ASoC: Merge tag 'v3.4-rc3' into for-3.5

Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.

Conflicts:
	sound/soc/soc-core.c
	sound/soc/tegra/tegra_i2s.c
	sound/soc/tegra/tegra_spdif.c

16 files changed, 127 insertions, 115 deletions

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/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..a36488d94aaa 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);
@@ -4220,7 +4192,7 @@ 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 +4247,7 @@ 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;
@@ -4466,15 +4438,17 @@ 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;
 		break;
+	default:
+		codec->pin_amp_workaround = 1;
 	}
 
 	apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
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..2508f8109f11 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3398,8 +3398,10 @@ 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 +3436,7 @@ 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 +3450,7 @@ 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 +4425,7 @@ 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);
 }
 
@@ -5269,7 +5271,9 @@ 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 +5316,9 @@ 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 +5365,7 @@ 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 +5391,7 @@ 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 +5407,13 @@ 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 +5454,8 @@ 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);
@@ -6079,7 +6095,7 @@ 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 +6312,7 @@ 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 +6730,7 @@ 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/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/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 9538d41c1e5c..8005fe6db66f 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/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 1765a197acb0..f23700359c67 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/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/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 453b0925a0b5..3075a426124c 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/soc-core.c b/sound/soc/soc-core.c
index 77f35292f1b0..5d308e92f9b0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1072,6 +1072,8 @@ static int soc_probe_platform(struct snd_soc_card *card,
 		snd_soc_dapm_new_dai_widgets(&platform->dapm, dai);
 	}
 
+	platform->dapm.idle_bias_off = 1;
+
 	if (driver->probe) {
 		ret = driver->probe(platform);
 		if (ret < 0) {