Merge tag 'regmap-3.4' into regmap-stride

regmap: Last minute bug fix for 3.4

This is a last minute bug fix that was only just noticed since the code
path that's being exercised here is one that is fairly rarely used.  The
changelog for the change itself is extremely clear and the code itself
is obvious to inspection so should be pretty safe.

Conflicts:
	drivers/base/regmap/regmap.c (overlap between the fix and stride code)

34 files changed, 476 insertions, 209 deletions

diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 14a286a7bf2b..857586135d18 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -419,6 +419,7 @@ 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/sound/isa/sscape.c b/sound/isa/sscape.c
index b4a6aa960f4b..8490f59709bb 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -1019,13 +1019,15 @@ 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]);
-		return -ENXIO;
+		err = -ENXIO;
+		goto _release_dma;
 	}
 
 	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]);
-		return -ENXIO;
+		err = -ENXIO;
+		goto _release_dma;
 	}
 
 	/*
diff --git a/sound/last.c b/sound/last.c
index bdd0857b8871..7ffc182e0844 100644
--- a/sound/last.c
+++ b/sound/last.c
@@ -38,4 +38,4 @@ static int __init alsa_sound_last_init(void)
 	return 0;
 }
 
-__initcall(alsa_sound_last_init);
+late_initcall_sync(alsa_sound_last_init);
diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c
index 2c79d60a725f..536c4c0514d3 100644
--- a/sound/oss/msnd_pinnacle.c
+++ b/sound/oss/msnd_pinnacle.c
@@ -1294,6 +1294,8 @@ 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);
@@ -1312,7 +1314,8 @@ 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");
-		return -ENODEV;
+		ret = -ENODEV;
+		goto out;
 	}
 #ifdef HAVE_DSPCODEH
 	printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n");
@@ -1320,12 +1323,13 @@ 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 0;
+	return ret;
 }
 
 #ifdef MSND_CLASSIC
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 88168044375f..5ca0939e4223 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -2,8 +2,8 @@
 
 config SND_TEA575X
 	tristate
-	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
-	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 || RADIO_MAXIRADIO
+	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 || RADIO_MAXIRADIO
 
 menuconfig SND_PCI
 	bool "PCI sound devices"
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index 8c63200cf339..bc86cb726d79 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2012  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.
 */
-int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
+u16 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/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c
index 87f4385fe8c7..5ef4fe964366 100644
--- a/sound/pci/asihpi/hpios.c
+++ b/sound/pci/asihpi/hpios.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2012  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)
 
 }
 
-/** Allocated an area of locked memory for bus master DMA operations.
+/** Allocate an area of locked memory for bus master DMA operations.
 
-On error, return -ENOMEM, and *pMemArea.size = 0
+If allocation fails, return 1, and *pMemArea.size = 0
 */
-int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
+u16 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 @@ int 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 -ENOMEM;
+		return 1;
 	}
 }
 
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 9a9f372e1be4..56b4f74c0b13 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -851,6 +851,9 @@ 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/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index b58b4b1687fa..4c054f4486b9 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/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';
 
-	printk(KERN_INFO "HDMI: supports coding type %s:"
+	_snd_printd(SND_PR_VERBOSE, "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;
 
-	printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n",
+	_snd_printd(SND_PR_VERBOSE, "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));
-		printk(KERN_INFO "HDMI: available speakers:%s\n", buf);
+		_snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf);
 	}
 
 	for (i = 0; i < e->sad_count; i++)
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 254ab5204603..e59e2f059b6e 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -651,9 +651,16 @@ 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: ");
-			print_amp_vals(buffer, codec, nid, HDA_INPUT,
-				       wid_caps & AC_WCAP_STEREO,
-				       wid_type == AC_WID_PIN ? 1 : conn_len);
+			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);
 		}
 		if (wid_caps & AC_WCAP_OUT_AMP) {
 			snd_iprintf(buffer, "  Amp-Out caps: ");
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 8c6523bbc797..d906c5b74cf0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -141,7 +141,6 @@ 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;
 
@@ -687,27 +686,26 @@ static const struct hda_channel_mode cxt5045_modes[1] = {
 static const struct hda_input_mux cxt5045_capture_source = {
 	.num_items = 2,
 	.items = {
-		{ "IntMic", 0x1 },
-		{ "ExtMic", 0x2 },
+		{ "Internal Mic", 0x1 },
+		{ "Mic",          0x2 },
 	}
 };
 
 static const struct hda_input_mux cxt5045_capture_source_benq = {
-	.num_items = 5,
+	.num_items = 4,
 	.items = {
-		{ "IntMic", 0x1 },
-		{ "ExtMic", 0x2 },
-		{ "LineIn", 0x3 },
-		{ "CD",     0x4 },
-		{ "Mixer",  0x0 },
+		{ "Internal Mic", 0x1 },
+		{ "Mic",          0x2 },
+		{ "Line",         0x3 },
+		{ "Mixer",        0x0 },
 	}
 };
 
 static const struct hda_input_mux cxt5045_capture_source_hp530 = {
 	.num_items = 2,
 	.items = {
-		{ "ExtMic", 0x1 },
-		{ "IntMic", 0x2 },
+		{ "Mic",          0x1 },
+		{ "Internal Mic", 0x2 },
 	}
 };
 
@@ -798,10 +796,8 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec,
 }
 
 static const struct snd_kcontrol_new cxt5045_mixers[] = {
-	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("Capture Volume", 0x1a, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, 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),
@@ -822,27 +818,15 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = {
 };
 
 static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
-	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),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT),
 
 	{}
 };
 
 static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
-	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("Capture Volume", 0x1a, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, 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),
@@ -946,10 +930,10 @@ 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("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),
+	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),
 	
 	/* Modes for retasking pin widgets */
 	CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT),
@@ -960,16 +944,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 
 	/* Loopback mixer controls */
 
-	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),
+	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),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Input Source",
@@ -978,16 +962,8 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 		.put = conexant_mux_enum_put,
 	},
 	/* Audio input controls */
-	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),
+	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -1009,10 +985,6 @@ 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
@@ -1027,11 +999,11 @@ 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, 0x00},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x01},
 
 	/* Mute all inputs to mixer widget (even unconnected ones) */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */
 	{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 */
@@ -1110,7 +1082,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
-	codec->pin_amp_workaround = 1;
+	codec->single_adc_amp = 1;
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
@@ -3999,9 +3971,14 @@ 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++)
+	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;
 		snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+	}
 	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);
@@ -4220,7 +4197,7 @@ 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 (spec->single_adc_amp)
+		if (codec->single_adc_amp)
 			idx = 0;
 		return cx_auto_add_volume_idx(codec, label, pfx,
 					      cidx, adc_nid, HDA_INPUT, idx);
@@ -4275,7 +4252,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
 		if (cidx < 0)
 			continue;
 		input_conn[i] = spec->imux_info[i].adc;
-		if (!spec->single_adc_amp)
+		if (!codec->single_adc_amp)
 			input_conn[i] |= cidx << 8;
 		if (i > 0 && input_conn[i] != input_conn[0])
 			multi_connection = 1;
@@ -4419,8 +4396,10 @@ 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 */
@@ -4429,15 +4408,33 @@ 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 cxt_fixups[] = {
+static const struct snd_pci_quirk cxt5051_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)
  */
@@ -4466,19 +4463,21 @@ 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:
-		spec->single_adc_amp = 1;
+		codec->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/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 540cd13f7f15..83f345f3c961 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -757,8 +757,6 @@ 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;
 
@@ -768,9 +766,10 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	pin_nid = jack->nid;
 	jack->jack_dirty = 1;
 
-	printk(KERN_INFO
+	_snd_printd(SND_PR_VERBOSE,
 		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid, pd, eldv);
+		codec->addr, pin_nid,
+		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
 
 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
 	if (pin_idx < 0)
@@ -992,7 +991,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 	if (eld->monitor_present)
 		eld_valid	= !!(present & AC_PINSENSE_ELDV);
 
-	printk(KERN_INFO
+	_snd_printd(SND_PR_VERBOSE,
 		"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/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9917e55d6f11..818f90bc7d57 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1445,6 +1445,13 @@ 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;
@@ -1478,9 +1485,7 @@ 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);
-			for (; cfg->nid; cfg++)
-				snd_hda_codec_set_pincfg(codec, cfg->nid,
-							 cfg->val);
+			alc_apply_pincfgs(codec, cfg);
 			break;
 		case ALC_FIXUP_VERBS:
 			if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs)
@@ -3398,8 +3403,10 @@ 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)
+		if (badness < 0) {
+			kfree(best_cfg);
 			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);
@@ -3434,7 +3441,7 @@ 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;
@@ -3448,7 +3455,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 			cfg->line_out_type = AUTO_PIN_HP_OUT;
 			fill_hardwired = true;
 			continue;
-		} 
+		}
 		break;
 	}
 
@@ -4423,7 +4430,7 @@ 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);
 }
 
@@ -4859,6 +4866,7 @@ enum {
 	ALC260_FIXUP_GPIO1_TOGGLE,
 	ALC260_FIXUP_REPLACER,
 	ALC260_FIXUP_HP_B1900,
+	ALC260_FIXUP_KN1,
 };
 
 static void alc260_gpio1_automute(struct hda_codec *codec)
@@ -4886,6 +4894,36 @@ 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,
@@ -4936,7 +4974,11 @@ 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[] = {
@@ -4946,6 +4988,7 @@ 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),
 	{}
@@ -5269,7 +5312,9 @@ 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,
@@ -5312,7 +5357,9 @@ 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,
@@ -5359,6 +5406,7 @@ 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),
@@ -5384,6 +5432,7 @@ 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),
@@ -5399,6 +5448,13 @@ 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
  */
@@ -5439,7 +5495,8 @@ static int patch_alc882(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
+	alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
+		       alc882_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
 	alc_auto_parse_customize_define(codec);
@@ -6052,6 +6109,7 @@ 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),
@@ -6079,7 +6137,7 @@ 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),
@@ -6296,7 +6354,7 @@ 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] = {
@@ -6714,7 +6772,7 @@ 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/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 33a9946b492c..4742cac26aa9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5063,12 +5063,11 @@ 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; /* orange */
+			spec->gpio_data |= spec->gpio_led;
 		else
-			spec->gpio_data |= spec->gpio_led; /* white */
+			spec->gpio_data &= ~spec->gpio_led;
 		stac_gpio_set(codec, spec->gpio_mask,
 				spec->gpio_dir, spec->gpio_data);
 	} else {
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index df3ac73f8778..b39ad356b92b 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -99,6 +99,7 @@ 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",
@@ -108,6 +109,7 @@ 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/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 6508e8b790bb..59d8efaa17e9 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -57,7 +57,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TPA6130A2 if I2C
 	select SND_SOC_TLV320DAC33 if I2C
 	select SND_SOC_TWL4030 if TWL4030_CORE
-	select SND_SOC_TWL6040 if TWL4030_CORE
+	select SND_SOC_TWL6040 if TWL6040_CORE
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
 	select SND_SOC_WL1273 if MFD_WL1273_CORE
@@ -276,7 +276,6 @@ config SND_SOC_TWL4030
 	tristate
 
 config SND_SOC_TWL6040
-	select TWL6040_CORE
 	tristate
 
 config SND_SOC_UDA134X
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index f8e10ced244a..b3e24f289421 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -140,7 +140,7 @@
  * min : 0xFE : -115.0 dB
  * mute: 0xFF
  */
-static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
 
 static const struct snd_kcontrol_new ak4642_snd_controls[] = {
 
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 78979b3e0e95..07c44b71f096 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -929,6 +929,8 @@ static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq)
 
 	/* MCLKX -> MCLK */
 	mclkx_coeff = cs42l73_get_mclkx_coeff(freq);
+	if (mclkx_coeff < 0)
+		return mclkx_coeff;
 
 	mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx /
 		cs42l73_mclkx_coeffs[mclkx_coeff].ratio;
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d1926266fe00..8e92fb88ed09 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 }
 
 /*
- * using codec assist to small pop, hp_powerup or lineout_powerup
- * should stay setting until vag_powerup is fully ramped down,
- * vag fully ramped down require 400ms.
+ * As manual described, ADC/DAC only works when VAG powerup,
+ * So enabled VAG before ADC/DAC up.
+ * In power down case, we need wait 400ms when vag fully ramped down.
  */
-static int small_pop_event(struct snd_soc_dapm_widget *w,
+static int power_vag_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
 	switch (event) {
@@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w,
 			SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
 		break;
 
-	case SND_SOC_DAPM_PRE_PMD:
+	case SND_SOC_DAPM_POST_PMD:
 		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
 			SGTL5000_VAG_POWERUP, 0);
 		msleep(400);
@@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 				mic_bias_event,
 				SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
-	SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
-			small_pop_event,
-			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
-	SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0,
-			small_pop_event,
-			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
 
 	SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
 	SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux),
@@ -221,8 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 				0, SGTL5000_CHIP_DIG_POWER,
 				1, 0),
 
-	SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
+	SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0,
+			    power_vag_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 
+	SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
 	SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
 };
 
@@ -231,9 +230,11 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = {
 	{"Capture Mux", "LINE_IN", "LINE_IN"},	/* line_in --> adc_mux */
 	{"Capture Mux", "MIC_IN", "MIC_IN"},	/* mic_in --> adc_mux */
 
+	{"ADC", NULL, "VAG_POWER"},
 	{"ADC", NULL, "Capture Mux"},		/* adc_mux --> adc */
 	{"AIFOUT", NULL, "ADC"},		/* adc --> i2s_out */
 
+	{"DAC", NULL, "VAG_POWER"},
 	{"DAC", NULL, "AIFIN"},			/* i2s-->dac,skip audio mux */
 	{"Headphone Mux", "DAC", "DAC"},	/* dac --> hp_mux */
 	{"LO", NULL, "DAC"},			/* dac --> line_out */
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 16d55f91a653..df1e07ffac32 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/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) & 0xff7f;
+	u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f;
 
 	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, 0xffff);
+		snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
 		break;
 	}
 	codec->dapm.bias_level = level;
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 2d8c6b825e57..dc7509b9d53a 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -26,7 +26,6 @@
 #include <linux/pm.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
-#include <linux/i2c/twl.h>
 #include <linux/mfd/twl6040.h>
 
 #include <sound/core.h>
@@ -1528,7 +1527,7 @@ static int twl6040_resume(struct snd_soc_codec *codec)
 static int twl6040_probe(struct snd_soc_codec *codec)
 {
 	struct twl6040_data *priv;
-	struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev);
+	struct twl6040_codec_data *pdata = dev_get_platdata(codec->dev);
 	struct platform_device *pdev = container_of(codec->dev,
 						   struct platform_device, dev);
 	int ret = 0;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 8c4c9591ec05..aa12c6b6beeb 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -60,7 +60,7 @@ struct wm8350_jack_data {
 };
 
 struct wm8350_data {
-	struct snd_soc_codec codec;
+	struct wm8350 *wm8350;
 	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->codec.control_data;
+	struct wm8350 *wm8350 = priv->wm8350;
 	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->codec.control_data;
+	struct wm8350 *wm8350 = priv->wm8350;
 	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->codec.control_data;
+	struct wm8350 *wm8350 = priv->wm8350;
 	u16 reg;
 	int report = 0;
 
@@ -1536,6 +1536,8 @@ 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];
 
@@ -1544,7 +1546,6 @@ 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/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 7c49642af052..6c1fe3afd4b5 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1000,61 +1000,170 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
 	}
 }
 
-static int late_enable_ev(struct snd_soc_dapm_widget *w,
-			  struct snd_kcontrol *kcontrol, int event)
+static int aif1clk_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+	struct wm8994 *control = codec->control_data;
+	int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
+	int dac;
+	int adc;
+	int val;
+
+	switch (control->type) {
+	case WM8994:
+	case WM8958:
+		mask |= WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA;
+		break;
+	default:
+		break;
+	}
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		if (wm8994->aif1clk_enable) {
-			snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
-					    WM8994_AIF1CLK_ENA_MASK,
-					    WM8994_AIF1CLK_ENA);
-			wm8994->aif1clk_enable = 0;
-		}
-		if (wm8994->aif2clk_enable) {
-			snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
-					    WM8994_AIF2CLK_ENA_MASK,
-					    WM8994_AIF2CLK_ENA);
-			wm8994->aif2clk_enable = 0;
-		}
+		val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1);
+		if ((val & WM8994_AIF1ADCL_SRC) &&
+		    (val & WM8994_AIF1ADCR_SRC))
+			adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA;
+		else if (!(val & WM8994_AIF1ADCL_SRC) &&
+			 !(val & WM8994_AIF1ADCR_SRC))
+			adc = WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA;
+		else
+			adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA |
+				WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA;
+
+		val = snd_soc_read(codec, WM8994_AIF1_CONTROL_2);
+		if ((val & WM8994_AIF1DACL_SRC) &&
+		    (val & WM8994_AIF1DACR_SRC))
+			dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA;
+		else if (!(val & WM8994_AIF1DACL_SRC) &&
+			 !(val & WM8994_AIF1DACR_SRC))
+			dac = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA;
+		else
+			dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA |
+				WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA;
+
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
+				    mask, adc);
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+				    mask, dac);
+		snd_soc_update_bits(codec, WM8994_CLOCKING_1,
+				    WM8994_AIF1DSPCLK_ENA |
+				    WM8994_SYSDSPCLK_ENA,
+				    WM8994_AIF1DSPCLK_ENA |
+				    WM8994_SYSDSPCLK_ENA);
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, mask,
+				    WM8994_AIF1ADC1R_ENA |
+				    WM8994_AIF1ADC1L_ENA |
+				    WM8994_AIF1ADC2R_ENA |
+				    WM8994_AIF1ADC2L_ENA);
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, mask,
+				    WM8994_AIF1DAC1R_ENA |
+				    WM8994_AIF1DAC1L_ENA |
+				    WM8994_AIF1DAC2R_ENA |
+				    WM8994_AIF1DAC2L_ENA);
 		break;
-	}
 
-	/* We may also have postponed startup of DSP, handle that. */
-	wm8958_aif_ev(w, kcontrol, event);
+	case SND_SOC_DAPM_PRE_PMD:
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+				    mask, 0);
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
+				    mask, 0);
+
+		val = snd_soc_read(codec, WM8994_CLOCKING_1);
+		if (val & WM8994_AIF2DSPCLK_ENA)
+			val = WM8994_SYSDSPCLK_ENA;
+		else
+			val = 0;
+		snd_soc_update_bits(codec, WM8994_CLOCKING_1,
+				    WM8994_SYSDSPCLK_ENA |
+				    WM8994_AIF1DSPCLK_ENA, val);
+		break;
+	}
 
 	return 0;
 }
 
-static int late_disable_ev(struct snd_soc_dapm_widget *w,
-			   struct snd_kcontrol *kcontrol, int event)
+static int aif2clk_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+	int dac;
+	int adc;
+	int val;
 
 	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		val = snd_soc_read(codec, WM8994_AIF2_CONTROL_1);
+		if ((val & WM8994_AIF2ADCL_SRC) &&
+		    (val & WM8994_AIF2ADCR_SRC))
+			adc = WM8994_AIF2ADCR_ENA;
+		else if (!(val & WM8994_AIF2ADCL_SRC) &&
+			 !(val & WM8994_AIF2ADCR_SRC))
+			adc = WM8994_AIF2ADCL_ENA;
+		else
+			adc = WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA;
+
+
+		val = snd_soc_read(codec, WM8994_AIF2_CONTROL_2);
+		if ((val & WM8994_AIF2DACL_SRC) &&
+		    (val & WM8994_AIF2DACR_SRC))
+			dac = WM8994_AIF2DACR_ENA;
+		else if (!(val & WM8994_AIF2DACL_SRC) &&
+			 !(val & WM8994_AIF2DACR_SRC))
+			dac = WM8994_AIF2DACL_ENA;
+		else
+			dac = WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA;
+
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
+				    WM8994_AIF2ADCL_ENA |
+				    WM8994_AIF2ADCR_ENA, adc);
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+				    WM8994_AIF2DACL_ENA |
+				    WM8994_AIF2DACR_ENA, dac);
+		snd_soc_update_bits(codec, WM8994_CLOCKING_1,
+				    WM8994_AIF2DSPCLK_ENA |
+				    WM8994_SYSDSPCLK_ENA,
+				    WM8994_AIF2DSPCLK_ENA |
+				    WM8994_SYSDSPCLK_ENA);
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
+				    WM8994_AIF2ADCL_ENA |
+				    WM8994_AIF2ADCR_ENA,
+				    WM8994_AIF2ADCL_ENA |
+				    WM8994_AIF2ADCR_ENA);
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+				    WM8994_AIF2DACL_ENA |
+				    WM8994_AIF2DACR_ENA,
+				    WM8994_AIF2DACL_ENA |
+				    WM8994_AIF2DACR_ENA);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
 	case SND_SOC_DAPM_POST_PMD:
-		if (wm8994->aif1clk_disable) {
-			snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
-					    WM8994_AIF1CLK_ENA_MASK, 0);
-			wm8994->aif1clk_disable = 0;
-		}
-		if (wm8994->aif2clk_disable) {
-			snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
-					    WM8994_AIF2CLK_ENA_MASK, 0);
-			wm8994->aif2clk_disable = 0;
-		}
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+				    WM8994_AIF2DACL_ENA |
+				    WM8994_AIF2DACR_ENA, 0);
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+				    WM8994_AIF2ADCL_ENA |
+				    WM8994_AIF2ADCR_ENA, 0);
+
+		val = snd_soc_read(codec, WM8994_CLOCKING_1);
+		if (val & WM8994_AIF1DSPCLK_ENA)
+			val = WM8994_SYSDSPCLK_ENA;
+		else
+			val = 0;
+		snd_soc_update_bits(codec, WM8994_CLOCKING_1,
+				    WM8994_SYSDSPCLK_ENA |
+				    WM8994_AIF2DSPCLK_ENA, val);
 		break;
 	}
 
 	return 0;
 }
 
-static int aif1clk_ev(struct snd_soc_dapm_widget *w,
-		      struct snd_kcontrol *kcontrol, int event)
+static int aif1clk_late_ev(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
@@ -1071,8 +1180,8 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int aif2clk_ev(struct snd_soc_dapm_widget *w,
-		      struct snd_kcontrol *kcontrol, int event)
+static int aif2clk_late_ev(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
@@ -1089,6 +1198,63 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int late_enable_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (wm8994->aif1clk_enable) {
+			aif1clk_ev(w, kcontrol, event);
+			snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+					    WM8994_AIF1CLK_ENA_MASK,
+					    WM8994_AIF1CLK_ENA);
+			wm8994->aif1clk_enable = 0;
+		}
+		if (wm8994->aif2clk_enable) {
+			aif2clk_ev(w, kcontrol, event);
+			snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+					    WM8994_AIF2CLK_ENA_MASK,
+					    WM8994_AIF2CLK_ENA);
+			wm8994->aif2clk_enable = 0;
+		}
+		break;
+	}
+
+	/* We may also have postponed startup of DSP, handle that. */
+	wm8958_aif_ev(w, kcontrol, event);
+
+	return 0;
+}
+
+static int late_disable_ev(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMD:
+		if (wm8994->aif1clk_disable) {
+			snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+					    WM8994_AIF1CLK_ENA_MASK, 0);
+			aif1clk_ev(w, kcontrol, event);
+			wm8994->aif1clk_disable = 0;
+		}
+		if (wm8994->aif2clk_disable) {
+			snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+					    WM8994_AIF2CLK_ENA_MASK, 0);
+			aif2clk_ev(w, kcontrol, event);
+			wm8994->aif2clk_disable = 0;
+		}
+		break;
+	}
+
+	return 0;
+}
+
 static int adc_mux_ev(struct snd_soc_dapm_widget *w,
 		      struct snd_kcontrol *kcontrol, int event)
 {
@@ -1385,9 +1551,9 @@ static const struct snd_kcontrol_new aif2dacr_src_mux =
 	SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
 
 static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = {
-SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev,
+SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_late_ev,
 	SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
-SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev,
+SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_late_ev,
 	SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 
 SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
@@ -1416,8 +1582,10 @@ SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
 };
 
 static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = {
-SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
 		   left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
@@ -1470,30 +1638,30 @@ SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event,
 SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
 		    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
-SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("DSPINTCLK", SND_SOC_NOPM, 1, 0, NULL, 0),
 
 SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
-		     0, WM8994_POWER_MANAGEMENT_4, 9, 0),
+		     0, SND_SOC_NOPM, 9, 0),
 SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
-		     0, WM8994_POWER_MANAGEMENT_4, 8, 0),
+		     0, SND_SOC_NOPM, 8, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
-		      WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
+		      SND_SOC_NOPM, 9, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
-		      WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
+		      SND_SOC_NOPM, 8, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
-		     0, WM8994_POWER_MANAGEMENT_4, 11, 0),
+		     0, SND_SOC_NOPM, 11, 0),
 SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
-		     0, WM8994_POWER_MANAGEMENT_4, 10, 0),
+		     0, SND_SOC_NOPM, 10, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
-		      WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
+		      SND_SOC_NOPM, 11, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0,
-		      WM8994_POWER_MANAGEMENT_5, 10, 0, wm8958_aif_ev,
+		      SND_SOC_NOPM, 10, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0,
@@ -1520,14 +1688,14 @@ SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0,
 		   dac1r_mix, ARRAY_SIZE(dac1r_mix)),
 
 SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0,
-		     WM8994_POWER_MANAGEMENT_4, 13, 0),
+		     SND_SOC_NOPM, 13, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0,
-		     WM8994_POWER_MANAGEMENT_4, 12, 0),
+		     SND_SOC_NOPM, 12, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF2DACL", NULL, 0,
-		      WM8994_POWER_MANAGEMENT_5, 13, 0, wm8958_aif_ev,
+		      SND_SOC_NOPM, 13, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
-		      WM8994_POWER_MANAGEMENT_5, 12, 0, wm8958_aif_ev,
+		      SND_SOC_NOPM, 12, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
 SND_SOC_DAPM_AIF_IN("AIF1DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0),
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index f13f2886339c..6c028c470601 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/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 val;
+	int mask, val;
 
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
@@ -1047,6 +1047,13 @@ 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;
@@ -1061,11 +1068,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 			val |= WM8993_LINEOUT2P_ENA;
 
 		snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3,
-				    WM8993_LINEOUT1N_ENA |
-				    WM8993_LINEOUT1P_ENA |
-				    WM8993_LINEOUT2N_ENA |
-				    WM8993_LINEOUT2P_ENA,
-				    val);
+				    mask, val);
 
 		/* Remove the input clamps */
 		snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG,
diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c
index 1765a197acb0..f23700359c67 100644
--- a/sound/soc/imx/imx-audmux.c
+++ b/sound/soc/imx/imx-audmux.c
@@ -73,6 +73,9 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
+	if (!audmux_base)
+		return -ENOSYS;
+
 	if (audmux_clk)
 		clk_prepare_enable(audmux_clk);
 
@@ -152,7 +155,7 @@ static void __init audmux_debugfs_init(void)
 		return;
 	}
 
-	for (i = 1; i < 8; i++) {
+	for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) {
 		snprintf(buf, sizeof(buf), "ssi%d", i);
 		if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
 					 (void *)i, &audmux_debugfs_fops))
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index e00dd0b1139c..deafbfaacdbf 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -97,7 +97,7 @@ config SND_OMAP_SOC_SDP3430
 
 config SND_OMAP_SOC_OMAP_ABE_TWL6040
 	tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
-	depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4
+	depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4
 	select SND_OMAP_SOC_DMIC
 	select SND_OMAP_SOC_MCPDM
 	select SND_SOC_TWL6040
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index a59bd352d342..5a649da9122a 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/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/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 609abd51e55f..d08583790d23 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -17,6 +17,7 @@
 #include <linux/delay.h>
 #include <linux/clk.h>
 #include <linux/platform_device.h>
+#include <linux/io.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 72185078ddf8..79fbeea99d46 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/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 snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai);
+	return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai);
 }
 
 static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev)
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 378cc5b056d7..74ed2dffbffd 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1001,11 +1001,10 @@ static void fsi_dma_do_tasklet(unsigned long data)
 	sg_dma_address(&sg) = buf;
 	sg_dma_len(&sg) = len;
 
-	desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir,
-						  DMA_PREP_INTERRUPT |
-						  DMA_CTRL_ACK);
+	desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir,
+				       DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
 	if (!desc) {
-		dev_err(dai->dev, "device_prep_slave_sg() fail\n");
+		dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
 		return;
 	}
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e19c24ade414..c88d9741b9e7 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1081,6 +1081,8 @@ static int soc_probe_platform(struct snd_soc_card *card,
 		snd_soc_dapm_new_controls(&platform->dapm,
 			driver->dapm_widgets, driver->num_dapm_widgets);
 
+	platform->dapm.idle_bias_off = 1;
+
 	if (driver->probe) {
 		ret = driver->probe(platform);
 		if (ret < 0) {
@@ -3111,6 +3113,7 @@ int snd_soc_register_card(struct snd_soc_card *card)
 				 GFP_KERNEL);
 	if (card->rtd == NULL)
 		return -ENOMEM;
+	card->num_rtd = 0;
 	card->rtd_aux = &card->rtd[card->num_links];
 
 	for (i = 0; i < card->num_links; i++)
@@ -3622,10 +3625,10 @@ 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 & 1) {
+	if (num_routes < 0 || num_routes & 1) {
 		dev_err(card->dev,
-			"Property '%s's length is not even\n",
-			propname);
+		     "Property '%s' does not exist or its length is not even\n",
+		     propname);
 		return -EINVAL;
 	}
 	num_routes /= 2;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 5cbd2d7623b8..1bb6d4a63cd8 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -67,6 +67,7 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_out_drv] = 10,
 	[snd_soc_dapm_hp] = 10,
 	[snd_soc_dapm_spk] = 10,
+	[snd_soc_dapm_line] = 10,
 	[snd_soc_dapm_post] = 11,
 };
 
@@ -75,6 +76,7 @@ static int dapm_down_seq[] = {
 	[snd_soc_dapm_adc] = 1,
 	[snd_soc_dapm_hp] = 2,
 	[snd_soc_dapm_spk] = 2,
+	[snd_soc_dapm_line] = 2,
 	[snd_soc_dapm_out_drv] = 2,
 	[snd_soc_dapm_pga] = 4,
 	[snd_soc_dapm_mixer_named_ctl] = 5,
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 33509de52540..e53349912b2e 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -79,11 +79,15 @@ static int tegra_i2s_show(struct seq_file *s, void *unused)
 	struct tegra_i2s *i2s = s->private;
 	int i;
 
+	clk_enable(i2s->clk_i2s);
+
 	for (i = 0; i < ARRAY_SIZE(regs); i++) {
 		u32 val = tegra_i2s_read(i2s, regs[i].offset);
 		seq_printf(s, "%s = %08x\n", regs[i].name, val);
 	}
 
+	clk_disable(i2s->clk_i2s);
+
 	return 0;
 }
 
@@ -112,7 +116,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
 		debugfs_remove(i2s->debug);
 }
 #else
-static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id)
+static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s)
 {
 }
 
diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c
index 475428cf270e..9ff2c601445f 100644
--- a/sound/soc/tegra/tegra_spdif.c
+++ b/sound/soc/tegra/tegra_spdif.c
@@ -79,11 +79,15 @@ static int tegra_spdif_show(struct seq_file *s, void *unused)
 	struct tegra_spdif *spdif = s->private;
 	int i;
 
+	clk_enable(spdif->clk_spdif_out);
+
 	for (i = 0; i < ARRAY_SIZE(regs); i++) {
 		u32 val = tegra_spdif_read(spdif, regs[i].offset);
 		seq_printf(s, "%s = %08x\n", regs[i].name, val);
 	}
 
+	clk_disable(spdif->clk_spdif_out);
+
 	return 0;
 }