const int omap2420_dma_reqs[][2] = {
 	{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
 	{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
-#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
-	{ OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
-	{ OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
-	{ OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
-#endif
 };
-#else
-static const int omap24xx_dma_reqs[][2] = {};
-#endif
-
-#if defined(CONFIG_ARCH_OMAP2420)
 static const unsigned long omap2420_mcbsp_port[][2] = {
 	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
 	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
@@ -102,43 +96,10 @@ static const unsigned long omap2420_mcbsp_port[][2] = {
 	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
 };
 #else
+static const int omap2420_dma_reqs[][2] = {};
 static const unsigned long omap2420_mcbsp_port[][2] = {};
 #endif
 
-#if defined(CONFIG_ARCH_OMAP2430)
-static const unsigned long omap2430_mcbsp_port[][2] = {
-	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
-};
-#else
-static const unsigned long omap2430_mcbsp_port[][2] = {};
-#endif
-
-#if defined(CONFIG_ARCH_OMAP34XX)
-static const unsigned long omap34xx_mcbsp_port[][2] = {
-	{ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
-	{ OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
-	  OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
-};
-#else
-static const unsigned long omap34xx_mcbsp_port[][2] = {};
-#endif
-
 static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -206,19 +167,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		dma = omap1_dma_reqs[bus_id][substream->stream];
 		port = omap1_mcbsp_port[bus_id][substream->stream];
 	} else if (cpu_is_omap2420()) {
-		dma = omap24xx_dma_reqs[bus_id][substream->stream];
+		dma = omap2420_dma_reqs[bus_id][substream->stream];
 		port = omap2420_mcbsp_port[bus_id][substream->stream];
-	} else if (cpu_is_omap2430()) {
-		dma = omap24xx_dma_reqs[bus_id][substream->stream];
-		port = omap2430_mcbsp_port[bus_id][substream->stream];
-	} else if (cpu_is_omap343x()) {
-		dma = omap24xx_dma_reqs[bus_id][substream->stream];
-		port = omap34xx_mcbsp_port[bus_id][substream->stream];
 	} else {
+		/*
+		 * TODO: Add support for 2430 and 3430
+		 */
 		return -ENODEV;
 	}
-	omap_mcbsp_dai_dma_params[id][substream->stream].name =
-		substream->stream ? "Audio Capture" : "Audio Playback";
 	omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
 	omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
 	cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
@@ -289,11 +245,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		regs->rcr2	|= RDATDLY(1);
 		regs->xcr2	|= XDATDLY(1);
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
-		/* 0-bit data delay */
-		regs->rcr2      |= RDATDLY(0);
-		regs->xcr2      |= XDATDLY(0);
-		break;
 	default:
 		/* Unsupported data format */
 		return -EINVAL;
@@ -359,7 +310,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 				       int clk_id)
 {
 	int sel_bit;
-	u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1;
+	u16 reg;
 
 	if (cpu_class_is_omap1()) {
 		/* OMAP1's can use only external source clock */
@@ -369,12 +320,6 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 			return 0;
 	}
 
-	if (cpu_is_omap2420() && mcbsp_data->bus_id > 1)
-		return -EINVAL;
-
-	if (cpu_is_omap343x())
-		reg_devconf1 = OMAP343X_CONTROL_DEVCONF1;
-
 	switch (mcbsp_data->bus_id) {
 	case 0:
 		reg = OMAP2_CONTROL_DEVCONF0;
@@ -384,26 +329,20 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 		reg = OMAP2_CONTROL_DEVCONF0;
 		sel_bit = 6;
 		break;
-	case 2:
-		reg = reg_devconf1;
-		sel_bit = 0;
-		break;
-	case 3:
-		reg = reg_devconf1;
-		sel_bit = 2;
-		break;
-	case 4:
-		reg = reg_devconf1;
-		sel_bit = 4;
-		break;
+	/* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */
 	default:
 		return -EINVAL;
 	}
 
-	if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK)
-		omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
-	else
-		omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
+	if (cpu_class_is_omap2()) {
+		if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) {
+			omap_ctrl_writel(omap_ctrl_readl(reg) &
+					 ~(1 << sel_bit), reg);
+		} else {
+			omap_ctrl_writel(omap_ctrl_readl(reg) |
+					 (1 << sel_bit), reg);
+		}
+	}
 
 	return 0;
 }
@@ -437,49 +376,37 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	return err;
 }
 
-#define OMAP_MCBSP_DAI_BUILDER(link_id)				\
-{								\
-	.name = "omap-mcbsp-dai-(link_id)",			\
-	.id = (link_id),					\
-	.type = SND_SOC_DAI_I2S,				\
-	.playback = {						\
-		.channels_min = 2,				\
-		.channels_max = 2,				\
-		.rates = OMAP_MCBSP_RATES,			\
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
-	},							\
-	.capture = {						\
-		.channels_min = 2,				\
-		.channels_max = 2,				\
-		.rates = OMAP_MCBSP_RATES,			\
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
-	},							\
-	.ops = {						\
-		.startup = omap_mcbsp_dai_startup,		\
-		.shutdown = omap_mcbsp_dai_shutdown,		\
-		.trigger = omap_mcbsp_dai_trigger,		\
-		.hw_params = omap_mcbsp_dai_hw_params,		\
-	},							\
-	.dai_ops = {						\
-		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
-		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
-		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
-	},							\
-	.private_data = &mcbsp_data[(link_id)].bus_id,		\
-}
-
-struct snd_soc_dai omap_mcbsp_dai[] = {
-	OMAP_MCBSP_DAI_BUILDER(0),
-	OMAP_MCBSP_DAI_BUILDER(1),
-#if NUM_LINKS >= 3
-	OMAP_MCBSP_DAI_BUILDER(2),
-#endif
-#if NUM_LINKS == 5
-	OMAP_MCBSP_DAI_BUILDER(3),
-	OMAP_MCBSP_DAI_BUILDER(4),
-#endif
+struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = {
+{
+	.name = "omap-mcbsp-dai",
+	.id = 0,
+	.type = SND_SOC_DAI_I2S,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = OMAP_MCBSP_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = OMAP_MCBSP_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = {
+		.startup = omap_mcbsp_dai_startup,
+		.shutdown = omap_mcbsp_dai_shutdown,
+		.trigger = omap_mcbsp_dai_trigger,
+		.hw_params = omap_mcbsp_dai_hw_params,
+	},
+	.dai_ops = {
+		.set_fmt = omap_mcbsp_dai_set_dai_fmt,
+		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,
+		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,
+	},
+	.private_data = &mcbsp_data[0].bus_id,
+},
 };
-
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
diff --git a/trunk/sound/soc/omap/omap-mcbsp.h b/trunk/sound/soc/omap/omap-mcbsp.h
index df7ad13ba73d..ed8afb550671 100644
--- a/trunk/sound/soc/omap/omap-mcbsp.h
+++ b/trunk/sound/soc/omap/omap-mcbsp.h
@@ -38,17 +38,11 @@ enum omap_mcbsp_div {
 	OMAP_MCBSP_CLKGDV,		/* Sample rate generator divider */
 };
 
-#if defined(CONFIG_ARCH_OMAP2420)
-#define NUM_LINKS	2
-#endif
-#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
-#undef  NUM_LINKS
-#define NUM_LINKS	3
-#endif
-#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
-#undef  NUM_LINKS
-#define NUM_LINKS	5
-#endif
+/*
+ * REVISIT: Preparation for the ASoC v2. Let the number of available links to
+ * be same than number of McBSP ports found in OMAP(s) we are compiling for.
+ */
+#define NUM_LINKS	1
 
 extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
 
diff --git a/trunk/sound/soc/omap/omap-pcm.c b/trunk/sound/soc/omap/omap-pcm.c
index e9084fdd2082..690bfeaec4a0 100644
--- a/trunk/sound/soc/omap/omap-pcm.c
+++ b/trunk/sound/soc/omap/omap-pcm.c
@@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	prtd->dma_data = dma_data;
 	err = omap_request_dma(dma_data->dma_req, dma_data->name,
 			       omap_pcm_dma_irq, substream, &prtd->dma_ch);
-	if (!err & !cpu_is_omap1510()) {
+	if (!cpu_is_omap1510()) {
 		/*
 		 * Link channel with itself so DMA doesn't need any
 		 * reprogramming while looping the buffer
@@ -147,14 +147,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 		dma_params.src_or_dst_synch	= OMAP_DMA_DST_SYNC;
 		dma_params.src_start		= runtime->dma_addr;
 		dma_params.dst_start		= dma_data->port_addr;
-		dma_params.dst_port		= OMAP_DMA_PORT_MPUI;
 	} else {
 		dma_params.src_amode		= OMAP_DMA_AMODE_CONSTANT;
 		dma_params.dst_amode		= OMAP_DMA_AMODE_POST_INC;
 		dma_params.src_or_dst_synch	= OMAP_DMA_SRC_SYNC;
 		dma_params.src_start		= dma_data->port_addr;
 		dma_params.dst_start		= runtime->dma_addr;
-		dma_params.src_port		= OMAP_DMA_PORT_MPUI;
 	}
 	/*
 	 * Set DMA transfer frame size equal to ALSA period size and frame
diff --git a/trunk/sound/soc/omap/osk5912.c b/trunk/sound/soc/omap/osk5912.c
deleted file mode 100644
index 0fe733796898..000000000000
--- a/trunk/sound/soc/omap/osk5912.c
+++ /dev/null
@@ -1,232 +0,0 @@
-/*
- * osk5912.c  --  SoC audio for OSK 5912
- *
- * Copyright (C) 2008 Mistral Solutions
- *
- * Contact: Arun KS  <arunks@mistralsolutions.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <linux/gpio.h>
-#include <mach/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-#include "../codecs/tlv320aic23.h"
-
-#define CODEC_CLOCK 	12000000
-
-static struct clk *tlv320aic23_mclk;
-
-static int osk_startup(struct snd_pcm_substream *substream)
-{
-	return clk_enable(tlv320aic23_mclk);
-}
-
-static void osk_shutdown(struct snd_pcm_substream *substream)
-{
-	clk_disable(tlv320aic23_mclk);
-}
-
-static int osk_hw_params(struct snd_pcm_substream *substream,
-			 struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	int err;
-
-	/* Set codec DAI configuration */
-	err = snd_soc_dai_set_fmt(codec_dai,
-				  SND_SOC_DAIFMT_DSP_A |
-				  SND_SOC_DAIFMT_NB_IF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-	if (err < 0) {
-		printk(KERN_ERR "can't set codec DAI configuration\n");
-		return err;
-	}
-
-	/* Set cpu DAI configuration */
-	err = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_DSP_A |
-				  SND_SOC_DAIFMT_NB_IF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-	if (err < 0) {
-		printk(KERN_ERR "can't set cpu DAI configuration\n");
-		return err;
-	}
-
-	/* Set the codec system clock for DAC and ADC */
-	err =
-	    snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
-
-	if (err < 0) {
-		printk(KERN_ERR "can't set codec system clock\n");
-		return err;
-	}
-
-	return err;
-}
-
-static struct snd_soc_ops osk_ops = {
-	.startup = osk_startup,
-	.hw_params = osk_hw_params,
-	.shutdown = osk_shutdown,
-};
-
-static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headphone Jack", NULL),
-	SND_SOC_DAPM_LINE("Line In", NULL),
-	SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
-	{"Headphone Jack", NULL, "LHPOUT"},
-	{"Headphone Jack", NULL, "RHPOUT"},
-
-	{"LLINEIN", NULL, "Line In"},
-	{"RLINEIN", NULL, "Line In"},
-
-	{"MICIN", NULL, "Mic Jack"},
-};
-
-static int osk_tlv320aic23_init(struct snd_soc_codec *codec)
-{
-
-	/* Add osk5912 specific widgets */
-	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
-				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
-
-	/* Set up osk5912 specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
-
-	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-	snd_soc_dapm_enable_pin(codec, "Line In");
-	snd_soc_dapm_enable_pin(codec, "Mic Jack");
-
-	snd_soc_dapm_sync(codec);
-
-	return 0;
-}
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link osk_dai = {
-	.name = "TLV320AIC23",
-	.stream_name = "AIC23",
-	.cpu_dai = &omap_mcbsp_dai[0],
-	.codec_dai = &tlv320aic23_dai,
-	.init = osk_tlv320aic23_init,
-	.ops = &osk_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_osk = {
-	.name = "OSK5912",
-	.dai_link = &osk_dai,
-	.num_links = 1,
-};
-
-/* Audio subsystem */
-static struct snd_soc_device osk_snd_devdata = {
-	.machine = &snd_soc_machine_osk,
-	.platform = &omap_soc_platform,
-	.codec_dev = &soc_codec_dev_tlv320aic23,
-};
-
-static struct platform_device *osk_snd_device;
-
-static int __init osk_soc_init(void)
-{
-	int err;
-	u32 curRate;
-	struct device *dev;
-
-	if (!(machine_is_omap_osk()))
-		return -ENODEV;
-
-	osk_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!osk_snd_device)
-		return -ENOMEM;
-
-	platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
-	osk_snd_devdata.dev = &osk_snd_device->dev;
-	*(unsigned int *)osk_dai.cpu_dai->private_data = 0;	/* McBSP1 */
-	err = platform_device_add(osk_snd_device);
-	if (err)
-		goto err1;
-
-	dev = &osk_snd_device->dev;
-
-	tlv320aic23_mclk = clk_get(dev, "mclk");
-	if (IS_ERR(tlv320aic23_mclk)) {
-		printk(KERN_ERR "Could not get mclk clock\n");
-		return -ENODEV;
-	}
-
-	if (clk_get_usecount(tlv320aic23_mclk) > 0) {
-		/* MCLK is already in use */
-		printk(KERN_WARNING
-		       "MCLK in use at %d Hz. We change it to %d Hz\n",
-		       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
-	}
-
-	/*
-	 * Configure 12 MHz output on MCLK.
-	 */
-	curRate = (uint) clk_get_rate(tlv320aic23_mclk);
-	if (curRate != CODEC_CLOCK) {
-		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
-			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
-			err = -ECANCELED;
-			goto err1;
-		}
-	}
-
-	printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
-	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
-	       clk_get_usecount(tlv320aic23_mclk));
-
-	return 0;
-err1:
-	clk_put(tlv320aic23_mclk);
-	platform_device_del(osk_snd_device);
-	platform_device_put(osk_snd_device);
-
-	return err;
-
-}
-
-static void __exit osk_soc_exit(void)
-{
-	platform_device_unregister(osk_snd_device);
-}
-
-module_init(osk_soc_init);
-module_exit(osk_soc_exit);
-
-MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
-MODULE_DESCRIPTION("ALSA SoC OSK 5912");
-MODULE_LICENSE("GPL");
diff --git a/trunk/sound/soc/pxa/corgi.c b/trunk/sound/soc/pxa/corgi.c
index dd7fa0b329c7..72b7a5140bf8 100644
--- a/trunk/sound/soc/pxa/corgi.c
+++ b/trunk/sound/soc/pxa/corgi.c
@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -289,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_nc_pin(codec, "LLINEIN");
-	snd_soc_dapm_nc_pin(codec, "RLINEIN");
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
 
 	/* Add corgi specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
diff --git a/trunk/sound/soc/pxa/em-x270.c b/trunk/sound/soc/pxa/em-x270.c
index e6ff6929ab4b..d9c3f7b28be2 100644
--- a/trunk/sound/soc/pxa/em-x270.c
+++ b/trunk/sound/soc/pxa/em-x270.c
@@ -9,7 +9,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/trunk/sound/soc/pxa/poodle.c b/trunk/sound/soc/pxa/poodle.c
index 4d9930c52789..f84f7d8db09a 100644
--- a/trunk/sound/soc/pxa/poodle.c
+++ b/trunk/sound/soc/pxa/poodle.c
@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -242,8 +242,8 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_nc_pin(codec, "LLINEIN");
-	snd_soc_dapm_nc_pin(codec, "RLINEIN");
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
 	snd_soc_dapm_enable_pin(codec, "MICIN");
 
 	/* Add poodle specific controls */
diff --git a/trunk/sound/soc/pxa/pxa2xx-i2s.c b/trunk/sound/soc/pxa/pxa2xx-i2s.c
index 64057b1d220d..39d19212f6d3 100644
--- a/trunk/sound/soc/pxa/pxa2xx-i2s.c
+++ b/trunk/sound/soc/pxa/pxa2xx-i2s.c
@@ -3,7 +3,7 @@
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood
- *         lrg@slimlogic.co.uk
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -366,6 +366,6 @@ module_init(pxa2xx_i2s_init);
 module_exit(pxa2xx_i2s_exit);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
 MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/trunk/sound/soc/pxa/spitz.c b/trunk/sound/soc/pxa/spitz.c
index 8f89188e541e..3d4738c06e7e 100644
--- a/trunk/sound/soc/pxa/spitz.c
+++ b/trunk/sound/soc/pxa/spitz.c
@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -291,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	int i, err;
 
 	/* NC codec pins */
-	snd_soc_dapm_nc_pin(codec, "RINPUT1");
-	snd_soc_dapm_nc_pin(codec, "LINPUT2");
-	snd_soc_dapm_nc_pin(codec, "RINPUT2");
-	snd_soc_dapm_nc_pin(codec, "LINPUT3");
-	snd_soc_dapm_nc_pin(codec, "RINPUT3");
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "MONO1");
+	snd_soc_dapm_disable_pin(codec, "RINPUT1");
+	snd_soc_dapm_disable_pin(codec, "LINPUT2");
+	snd_soc_dapm_disable_pin(codec, "RINPUT2");
+	snd_soc_dapm_disable_pin(codec, "LINPUT3");
+	snd_soc_dapm_disable_pin(codec, "RINPUT3");
+	snd_soc_dapm_disable_pin(codec, "OUT3");
+	snd_soc_dapm_disable_pin(codec, "MONO1");
 
 	/* Add spitz specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
diff --git a/trunk/sound/soc/pxa/tosa.c b/trunk/sound/soc/pxa/tosa.c
index afefe41b8c46..2baaa750f123 100644
--- a/trunk/sound/soc/pxa/tosa.c
+++ b/trunk/sound/soc/pxa/tosa.c
@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -190,8 +190,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "MONOOUT");
+	snd_soc_dapm_disable_pin(codec, "OUT3");
+	snd_soc_dapm_disable_pin(codec, "MONOOUT");
 
 	/* add tosa specific controls */
 	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
diff --git a/trunk/sound/soc/s3c24xx/neo1973_wm8753.c b/trunk/sound/soc/s3c24xx/neo1973_wm8753.c
index 87ddfefcc2fb..73a50e93a9a2 100644
--- a/trunk/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/trunk/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -511,20 +511,21 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 	DBG("Entered %s\n", __func__);
 
 	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(codec, "LOUT2");
-	snd_soc_dapm_nc_pin(codec, "ROUT2");
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "OUT4");
-	snd_soc_dapm_nc_pin(codec, "LINE1");
-	snd_soc_dapm_nc_pin(codec, "LINE2");
+	snd_soc_dapm_disable_pin(codec, "LOUT2");
+	snd_soc_dapm_disable_pin(codec, "ROUT2");
+	snd_soc_dapm_disable_pin(codec, "OUT3");
+	snd_soc_dapm_disable_pin(codec, "OUT4");
+	snd_soc_dapm_disable_pin(codec, "LINE1");
+	snd_soc_dapm_disable_pin(codec, "LINE2");
 
-	/* Add neo1973 specific widgets */
-	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
-				  ARRAY_SIZE(wm8753_dapm_widgets));
 
 	/* set endpoints to default mode */
 	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
 
+	/* Add neo1973 specific widgets */
+	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
+
 	/* add neo1973 specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
 		err = snd_ctl_add(codec->card,
@@ -602,8 +603,6 @@ static int lm4857_i2c_probe(struct i2c_client *client,
 {
 	DBG("Entered %s\n", __func__);
 
-	i2c = client;
-
 	lm4857_write_regs();
 	return 0;
 }
@@ -612,8 +611,6 @@ static int lm4857_i2c_remove(struct i2c_client *client)
 {
 	DBG("Entered %s\n", __func__);
 
-	i2c = NULL;
-
 	return 0;
 }
 
@@ -653,7 +650,7 @@ static void lm4857_shutdown(struct i2c_client *dev)
 }
 
 static const struct i2c_device_id lm4857_i2c_id[] = {
-	{ "neo1973_lm4857", 0 },
+	{ "neo1973_lm4857", 0 }
 	{ }
 };
 
@@ -671,6 +668,48 @@ static struct i2c_driver lm4857_i2c_driver = {
 };
 
 static struct platform_device *neo1973_snd_device;
+static struct i2c_client *lm4857_client;
+
+static int __init neo1973_add_lm4857_device(struct platform_device *pdev,
+					    int i2c_bus,
+					    unsigned short i2c_address)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+	int ret;
+
+	ret = i2c_add_driver(&lm4857_i2c_driver);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "can't add lm4857 driver\n");
+		return ret;
+	}
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = i2c_address;
+	strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(i2c_bus);
+	if (!adapter) {
+		dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus);
+		goto err_driver;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n",
+			(unsigned int)info.addr);
+		goto err_driver;
+	}
+
+	lm4857_client = client;
+	return 0;
+
+err_driver:
+	i2c_del_driver(&lm4857_i2c_driver);
+	return -ENODEV;
+}
 
 static int __init neo1973_init(void)
 {
@@ -697,8 +736,8 @@ static int __init neo1973_init(void)
 		return ret;
 	}
 
-	ret = i2c_add_driver(&lm4857_i2c_driver);
-
+	ret = neo1973_add_lm4857_device(neo1973_snd_device,
+					neo1973_wm8753_setup, 0x7C);
 	if (ret != 0)
 		platform_device_unregister(neo1973_snd_device);
 
@@ -709,6 +748,7 @@ static void __exit neo1973_exit(void)
 {
 	DBG("Entered %s\n", __func__);
 
+	i2c_unregister_device(lm4857_client);
 	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
 }
diff --git a/trunk/sound/soc/soc-core.c b/trunk/sound/soc/soc-core.c
index 0af3bbde5bbe..ad381138fc2e 100644
--- a/trunk/sound/soc/soc-core.c
+++ b/trunk/sound/soc/soc-core.c
@@ -4,7 +4,8 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
  *         with code, comments and ideas from :-
  *         Richard Purdie <richard@openedhand.com>
  *
@@ -95,8 +96,8 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 	codec->ac97->dev.parent = NULL;
 	codec->ac97->dev.release = soc_ac97_device_release;
 
-	dev_set_name(&codec->ac97->dev, "%d-%d:%s",
-		     codec->card->number, 0, codec->name);
+	snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
+		 codec->card->number, 0, codec->name);
 	err = device_register(&codec->ac97->dev);
 	if (err < 0) {
 		snd_printk(KERN_ERR "Can't register ac97 bus\n");
@@ -1885,7 +1886,7 @@ module_init(snd_soc_init);
 module_exit(snd_soc_exit);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
 MODULE_DESCRIPTION("ALSA SoC Core");
 MODULE_LICENSE("GPL");
 MODULE_ALIAS("platform:soc-audio");
diff --git a/trunk/sound/soc/soc-dapm.c b/trunk/sound/soc/soc-dapm.c
index efbd0b37810a..9ca9c08610fa 100644
--- a/trunk/sound/soc/soc-dapm.c
+++ b/trunk/sound/soc/soc-dapm.c
@@ -2,7 +2,8 @@
  * soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -1482,26 +1483,6 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
 
-/**
- * snd_soc_dapm_nc_pin - permanently disable pin.
- * @codec: SoC codec
- * @pin: pin name
- *
- * Marks the specified pin as being not connected, disabling it along
- * any parent or child widgets.  At present this is identical to
- * snd_soc_dapm_disable_pin() but in future it will be extended to do
- * additional things such as disabling controls which only affect
- * paths through the pin.
- *
- * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
- * do any widget power switching.
- */
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin)
-{
-	return snd_soc_dapm_set_pin(codec, pin, 0);
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
-
 /**
  * snd_soc_dapm_get_pin_status - get audio pin status
  * @codec: audio codec
@@ -1540,6 +1521,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev)
 EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
 MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
 MODULE_LICENSE("GPL");
diff --git a/trunk/sound/sound_core.c b/trunk/sound/sound_core.c
index ee66d43f848c..4ae07e236b36 100644
--- a/trunk/sound/sound_core.c
+++ b/trunk/sound/sound_core.c
@@ -57,7 +57,7 @@ module_exit(cleanup_soundcore);
 /*
  *	OSS sound core handling. Breaks out sound functions to submodules
  *	
- *	Author:		Alan Cox <alan@lxorguk.ukuu.org.uk>
+ *	Author:		Alan Cox <alan.cox@linux.org>
  *
  *	Fixes:
  *
diff --git a/trunk/sound/usb/usbquirks.h b/trunk/sound/usb/usbquirks.h
index 92115755d98e..69689e79bf79 100644
--- a/trunk/sound/usb/usbquirks.h
+++ b/trunk/sound/usb/usbquirks.h
@@ -1480,36 +1480,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
-{
-	/* Advanced modes of the Edirol UA-25EX.
-	 * For the standard mode, UA-25EX has ID 0582:00e7, which
-	 * offers only 16-bit PCM at 44.1 kHz and no MIDI.
-	 */
-	USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e6),
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "EDIROL",
-		.product_name = "UA-25EX",
-		.ifnum = QUIRK_ANY_INTERFACE,
-		.type = QUIRK_COMPOSITE,
-		.data = (const struct snd_usb_audio_quirk[]) {
-			{
-				.ifnum = 0,
-				.type = QUIRK_AUDIO_EDIROL_UAXX
-			},
-			{
-				.ifnum = 1,
-				.type = QUIRK_AUDIO_EDIROL_UAXX
-			},
-			{
-				.ifnum = 2,
-				.type = QUIRK_AUDIO_EDIROL_UAXX
-			},
-			{
-				.ifnum = -1
-			}
-		}
-	}
-},
 
 /* Guillemot devices */
 {
diff --git a/trunk/sound/usb/usx2y/us122l.c b/trunk/sound/usb/usx2y/us122l.c
index c2515b680f9f..b441fe2cd190 100644
--- a/trunk/sound/usb/usx2y/us122l.c
+++ b/trunk/sound/usb/usx2y/us122l.c
@@ -118,11 +118,12 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area,
 	void *vaddr;
 	struct us122l *us122l = area->vm_private_data;
 	struct usb_stream *s;
+	int vm_f = VM_FAULT_SIGBUS;
 
 	mutex_lock(&us122l->mutex);
 	s = us122l->sk.s;
 	if (!s)
-		goto unlock;
+		goto out;
 
 	offset = vmf->pgoff << PAGE_SHIFT;
 	if (offset < PAGE_ALIGN(s->read_size))
@@ -130,7 +131,7 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area,
 	else {
 		offset -= PAGE_ALIGN(s->read_size);
 		if (offset >= PAGE_ALIGN(s->write_size))
-			goto unlock;
+			goto out;
 
 		vaddr = us122l->sk.write_page + offset;
 	}
@@ -140,11 +141,9 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area,
 	mutex_unlock(&us122l->mutex);
 
 	vmf->page = page;
-
-	return 0;
-unlock:
-	mutex_unlock(&us122l->mutex);
-	return VM_FAULT_SIGBUS;
+	vm_f = 0;
+out:
+	return vm_f;
 }
 
 static void usb_stream_hwdep_vm_close(struct vm_area_struct *area)