diff --git a/[refs] b/[refs]
index 0d506d785482..cdb56f24ec3b 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: 20c76945d05e26dc24eabcfc2f61f3929590cc29
+refs/heads/master: fad9365bcc2a69ae16adc092e8ac192354980665
diff --git a/trunk/Documentation/devicetree/bindings/sound/sgtl5000.txt b/trunk/Documentation/devicetree/bindings/sound/sgtl5000.txt
index 9cc44449508d..2c3cd413f042 100644
--- a/trunk/Documentation/devicetree/bindings/sound/sgtl5000.txt
+++ b/trunk/Documentation/devicetree/bindings/sound/sgtl5000.txt
@@ -3,8 +3,6 @@
 Required properties:
 - compatible : "fsl,sgtl5000".
 
-- reg : the I2C address of the device
-
 Example:
 
 codec: sgtl5000@0a {
diff --git a/trunk/Documentation/sound/alsa/HD-Audio-Models.txt b/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
index 03f7897c6414..d97d992ced14 100644
--- a/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -43,9 +43,7 @@ ALC680
 
 ALC882/883/885/888/889
 ======================
-  acer-aspire-4930g	Acer Aspire 4930G/5930G/6530G/6930G/7730G
-  acer-aspire-8930g	Acer Aspire 8330G/6935G
-  acer-aspire		Acer Aspire others
+  N/A
 
 ALC861/660
 ==========
diff --git a/trunk/include/sound/core.h b/trunk/include/sound/core.h
index bc056687f647..b6e0f57d451d 100644
--- a/trunk/include/sound/core.h
+++ b/trunk/include/sound/core.h
@@ -325,13 +325,6 @@ void release_and_free_resource(struct resource *res);
 
 /* --- */
 
-/* sound printk debug levels */
-enum {
-	SND_PR_ALWAYS,
-	SND_PR_DEBUG,
-	SND_PR_VERBOSE,
-};
-
 #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
 __printf(4, 5)
 void __snd_printk(unsigned int level, const char *file, int line,
@@ -361,8 +354,6 @@ void __snd_printk(unsigned int level, const char *file, int line,
  */
 #define snd_printd(fmt, args...) \
 	__snd_printk(1, __FILE__, __LINE__, fmt, ##args)
-#define _snd_printd(level, fmt, args...) \
-	__snd_printk(level, __FILE__, __LINE__, fmt, ##args)
 
 /**
  * snd_BUG - give a BUG warning message and stack trace
@@ -392,7 +383,6 @@ void __snd_printk(unsigned int level, const char *file, int line,
 #else /* !CONFIG_SND_DEBUG */
 
 #define snd_printd(fmt, args...)	do { } while (0)
-#define _snd_printd(level, fmt, args...) do { } while (0)
 #define snd_BUG()			do { } while (0)
 static inline int __snd_bug_on(int cond)
 {
diff --git a/trunk/sound/core/vmaster.c b/trunk/sound/core/vmaster.c
index 857586135d18..14a286a7bf2b 100644
--- a/trunk/sound/core/vmaster.c
+++ b/trunk/sound/core/vmaster.c
@@ -419,7 +419,6 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_master);
  * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control
  * @kcontrol: vmaster kctl element
  * @hook: the hook function
- * @private_data: the private_data pointer to be saved
  *
  * Adds the given hook to the vmaster control element so that it's called
  * at each time when the value is changed.
diff --git a/trunk/sound/isa/sscape.c b/trunk/sound/isa/sscape.c
index 8490f59709bb..b4a6aa960f4b 100644
--- a/trunk/sound/isa/sscape.c
+++ b/trunk/sound/isa/sscape.c
@@ -1019,15 +1019,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
 	irq_cfg = get_irq_config(sscape->type, irq[dev]);
 	if (irq_cfg == INVALID_IRQ) {
 		snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]);
-		err = -ENXIO;
-		goto _release_dma;
+		return -ENXIO;
 	}
 
 	mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]);
 	if (mpu_irq_cfg == INVALID_IRQ) {
 		snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]);
-		err = -ENXIO;
-		goto _release_dma;
+		return -ENXIO;
 	}
 
 	/*
diff --git a/trunk/sound/last.c b/trunk/sound/last.c
index 7ffc182e0844..bdd0857b8871 100644
--- a/trunk/sound/last.c
+++ b/trunk/sound/last.c
@@ -38,4 +38,4 @@ static int __init alsa_sound_last_init(void)
 	return 0;
 }
 
-late_initcall_sync(alsa_sound_last_init);
+__initcall(alsa_sound_last_init);
diff --git a/trunk/sound/oss/msnd_pinnacle.c b/trunk/sound/oss/msnd_pinnacle.c
index 536c4c0514d3..2c79d60a725f 100644
--- a/trunk/sound/oss/msnd_pinnacle.c
+++ b/trunk/sound/oss/msnd_pinnacle.c
@@ -1294,8 +1294,6 @@ static int __init calibrate_adc(WORD srate)
 
 static int upload_dsp_code(void)
 {
-	int ret = 0;
-
 	msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS);
 #ifndef HAVE_DSPCODEH
 	INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE);
@@ -1314,8 +1312,7 @@ static int upload_dsp_code(void)
 	memcpy_toio(dev.base, PERMCODE, PERMCODESIZE);
 	if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) {
 		printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n");
-		ret = -ENODEV;
-		goto out;
+		return -ENODEV;
 	}
 #ifdef HAVE_DSPCODEH
 	printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n");
@@ -1323,13 +1320,12 @@ static int upload_dsp_code(void)
 	printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n");
 #endif
 
-out:
 #ifndef HAVE_DSPCODEH
 	vfree(INITCODE);
 	vfree(PERMCODE);
 #endif
 
-	return ret;
+	return 0;
 }
 
 #ifdef MSND_CLASSIC
diff --git a/trunk/sound/pci/asihpi/hpi_internal.h b/trunk/sound/pci/asihpi/hpi_internal.h
index bc86cb726d79..8c63200cf339 100644
--- a/trunk/sound/pci/asihpi/hpi_internal.h
+++ b/trunk/sound/pci/asihpi/hpi_internal.h
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2012  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned.
 If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and
 HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle.
 */
-u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
+int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
 							   /**< memory handle */
 	u32 size, /**< Size in bytes to allocate */
 	struct pci_dev *p_os_reference
diff --git a/trunk/sound/pci/asihpi/hpios.c b/trunk/sound/pci/asihpi/hpios.c
index 5ef4fe964366..87f4385fe8c7 100644
--- a/trunk/sound/pci/asihpi/hpios.c
+++ b/trunk/sound/pci/asihpi/hpios.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2012  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -39,11 +39,11 @@ void hpios_delay_micro_seconds(u32 num_micro_sec)
 
 }
 
-/** Allocate an area of locked memory for bus master DMA operations.
+/** Allocated an area of locked memory for bus master DMA operations.
 
-If allocation fails, return 1, and *pMemArea.size = 0
+On error, return -ENOMEM, and *pMemArea.size = 0
 */
-u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
+int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
 	struct pci_dev *pdev)
 {
 	/*?? any benefit in using managed dmam_alloc_coherent? */
@@ -62,7 +62,7 @@ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
 		HPI_DEBUG_LOG(WARNING,
 			"failed to allocate %d bytes locked memory\n", size);
 		p_mem_area->size = 0;
-		return 1;
+		return -ENOMEM;
 	}
 }
 
diff --git a/trunk/sound/pci/hda/hda_codec.h b/trunk/sound/pci/hda/hda_codec.h
index 56b4f74c0b13..9a9f372e1be4 100644
--- a/trunk/sound/pci/hda/hda_codec.h
+++ b/trunk/sound/pci/hda/hda_codec.h
@@ -851,9 +851,6 @@ struct hda_codec {
 	unsigned int pin_amp_workaround:1; /* pin out-amp takes index
 					    * (e.g. Conexant codecs)
 					    */
-	unsigned int single_adc_amp:1; /* adc in-amp takes no index
-					* (e.g. CX20549 codec)
-					*/
 	unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */
 	unsigned int pins_shutup:1;	/* pins are shut up */
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
diff --git a/trunk/sound/pci/hda/hda_eld.c b/trunk/sound/pci/hda/hda_eld.c
index 4c054f4486b9..b58b4b1687fa 100644
--- a/trunk/sound/pci/hda/hda_eld.c
+++ b/trunk/sound/pci/hda/hda_eld.c
@@ -418,7 +418,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
 	else
 		buf2[0] = '\0';
 
-	_snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:"
+	printk(KERN_INFO "HDMI: supports coding type %s:"
 			" channels = %d, rates =%s%s\n",
 			cea_audio_coding_type_names[a->format],
 			a->channels,
@@ -442,14 +442,14 @@ void snd_hdmi_show_eld(struct hdmi_eld *e)
 {
 	int i;
 
-	_snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n",
+	printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n",
 			e->monitor_name,
 			eld_connection_type_names[e->conn_type]);
 
 	if (e->spk_alloc) {
 		char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
 		snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
-		_snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf);
+		printk(KERN_INFO "HDMI: available speakers:%s\n", buf);
 	}
 
 	for (i = 0; i < e->sad_count; i++)
diff --git a/trunk/sound/pci/hda/hda_proc.c b/trunk/sound/pci/hda/hda_proc.c
index e59e2f059b6e..254ab5204603 100644
--- a/trunk/sound/pci/hda/hda_proc.c
+++ b/trunk/sound/pci/hda/hda_proc.c
@@ -651,16 +651,9 @@ static void print_codec_info(struct snd_info_entry *entry,
 			snd_iprintf(buffer, "  Amp-In caps: ");
 			print_amp_caps(buffer, codec, nid, HDA_INPUT);
 			snd_iprintf(buffer, "  Amp-In vals: ");
-			if (wid_type == AC_WID_PIN ||
-			    (codec->single_adc_amp &&
-			     wid_type == AC_WID_AUD_IN))
-				print_amp_vals(buffer, codec, nid, HDA_INPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       1);
-			else
-				print_amp_vals(buffer, codec, nid, HDA_INPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       conn_len);
+			print_amp_vals(buffer, codec, nid, HDA_INPUT,
+				       wid_caps & AC_WCAP_STEREO,
+				       wid_type == AC_WID_PIN ? 1 : conn_len);
 		}
 		if (wid_caps & AC_WCAP_OUT_AMP) {
 			snd_iprintf(buffer, "  Amp-Out caps: ");
diff --git a/trunk/sound/pci/hda/patch_conexant.c b/trunk/sound/pci/hda/patch_conexant.c
index d906c5b74cf0..8c6523bbc797 100644
--- a/trunk/sound/pci/hda/patch_conexant.c
+++ b/trunk/sound/pci/hda/patch_conexant.c
@@ -141,6 +141,7 @@ struct conexant_spec {
 	unsigned int hp_laptop:1;
 	unsigned int asus:1;
 	unsigned int pin_eapd_ctrls:1;
+	unsigned int single_adc_amp:1;
 
 	unsigned int adc_switching:1;
 
@@ -686,26 +687,27 @@ static const struct hda_channel_mode cxt5045_modes[1] = {
 static const struct hda_input_mux cxt5045_capture_source = {
 	.num_items = 2,
 	.items = {
-		{ "Internal Mic", 0x1 },
-		{ "Mic",          0x2 },
+		{ "IntMic", 0x1 },
+		{ "ExtMic", 0x2 },
 	}
 };
 
 static const struct hda_input_mux cxt5045_capture_source_benq = {
-	.num_items = 4,
+	.num_items = 5,
 	.items = {
-		{ "Internal Mic", 0x1 },
-		{ "Mic",          0x2 },
-		{ "Line",         0x3 },
-		{ "Mixer",        0x0 },
+		{ "IntMic", 0x1 },
+		{ "ExtMic", 0x2 },
+		{ "LineIn", 0x3 },
+		{ "CD",     0x4 },
+		{ "Mixer",  0x0 },
 	}
 };
 
 static const struct hda_input_mux cxt5045_capture_source_hp530 = {
 	.num_items = 2,
 	.items = {
-		{ "Mic",          0x1 },
-		{ "Internal Mic", 0x2 },
+		{ "ExtMic", 0x1 },
+		{ "IntMic", 0x2 },
 	}
 };
 
@@ -796,8 +798,10 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec,
 }
 
 static const struct snd_kcontrol_new cxt5045_mixers[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
@@ -818,15 +822,27 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = {
 };
 
 static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
+	HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
 
 	{}
 };
 
 static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
@@ -930,10 +946,10 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 	/* Output controls */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
 	
 	/* Modes for retasking pin widgets */
 	CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT),
@@ -944,16 +960,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 
 	/* Loopback mixer controls */
 
-	HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT),
-	HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT),
-	HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Input Source",
@@ -962,8 +978,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 		.put = conexant_mux_enum_put,
 	},
 	/* Audio input controls */
-	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -985,6 +1009,10 @@ static const struct hda_verb cxt5045_test_init_verbs[] = {
 	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
 
+	/* Start with output sum widgets muted and their output gains at min */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
 	/* Unmute retasking pin widget output buffers since the default
 	 * state appears to be output.  As the pin mode is changed by the
 	 * user the pin mode control will take care of enabling the pin's
@@ -999,11 +1027,11 @@ static const struct hda_verb cxt5045_test_init_verbs[] = {
 	/* Set ADC connection select to match default mixer setting (mic1
 	 * pin)
 	 */
-	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
 
 	/* Mute all inputs to mixer widget (even unconnected ones) */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */
@@ -1082,7 +1110,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
-	codec->single_adc_amp = 1;
+	codec->pin_amp_workaround = 1;
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
@@ -3971,14 +3999,9 @@ static void cx_auto_init_output(struct hda_codec *codec)
 	int i;
 
 	mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids);
-	for (i = 0; i < cfg->hp_outs; i++) {
-		unsigned int val = PIN_OUT;
-		if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) &
-		    AC_PINCAP_HP_DRV)
-			val |= AC_PINCTL_HP_EN;
+	for (i = 0; i < cfg->hp_outs; i++)
 		snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, val);
-	}
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
 	mute_outputs(codec, cfg->hp_outs, cfg->hp_pins);
 	mute_outputs(codec, cfg->line_outs, cfg->line_out_pins);
 	mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins);
@@ -4197,7 +4220,7 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
 		int idx = get_input_connection(codec, adc_nid, nid);
 		if (idx < 0)
 			continue;
-		if (codec->single_adc_amp)
+		if (spec->single_adc_amp)
 			idx = 0;
 		return cx_auto_add_volume_idx(codec, label, pfx,
 					      cidx, adc_nid, HDA_INPUT, idx);
@@ -4252,7 +4275,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
 		if (cidx < 0)
 			continue;
 		input_conn[i] = spec->imux_info[i].adc;
-		if (!codec->single_adc_amp)
+		if (!spec->single_adc_amp)
 			input_conn[i] |= cidx << 8;
 		if (i > 0 && input_conn[i] != input_conn[0])
 			multi_connection = 1;
@@ -4396,10 +4419,8 @@ static void apply_pin_fixup(struct hda_codec *codec,
 
 enum {
 	CXT_PINCFG_LENOVO_X200,
-	CXT_PINCFG_LENOVO_TP410,
 };
 
-/* ThinkPad X200 & co with cxt5051 */
 static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
 	{ 0x17, 0x21a11000 }, /* dock-mic */
@@ -4408,33 +4429,15 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
 	{}
 };
 
-/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */
-static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = {
-	{ 0x19, 0x042110ff }, /* HP (seq# overridden) */
-	{ 0x1a, 0x21a190f0 }, /* dock-mic */
-	{ 0x1c, 0x212140ff }, /* dock-HP */
-	{}
-};
-
 static const struct cxt_pincfg *cxt_pincfg_tbl[] = {
 	[CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200,
-	[CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410,
 };
 
-static const struct snd_pci_quirk cxt5051_fixups[] = {
+static const struct snd_pci_quirk cxt_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
 	{}
 };
 
-static const struct snd_pci_quirk cxt5066_fixups[] = {
-	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
-	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
-	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
-	SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
-	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
-	{}
-};
-
 /* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
  * can be created (bko#42825)
  */
@@ -4463,21 +4466,19 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->pin_amp_workaround = 1;
 
 	switch (codec->vendor_id) {
 	case 0x14f15045:
-		codec->single_adc_amp = 1;
+		spec->single_adc_amp = 1;
 		break;
 	case 0x14f15051:
 		add_cx5051_fake_mutes(codec);
-		codec->pin_amp_workaround = 1;
-		apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl);
 		break;
-	default:
-		codec->pin_amp_workaround = 1;
-		apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl);
 	}
 
+	apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
+
 	/* Show mute-led control only on HP laptops
 	 * This is a sort of white-list: on HP laptops, EAPD corresponds
 	 * only to the mute-LED without actualy amp function.  Meanwhile,
diff --git a/trunk/sound/pci/hda/patch_hdmi.c b/trunk/sound/pci/hda/patch_hdmi.c
index 83f345f3c961..540cd13f7f15 100644
--- a/trunk/sound/pci/hda/patch_hdmi.c
+++ b/trunk/sound/pci/hda/patch_hdmi.c
@@ -757,6 +757,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	struct hdmi_spec *spec = codec->spec;
 	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
 	int pin_nid;
+	int pd = !!(res & AC_UNSOL_RES_PD);
+	int eldv = !!(res & AC_UNSOL_RES_ELDV);
 	int pin_idx;
 	struct hda_jack_tbl *jack;
 
@@ -766,10 +768,9 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	pin_nid = jack->nid;
 	jack->jack_dirty = 1;
 
-	_snd_printd(SND_PR_VERBOSE,
+	printk(KERN_INFO
 		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid,
-		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
+		codec->addr, pin_nid, pd, eldv);
 
 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
 	if (pin_idx < 0)
@@ -991,7 +992,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 	if (eld->monitor_present)
 		eld_valid	= !!(present & AC_PINSENSE_ELDV);
 
-	_snd_printd(SND_PR_VERBOSE,
+	printk(KERN_INFO
 		"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
 		codec->addr, pin_nid, eld->monitor_present, eld_valid);
 
diff --git a/trunk/sound/pci/hda/patch_realtek.c b/trunk/sound/pci/hda/patch_realtek.c
index 818f90bc7d57..9917e55d6f11 100644
--- a/trunk/sound/pci/hda/patch_realtek.c
+++ b/trunk/sound/pci/hda/patch_realtek.c
@@ -1445,13 +1445,6 @@ enum {
 	ALC_FIXUP_ACT_BUILD,
 };
 
-static void alc_apply_pincfgs(struct hda_codec *codec,
-			      const struct alc_pincfg *cfg)
-{
-	for (; cfg->nid; cfg++)
-		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-}
-
 static void alc_apply_fixup(struct hda_codec *codec, int action)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1485,7 +1478,9 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
 			snd_printdd(KERN_INFO "hda_codec: %s: "
 				    "Apply pincfg for %s\n",
 				    codec->chip_name, modelname);
-			alc_apply_pincfgs(codec, cfg);
+			for (; cfg->nid; cfg++)
+				snd_hda_codec_set_pincfg(codec, cfg->nid,
+							 cfg->val);
 			break;
 		case ALC_FIXUP_VERBS:
 			if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs)
@@ -3403,10 +3398,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 	for (;;) {
 		badness = fill_and_eval_dacs(codec, fill_hardwired,
 					     fill_mio_first);
-		if (badness < 0) {
-			kfree(best_cfg);
+		if (badness < 0)
 			return badness;
-		}
 		debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n",
 			      cfg->line_out_type, fill_hardwired, fill_mio_first,
 			      badness);
@@ -3441,7 +3434,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 			cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
 			fill_hardwired = true;
 			continue;
-		}
+		} 
 		if (cfg->hp_outs > 0 &&
 		    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
 			cfg->speaker_outs = cfg->line_outs;
@@ -3455,7 +3448,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 			cfg->line_out_type = AUTO_PIN_HP_OUT;
 			fill_hardwired = true;
 			continue;
-		}
+		} 
 		break;
 	}
 
@@ -4430,7 +4423,7 @@ static int alc_parse_auto_config(struct hda_codec *codec,
 static int alc880_parse_auto_config(struct hda_codec *codec)
 {
 	static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
-	static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+	static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; 
 	return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids);
 }
 
@@ -4866,7 +4859,6 @@ enum {
 	ALC260_FIXUP_GPIO1_TOGGLE,
 	ALC260_FIXUP_REPLACER,
 	ALC260_FIXUP_HP_B1900,
-	ALC260_FIXUP_KN1,
 };
 
 static void alc260_gpio1_automute(struct hda_codec *codec)
@@ -4894,36 +4886,6 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
 	}
 }
 
-static void alc260_fixup_kn1(struct hda_codec *codec,
-			     const struct alc_fixup *fix, int action)
-{
-	struct alc_spec *spec = codec->spec;
-	static const struct alc_pincfg pincfgs[] = {
-		{ 0x0f, 0x02214000 }, /* HP/speaker */
-		{ 0x12, 0x90a60160 }, /* int mic */
-		{ 0x13, 0x02a19000 }, /* ext mic */
-		{ 0x18, 0x01446000 }, /* SPDIF out */
-		/* disable bogus I/O pins */
-		{ 0x10, 0x411111f0 },
-		{ 0x11, 0x411111f0 },
-		{ 0x14, 0x411111f0 },
-		{ 0x15, 0x411111f0 },
-		{ 0x16, 0x411111f0 },
-		{ 0x17, 0x411111f0 },
-		{ 0x19, 0x411111f0 },
-		{ }
-	};
-
-	switch (action) {
-	case ALC_FIXUP_ACT_PRE_PROBE:
-		alc_apply_pincfgs(codec, pincfgs);
-		break;
-	case ALC_FIXUP_ACT_PROBE:
-		spec->init_amp = ALC_INIT_NONE;
-		break;
-	}
-}
-
 static const struct alc_fixup alc260_fixups[] = {
 	[ALC260_FIXUP_HP_DC5750] = {
 		.type = ALC_FIXUP_PINS,
@@ -4974,11 +4936,7 @@ static const struct alc_fixup alc260_fixups[] = {
 		.v.func = alc260_fixup_gpio1_toggle,
 		.chained = true,
 		.chain_id = ALC260_FIXUP_COEF,
-	},
-	[ALC260_FIXUP_KN1] = {
-		.type = ALC_FIXUP_FUNC,
-		.v.func = alc260_fixup_kn1,
-	},
+	}
 };
 
 static const struct snd_pci_quirk alc260_fixup_tbl[] = {
@@ -4988,7 +4946,6 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750),
 	SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900),
 	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1),
-	SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1),
 	SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER),
 	SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF),
 	{}
@@ -5312,9 +5269,7 @@ static const struct alc_fixup alc882_fixups[] = {
 			{ 0x16, 0x99130111 }, /* CLFE speaker */
 			{ 0x17, 0x99130112 }, /* surround speaker */
 			{ }
-		},
-		.chained = true,
-		.chain_id = ALC882_FIXUP_GPIO1,
+		}
 	},
 	[ALC882_FIXUP_ACER_ASPIRE_8930G] = {
 		.type = ALC_FIXUP_PINS,
@@ -5357,9 +5312,7 @@ static const struct alc_fixup alc882_fixups[] = {
 			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
 			{ 0x20, AC_VERB_SET_PROC_COEF, 0x3050 },
 			{ }
-		},
-		.chained = true,
-		.chain_id = ALC882_FIXUP_GPIO1,
+		}
 	},
 	[ALC885_FIXUP_MACPRO_GPIO] = {
 		.type = ALC_FIXUP_FUNC,
@@ -5406,7 +5359,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
 	SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
-	SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G),
 	SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
 	SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
@@ -5432,7 +5384,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF),
 	SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF),
 	SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF),
-	SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO),
 	SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF),
 	SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF),
 	SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF),
@@ -5448,13 +5399,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	{}
 };
 
-static const struct alc_model_fixup alc882_fixup_models[] = {
-	{.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"},
-	{.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"},
-	{.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"},
-	{}
-};
-
 /*
  * BIOS auto configuration
  */
@@ -5495,8 +5439,7 @@ static int patch_alc882(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
-		       alc882_fixups);
+	alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
 	alc_auto_parse_customize_define(codec);
@@ -6109,7 +6052,6 @@ static const struct alc_fixup alc269_fixups[] = {
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
-	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
@@ -6137,7 +6079,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	 * Basically the device should work as is without the fixup table.
 	 * If BIOS doesn't give a proper info, enable the corresponding
 	 * fixup entry.
-	 */
+	 */ 
 	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
 		      ALC269_FIXUP_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC),
@@ -6354,7 +6296,7 @@ static void alc_fixup_no_jack_detect(struct hda_codec *codec,
 {
 	if (action == ALC_FIXUP_ACT_PRE_PROBE)
 		codec->no_jack_detect = 1;
-}
+}	
 
 static const struct alc_fixup alc861_fixups[] = {
 	[ALC861_FIXUP_FSC_AMILO_PI1505] = {
@@ -6772,7 +6714,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	 * Basically the device should work as is without the fixup table.
 	 * If BIOS doesn't give a proper info, enable the corresponding
 	 * fixup entry.
-	 */
+	 */ 
 	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3),
 	SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1),
diff --git a/trunk/sound/pci/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c
index 4742cac26aa9..33a9946b492c 100644
--- a/trunk/sound/pci/hda/patch_sigmatel.c
+++ b/trunk/sound/pci/hda/patch_sigmatel.c
@@ -5063,11 +5063,12 @@ static void stac92xx_update_led_status(struct hda_codec *codec, int enabled)
 	if (spec->gpio_led_polarity)
 		muted = !muted;
 
+	/*polarity defines *not* muted state level*/
 	if (!spec->vref_mute_led_nid) {
 		if (muted)
-			spec->gpio_data |= spec->gpio_led;
+			spec->gpio_data &= ~spec->gpio_led; /* orange */
 		else
-			spec->gpio_data &= ~spec->gpio_led;
+			spec->gpio_data |= spec->gpio_led; /* white */
 		stac_gpio_set(codec, spec->gpio_mask,
 				spec->gpio_dir, spec->gpio_data);
 	} else {
diff --git a/trunk/sound/soc/blackfin/bf5xx-ssm2602.c b/trunk/sound/soc/blackfin/bf5xx-ssm2602.c
index b39ad356b92b..df3ac73f8778 100644
--- a/trunk/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/trunk/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -99,7 +99,6 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
 		.platform_name = "bfin-i2s-pcm-audio",
 		.codec_name = "ssm2602.0-001b",
 		.ops = &bf5xx_ssm2602_ops,
-		.dai_fmt = BF5XX_SSM2602_DAIFMT,
 	},
 	{
 		.name = "ssm2602",
@@ -109,7 +108,6 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
 		.platform_name = "bfin-i2s-pcm-audio",
 		.codec_name = "ssm2602.0-001b",
 		.ops = &bf5xx_ssm2602_ops,
-		.dai_fmt = BF5XX_SSM2602_DAIFMT,
 	},
 };
 
diff --git a/trunk/sound/soc/codecs/tlv320aic23.c b/trunk/sound/soc/codecs/tlv320aic23.c
index df1e07ffac32..16d55f91a653 100644
--- a/trunk/sound/soc/codecs/tlv320aic23.c
+++ b/trunk/sound/soc/codecs/tlv320aic23.c
@@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 				      enum snd_soc_bias_level level)
 {
-	u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f;
+	u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
@@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_OFF:
 		/* everything off, dac mute, inactive */
 		snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
-		snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
+		snd_soc_write(codec, TLV320AIC23_PWR, 0xffff);
 		break;
 	}
 	codec->dapm.bias_level = level;
diff --git a/trunk/sound/soc/codecs/wm8350.c b/trunk/sound/soc/codecs/wm8350.c
index aa12c6b6beeb..8c4c9591ec05 100644
--- a/trunk/sound/soc/codecs/wm8350.c
+++ b/trunk/sound/soc/codecs/wm8350.c
@@ -60,7 +60,7 @@ struct wm8350_jack_data {
 };
 
 struct wm8350_data {
-	struct wm8350 *wm8350;
+	struct snd_soc_codec codec;
 	struct wm8350_output out1;
 	struct wm8350_output out2;
 	struct wm8350_jack_data hpl;
@@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv,
 			   struct wm8350_jack_data *jack,
 			   u16 mask)
 {
-	struct wm8350 *wm8350 = priv->wm8350;
+	struct wm8350 *wm8350 = priv->codec.control_data;
 	u16 reg;
 	int report;
 
@@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work)
 static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
 {
 	struct wm8350_data *priv = data;
-	struct wm8350 *wm8350 = priv->wm8350;
+	struct wm8350 *wm8350 = priv->codec.control_data;
 	struct wm8350_jack_data *jack = NULL;
 
 	switch (irq - wm8350->irq_base) {
@@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
 static irqreturn_t wm8350_mic_handler(int irq, void *data)
 {
 	struct wm8350_data *priv = data;
-	struct wm8350 *wm8350 = priv->wm8350;
+	struct wm8350 *wm8350 = priv->codec.control_data;
 	u16 reg;
 	int report = 0;
 
@@ -1536,8 +1536,6 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 		return -ENOMEM;
 	snd_soc_codec_set_drvdata(codec, priv);
 
-	priv->wm8350 = wm8350;
-
 	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
 		priv->supplies[i].supply = supply_names[i];
 
@@ -1546,6 +1544,7 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;
 
+	wm8350->codec.codec = codec;
 	codec->control_data = wm8350;
 
 	/* Put the codec into reset if it wasn't already */
diff --git a/trunk/sound/soc/codecs/wm_hubs.c b/trunk/sound/soc/codecs/wm_hubs.c
index 6c028c470601..f13f2886339c 100644
--- a/trunk/sound/soc/codecs/wm_hubs.c
+++ b/trunk/sound/soc/codecs/wm_hubs.c
@@ -1035,7 +1035,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 			    enum snd_soc_bias_level level)
 {
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
-	int mask, val;
+	int val;
 
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
@@ -1047,13 +1047,6 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_ON:
 		/* Turn off any unneded single ended outputs */
 		val = 0;
-		mask = 0;
-
-		if (hubs->lineout1_se)
-			mask |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA;
-
-		if (hubs->lineout2_se)
-			mask |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA;
 
 		if (hubs->lineout1_se && hubs->lineout1n_ena)
 			val |= WM8993_LINEOUT1N_ENA;
@@ -1068,7 +1061,11 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 			val |= WM8993_LINEOUT2P_ENA;
 
 		snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3,
-				    mask, val);
+				    WM8993_LINEOUT1N_ENA |
+				    WM8993_LINEOUT1P_ENA |
+				    WM8993_LINEOUT2N_ENA |
+				    WM8993_LINEOUT2P_ENA,
+				    val);
 
 		/* Remove the input clamps */
 		snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG,
diff --git a/trunk/sound/soc/omap/omap-pcm.c b/trunk/sound/soc/omap/omap-pcm.c
index a59bd352d342..5a649da9122a 100644
--- a/trunk/sound/soc/omap/omap-pcm.c
+++ b/trunk/sound/soc/omap/omap-pcm.c
@@ -401,6 +401,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	}
 
 out:
+	/* free preallocated buffers in case of error */
+	if (ret)
+		omap_pcm_free_dma_buffers(pcm);
+
 	return ret;
 }
 
diff --git a/trunk/sound/soc/samsung/s3c2412-i2s.c b/trunk/sound/soc/samsung/s3c2412-i2s.c
index 79fbeea99d46..72185078ddf8 100644
--- a/trunk/sound/soc/samsung/s3c2412-i2s.c
+++ b/trunk/sound/soc/samsung/s3c2412-i2s.c
@@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = {
 
 static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev)
 {
-	return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai);
+	return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai);
 }
 
 static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev)
diff --git a/trunk/sound/soc/soc-core.c b/trunk/sound/soc/soc-core.c
index b390f00b4e99..3a4e93e52b6d 100644
--- a/trunk/sound/soc/soc-core.c
+++ b/trunk/sound/soc/soc-core.c
@@ -3631,10 +3631,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 	int i, ret;
 
 	num_routes = of_property_count_strings(np, propname);
-	if (num_routes < 0 || num_routes & 1) {
+	if (num_routes & 1) {
 		dev_err(card->dev,
-		     "Property '%s' does not exist or its length is not even\n",
-		     propname);
+			"Property '%s's length is not even\n",
+			propname);
 		return -EINVAL;
 	}
 	num_routes /= 2;