diff --git a/[refs] b/[refs]
index 9a55a6f0df85..43387e614cba 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: 8cd2b264d92bb949ddccfb6db6a82254bb07af75
+refs/heads/master: 762b8df748d83c14070afbf0c6f8c0f4a91a13bf
diff --git a/trunk/include/sound/soc.h b/trunk/include/sound/soc.h
index a1e0357a84d7..5e0189876afd 100644
--- a/trunk/include/sound/soc.h
+++ b/trunk/include/sound/soc.h
@@ -28,7 +28,8 @@
  */
 #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
-	{.reg = xreg, .shift = xshift, .max = xmax, .invert = xinvert})
+	{.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \
+	.invert = xinvert})
 #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xreg, .max = xmax, .invert = xinvert})
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/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/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/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/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c
index c59065513118..c461baa83c2a 100644
--- a/trunk/sound/pci/hda/patch_sigmatel.c
+++ b/trunk/sound/pci/hda/patch_sigmatel.c
@@ -322,8 +322,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = {
 	0x1a, 0x1b
 };
 
-static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
-	0x1c, 0x1d,
+static hda_nid_t stac92hd71bxx_dmux_nids[1] = {
+	0x1c,
 };
 
 static hda_nid_t stac92hd71bxx_smux_nids[2] = {
@@ -861,18 +861,20 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* connect headphone jack to dac1 */
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
 	/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 };
 
-#define HD_DISABLE_PORTF 2
+#define HD_DISABLE_PORTF 3
 static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 	/* start of config #1 */
 
 	/* connect port 0f to audio mixer */
 	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
 	/* unmute right and left channels for node 0x0f */
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* start of config #2 */
@@ -881,6 +883,10 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* connect headphone jack to dac1 */
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
+	/* connect port 0d to audio mixer */
+	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2},
+	/* unmute dac0 input in audio mixer */
+	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
 	/* unmute right and left channels for nodes 0x0a, 0xd */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -1101,7 +1107,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 
 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	STAC_INPUT_SOURCE(2),
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
@@ -1114,17 +1119,8 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
 	*/
 
-	HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT),
-
-	HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT),
-
-	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),
-
-	HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
-	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -1653,7 +1649,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 
 static unsigned int ref92hd71bxx_pin_configs[11] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
-	0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
+	0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
 	0x90a000f0, 0x01452050, 0x01452050,
 };
 
@@ -3004,7 +3000,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
 
 /* labels for amp mux outputs */
 static const char *stac92xx_amp_labels[3] = {
-	"Front Microphone", "Microphone", "Line In",
+	"Front Microphone", "Microphone", "Line In"
 };
 
 /* create amp out controls mux on capable codecs */
@@ -4331,16 +4327,6 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = {
 #endif
 };
 
-static struct hda_input_mux stac92hd71bxx_dmux = {
-	.num_items = 4,
-	.items = {
-		{ "Analog Inputs", 0x00 },
-		{ "Mixer", 0x01 },
-		{ "Digital Mic 1", 0x02 },
-		{ "Digital Mic 2", 0x03 },
-	}
-};
-
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
@@ -4355,8 +4341,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 	spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
 	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
 	spec->pin_nids = stac92hd71bxx_pin_nids;
-	memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
-			sizeof(stac92hd71bxx_dmux));
 	spec->board_config = snd_hda_check_board_config(codec,
 							STAC_92HD71BXX_MODELS,
 							stac92hd71bxx_models,
@@ -4408,7 +4392,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 		/* no output amps */
 		spec->num_pwrs = 0;
 		spec->mixer = stac92hd71bxx_analog_mixer;
-		spec->dinput_mux = &spec->private_dimux;
 
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
@@ -4426,13 +4409,12 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
-		spec->dinput_mux = &spec->private_dimux;
 		spec->mixer = stac92hd71bxx_analog_mixer;
 		spec->init = stac92hd71bxx_analog_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 	}
 
-	spec->aloopback_mask = 0x50;
+	spec->aloopback_mask = 0x20;
 	spec->aloopback_shift = 0;
 
 	if (spec->board_config > STAC_92HD71BXX_REF) {
@@ -4474,10 +4456,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 	spec->multiout.num_dacs = 1;
 	spec->multiout.hp_nid = 0x11;
 	spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
-	if (spec->dinput_mux)
-		spec->private_dimux.num_items +=
-			spec->num_dmics -
-				(ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);
 
 	err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
 	if (!err) {
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/soc/at32/playpaq_wm8510.c b/trunk/sound/soc/at32/playpaq_wm8510.c
index 98a2d5826a85..b1966e4dfcd3 100644
--- a/trunk/sound/soc/at32/playpaq_wm8510.c
+++ b/trunk/sound/soc/at32/playpaq_wm8510.c
@@ -304,7 +304,7 @@ static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
 
 
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 	/* speaker connected to SPKOUT */
 	{"Ext Spk", NULL, "SPKOUTP"},
 	{"Ext Spk", NULL, "SPKOUTN"},
@@ -312,9 +312,6 @@ static const char *intercon[][3] = {
 	{"Mic Bias", NULL, "Int Mic"},
 	{"MICN", NULL, "Mic Bias"},
 	{"MICP", NULL, "Mic Bias"},
-
-	/* Terminator */
-	{NULL, NULL, NULL},
 };
 
 
@@ -334,11 +331,8 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec)
 	/*
 	 * Setup audio path interconnects
 	 */
-	for (i = 0; intercon[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec,
-					   intercon[i][0],
-					   intercon[i][1], intercon[i][2]);
-	}
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
 
 
 	/* always connected pins */
diff --git a/trunk/sound/soc/blackfin/bf5xx-i2s.c b/trunk/sound/soc/blackfin/bf5xx-i2s.c
index 827587f08180..e020c160ee44 100644
--- a/trunk/sound/soc/blackfin/bf5xx-i2s.c
+++ b/trunk/sound/soc/blackfin/bf5xx-i2s.c
@@ -70,12 +70,24 @@ 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},
-};
+/*
+ * Setting the TFS pin selector for SPORT 0 based on whether the selected
+ * port id F or G. If the port is F then no conflict should exist for the
+ * TFS. When Port G is selected and EMAC then there is a conflict between
+ * the PHY interrupt line and TFS.  Current settings prevent the conflict
+ * by ignoring the TFS pin when Port G is selected. This allows both
+ * ssm2602 using Port G and EMAC concurrently.
+ */
+#ifdef CONFIG_BF527_SPORT0_PORTF
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
+#else
+#define LOCAL_SPORT0_TFS (0)
+#endif
+
+static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
+		P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0},
+		{P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI,
+		P_SPORT1_RSCLK, P_SPORT1_TFS, 0} };
 
 static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		unsigned int fmt)
@@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		ret = -EINVAL;
 		break;
 	default:
+		printk(KERN_ERR "%s: Unknown DAI format type\n", __func__);
 		ret = -EINVAL;
 		break;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBS_CFS:
-		ret = -EINVAL;
-		break;
-	case SND_SOC_DAIFMT_CBM_CFS:
-		ret = -EINVAL;
-		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBM_CFS:
 	case SND_SOC_DAIFMT_CBS_CFM:
 		ret = -EINVAL;
 		break;
 	default:
+		printk(KERN_ERR "%s: Unknown DAI master type\n", __func__);
 		ret = -EINVAL;
 		break;
 	}
diff --git a/trunk/sound/soc/codecs/Kconfig b/trunk/sound/soc/codecs/Kconfig
index 4975d8573e4f..38a0e3b620a7 100644
--- a/trunk/sound/soc/codecs/Kconfig
+++ b/trunk/sound/soc/codecs/Kconfig
@@ -68,7 +68,7 @@ config SND_SOC_TLV320AIC23
 	depends on I2C
 
 config SND_SOC_TLV320AIC26
-	tristate "TI TLV320AIC26 Codec support"
+	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
 	depends on SPI
 
 config SND_SOC_TLV320AIC3X
diff --git a/trunk/sound/soc/codecs/tlv320aic23.c b/trunk/sound/soc/codecs/tlv320aic23.c
index bac7815e00fb..44308dac9e18 100644
--- a/trunk/sound/soc/codecs/tlv320aic23.c
+++ b/trunk/sound/soc/codecs/tlv320aic23.c
@@ -84,7 +84,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
 			     unsigned int value)
 {
 
-	u8 data;
+	u8 data[2];
 
 	/* TLV320AIC23 has 7 bit address and 9 bits of data
 	 * so we need to switch one data bit into reg and rest
@@ -96,12 +96,12 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
 		return -1;
 	}
 
-	data = (reg << 1) | (value >> 8 & 0x01);
+	data[0] = (reg << 1) | (value >> 8 & 0x01);
+	data[1] = value & 0xff;
 
 	tlv320aic23_write_reg_cache(codec, reg, value);
 
-	if (codec->hw_write(codec->control_data, data,
-			    (value & 0xff)) == 0)
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
 		return 0;
 
 	printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
@@ -674,7 +674,7 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 
 	tlv320aic23_socdev = socdev;
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data;
+	codec->hw_write = (hw_write_t) i2c_master_send;
 	codec->hw_read = NULL;
 	ret = i2c_add_driver(&tlv320aic23_i2c_driver);
 	if (ret != 0)
diff --git a/trunk/sound/soc/codecs/tlv320aic3x.c b/trunk/sound/soc/codecs/tlv320aic3x.c
index 05336ed7e493..cff276ee261e 100644
--- a/trunk/sound/soc/codecs/tlv320aic3x.c
+++ b/trunk/sound/soc/codecs/tlv320aic3x.c
@@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	/* interface format */
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_I2S:
+	/*
+	 * match both interface format and signal polarities since they
+	 * are fixed
+	 */
+	switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+		       SND_SOC_DAIFMT_INV_MASK)) {
+	case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
+	case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
 		iface_breg |= (0x01 << 6);
 		break;
-	case SND_SOC_DAIFMT_RIGHT_J:
+	case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
 		iface_breg |= (0x02 << 6);
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
+	case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
 		iface_breg |= (0x03 << 6);
 		break;
 	default:
diff --git a/trunk/sound/soc/omap/omap-mcbsp.c b/trunk/sound/soc/omap/omap-mcbsp.c
index 0a063a98a661..8485a8a9d0ff 100644
--- a/trunk/sound/soc/omap/omap-mcbsp.c
+++ b/trunk/sound/soc/omap/omap-mcbsp.c
@@ -43,6 +43,7 @@
 struct omap_mcbsp_data {
 	unsigned int			bus_id;
 	struct omap_mcbsp_reg_cfg	regs;
+	unsigned int			fmt;
 	/*
 	 * Flags indicating is the bus already activated and configured by
 	 * another substream
@@ -200,6 +201,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
+	int wlen;
 	unsigned long port;
 
 	if (cpu_class_is_omap1()) {
@@ -244,19 +246,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		/* Set word lengths */
+		wlen = 16;
 		regs->rcr2	|= RWDLEN2(OMAP_MCBSP_WORD_16);
 		regs->rcr1	|= RWDLEN1(OMAP_MCBSP_WORD_16);
 		regs->xcr2	|= XWDLEN2(OMAP_MCBSP_WORD_16);
 		regs->xcr1	|= XWDLEN1(OMAP_MCBSP_WORD_16);
-		/* Set FS period and length in terms of bit clock periods */
-		regs->srgr2	|= FPER(16 * 2 - 1);
-		regs->srgr1	|= FWID(16 - 1);
 		break;
 	default:
 		/* Unsupported PCM format */
 		return -EINVAL;
 	}
 
+	/* Set FS period and length in terms of bit clock periods */
+	switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		regs->srgr2	|= FPER(wlen * 2 - 1);
+		regs->srgr1	|= FWID(wlen - 1);
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		regs->srgr2	|= FPER(wlen * 2 - 1);
+		regs->srgr1	|= FWID(wlen * 2 - 2);
+		break;
+	}
+
 	omap_mcbsp_config(bus_id, &mcbsp_data->regs);
 	mcbsp_data->configured = 1;
 
@@ -276,6 +288,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	if (mcbsp_data->configured)
 		return 0;
 
+	mcbsp_data->fmt = fmt;
 	memset(regs, 0, sizeof(*regs));
 	/* Generic McBSP register settings */
 	regs->spcr2	|= XINTM(3) | FREE;
diff --git a/trunk/sound/soc/soc-core.c b/trunk/sound/soc/soc-core.c
index 462e635dfc74..a3adbf06b1e5 100644
--- a/trunk/sound/soc/soc-core.c
+++ b/trunk/sound/soc/soc-core.c
@@ -1462,7 +1462,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	int max = mc->max;
-	unsigned int shift = mc->min;
+	unsigned int shift = mc->shift;
 	unsigned int rshift = mc->rshift;
 
 	if (max == 1)
diff --git a/trunk/sound/soc/soc-dapm.c b/trunk/sound/soc/soc-dapm.c
index efbd0b37810a..7351db9606e4 100644
--- a/trunk/sound/soc/soc-dapm.c
+++ b/trunk/sound/soc/soc-dapm.c
@@ -831,7 +831,7 @@ int snd_soc_dapm_sys_add(struct device *dev)
 		return ret;
 
 	asoc_debugfs = debugfs_create_dir("asoc", NULL);
-	if (!IS_ERR(asoc_debugfs))
+	if (!IS_ERR(asoc_debugfs) && asoc_debugfs)
 		debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs,
 				   &pop_time);
 	else
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)