diff --git a/[refs] b/[refs]
index ffecb8580a47..2cc729099ace 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: ba9d0fd0f0af3eb7ec463847b409dd88f8e5c031
+refs/heads/master: a66547f3a1295d782fc0923d872b309994cea32a
diff --git a/trunk/sound/arm/pxa2xx-ac97-lib.c b/trunk/sound/arm/pxa2xx-ac97-lib.c
index 99026dfb81ea..34c1d94f921e 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_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS)
+	if (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_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS)
+	if (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_pxa21x() || cpu_is_pxa25x())
+	if (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_pxa21x() || cpu_is_pxa25x())
+	if (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_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) {
+	if (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_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) {
+	if (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/core/jack.c b/trunk/sound/core/jack.c
index 8133a2b173a5..bd2d9e6b55e9 100644
--- a/trunk/sound/core/jack.c
+++ b/trunk/sound/core/jack.c
@@ -147,6 +147,9 @@ 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 6ea5cfb83998..921691080f35 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
+ * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint
  * @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 mask parameter.
+ * Apply the constraint of the given bitmap mask to a 32-bit 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
+ * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint
  * @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 mask parameter.
+ * Apply the constraint of the given bitmap mask to a 64-bit 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
+ * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval
  * @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
+ * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval
  * @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
+ * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter
  * @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
+ * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter
  * @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
+ * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter
  * @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
+ * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule
  * @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
+ * snd_pcm_hw_constraint_step - add a hw constraint step rule
  * @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
+ * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule
  * @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
+ * snd_pcm_hw_param_value - return @params field @var 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 PAR if it's fixed in configuration space 
- *  defined by PARAMS. Return -EINVAL otherwise
+ * Return the value for field @var 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
+ * snd_pcm_hw_param_first - refine config space and return minimum value
  * @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 PAR all 
+ * Inside configuration space defined by @params remove from @var 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
+ * snd_pcm_hw_param_last - refine config space and return maximum value
  * @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 PAR all 
+ * Inside configuration space defined by @params remove from @var 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
+ * snd_pcm_hw_param_choose - choose a configuration defined by @params
  * @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 89b7f549bebd..ea2bf82c9373 100644
--- a/trunk/sound/core/pcm_misc.c
+++ b/trunk/sound/core/pcm_misc.c
@@ -319,6 +319,7 @@ 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 e61e12506ded..aef18682c035 100644
--- a/trunk/sound/core/pcm_native.c
+++ b/trunk/sound/core/pcm_native.c
@@ -875,10 +875,8 @@ static struct action_ops snd_pcm_action_start = {
 };
 
 /**
- * snd_pcm_start
+ * snd_pcm_start - start all linked streams
  * @substream: the PCM substream instance
- *
- * Start all linked streams.
  */
 int snd_pcm_start(struct snd_pcm_substream *substream)
 {
@@ -926,12 +924,11 @@ static struct action_ops snd_pcm_action_stop = {
 };
 
 /**
- * snd_pcm_stop
+ * snd_pcm_stop - try to stop all running streams in the substream group
  * @substream: the PCM substream instance
  * @state: PCM state after stopping the stream
  *
- * Try to stop all running streams in the substream group.
- * The state of each stream is changed to the given value after that unconditionally.
+ * The state of each stream is then changed to the given state unconditionally.
  */
 int snd_pcm_stop(struct snd_pcm_substream *substream, int state)
 {
@@ -941,11 +938,10 @@ int snd_pcm_stop(struct snd_pcm_substream *substream, int state)
 EXPORT_SYMBOL(snd_pcm_stop);
 
 /**
- * snd_pcm_drain_done
+ * snd_pcm_drain_done - stop the DMA only when the given stream is playback
  * @substream: the PCM substream
  *
- * Stop the DMA only when the given stream is playback.
- * The state is changed to SETUP.
+ * After stopping, 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)
@@ -1065,10 +1061,9 @@ static struct action_ops snd_pcm_action_suspend = {
 };
 
 /**
- * snd_pcm_suspend
+ * snd_pcm_suspend - trigger SUSPEND to all linked streams
  * @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)
@@ -1088,10 +1083,9 @@ int snd_pcm_suspend(struct snd_pcm_substream *substream)
 EXPORT_SYMBOL(snd_pcm_suspend);
 
 /**
- * snd_pcm_suspend_all
+ * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm
  * @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)
@@ -1313,11 +1307,9 @@ static struct action_ops snd_pcm_action_prepare = {
 };
 
 /**
- * snd_pcm_prepare
+ * snd_pcm_prepare - prepare the PCM substream to be triggerable
  * @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/pci/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c
index c461baa83c2a..c59065513118 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[1] = {
-	0x1c,
+static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
+	0x1c, 0x1d,
 };
 
 static hda_nid_t stac92hd71bxx_smux_nids[2] = {
@@ -861,20 +861,18 @@ 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 3
+#define HD_DISABLE_PORTF 2
 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 */
@@ -883,10 +881,6 @@ 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)},
@@ -1107,6 +1101,7 @@ 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),
@@ -1119,8 +1114,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
 	*/
 
-	HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, 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),
 	{ } /* end */
 };
 
@@ -1649,7 +1653,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 
 static unsigned int ref92hd71bxx_pin_configs[11] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
-	0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
+	0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
 	0x90a000f0, 0x01452050, 0x01452050,
 };
 
@@ -3000,7 +3004,7 @@ 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 */
@@ -4327,6 +4331,16 @@ 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;
@@ -4341,6 +4355,8 @@ 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,
@@ -4392,6 +4408,7 @@ 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];
@@ -4409,12 +4426,13 @@ 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 = 0x20;
+	spec->aloopback_mask = 0x50;
 	spec->aloopback_shift = 0;
 
 	if (spec->board_config > STAC_92HD71BXX_REF) {
@@ -4456,6 +4474,10 @@ 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/soc/at32/playpaq_wm8510.c b/trunk/sound/soc/at32/playpaq_wm8510.c
index b1966e4dfcd3..98a2d5826a85 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 struct snd_soc_dapm_route intercon[] = {
+static const char *intercon[][3] = {
 	/* speaker connected to SPKOUT */
 	{"Ext Spk", NULL, "SPKOUTP"},
 	{"Ext Spk", NULL, "SPKOUTN"},
@@ -312,6 +312,9 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Mic Bias", NULL, "Int Mic"},
 	{"MICN", NULL, "Mic Bias"},
 	{"MICP", NULL, "Mic Bias"},
+
+	/* Terminator */
+	{NULL, NULL, NULL},
 };
 
 
@@ -331,8 +334,11 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec)
 	/*
 	 * Setup audio path interconnects
 	 */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
-
+	for (i = 0; intercon[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec,
+					   intercon[i][0],
+					   intercon[i][1], intercon[i][2]);
+	}
 
 
 	/* always connected pins */
diff --git a/trunk/sound/soc/codecs/Kconfig b/trunk/sound/soc/codecs/Kconfig
index 38a0e3b620a7..4975d8573e4f 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" if SND_SOC_OF_SIMPLE
+	tristate "TI TLV320AIC26 Codec support"
 	depends on SPI
 
 config SND_SOC_TLV320AIC3X
diff --git a/trunk/sound/soc/codecs/tlv320aic23.c b/trunk/sound/soc/codecs/tlv320aic23.c
index 44308dac9e18..bac7815e00fb 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[2];
+	u8 data;
 
 	/* 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[0] = (reg << 1) | (value >> 8 & 0x01);
-	data[1] = value & 0xff;
+	data = (reg << 1) | (value >> 8 & 0x01);
 
 	tlv320aic23_write_reg_cache(codec, reg, value);
 
-	if (codec->hw_write(codec->control_data, data, 2) == 2)
+	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__,
@@ -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_master_send;
+	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)
diff --git a/trunk/sound/soc/omap/omap-mcbsp.c b/trunk/sound/soc/omap/omap-mcbsp.c
index 853b33ae3435..0a063a98a661 100644
--- a/trunk/sound/soc/omap/omap-mcbsp.c
+++ b/trunk/sound/soc/omap/omap-mcbsp.c
@@ -43,7 +43,6 @@
 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
@@ -201,7 +200,6 @@ 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()) {
@@ -246,29 +244,19 @@ 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(0);
-		break;
-	}
-
 	omap_mcbsp_config(bus_id, &mcbsp_data->regs);
 	mcbsp_data->configured = 1;
 
@@ -284,12 +272,10 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 {
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
-	unsigned int temp_fmt = fmt;
 
 	if (mcbsp_data->configured)
 		return 0;
 
-	mcbsp_data->fmt = fmt;
 	memset(regs, 0, sizeof(*regs));
 	/* Generic McBSP register settings */
 	regs->spcr2	|= XINTM(3) | FREE;
@@ -307,8 +293,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		/* 0-bit data delay */
 		regs->rcr2      |= RDATDLY(0);
 		regs->xcr2      |= XDATDLY(0);
-		/* Invert bit clock and FS polarity configuration for DSP_A */
-		temp_fmt ^= SND_SOC_DAIFMT_IB_IF;
 		break;
 	default:
 		/* Unsupported data format */
@@ -332,7 +316,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	}
 
 	/* Set bit clock (CLKX/CLKR) and FS polarities */
-	switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
 		/*
 		 * Normal BCLK + FS.
diff --git a/trunk/sound/soc/soc-dapm.c b/trunk/sound/soc/soc-dapm.c
index 7351db9606e4..efbd0b37810a 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) && asoc_debugfs)
+	if (!IS_ERR(asoc_debugfs))
 		debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs,
 				   &pop_time);
 	else
diff --git a/trunk/sound/usb/usx2y/us122l.c b/trunk/sound/usb/usx2y/us122l.c
index b441fe2cd190..c2515b680f9f 100644
--- a/trunk/sound/usb/usx2y/us122l.c
+++ b/trunk/sound/usb/usx2y/us122l.c
@@ -118,12 +118,11 @@ 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 out;
+		goto unlock;
 
 	offset = vmf->pgoff << PAGE_SHIFT;
 	if (offset < PAGE_ALIGN(s->read_size))
@@ -131,7 +130,7 @@ 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 out;
+			goto unlock;
 
 		vaddr = us122l->sk.write_page + offset;
 	}
@@ -141,9 +140,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area,
 	mutex_unlock(&us122l->mutex);
 
 	vmf->page = page;
-	vm_f = 0;
-out:
-	return vm_f;
+
+	return 0;
+unlock:
+	mutex_unlock(&us122l->mutex);
+	return VM_FAULT_SIGBUS;
 }
 
 static void usb_stream_hwdep_vm_close(struct vm_area_struct *area)