Merge tag 'v3.4' with SCSI updates, needed for subsequent firewire-sbp2 changes

Linux 3.4

39 files changed, 517 insertions, 235 deletions

diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 14a286a..8575861 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 b4a6aa9..8490f59 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 bdd0857..7ffc182 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 2c79d60..536c4c0 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 8816804..5ca0939 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 8c63200..bc86cb7 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 87f4385..5ef4fe9 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/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 64417a7..d8c670c 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip)
 	const struct firmware *fw;
 	int box_type, err;
 
-	if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page))
+	if (snd_BUG_ON(!chip->comm_page))
 		return -EPERM;
 
 	/* See if the ASIC is present and working - only if the DSP is already loaded */
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 7a8fcc4..841475c 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -5444,10 +5444,6 @@ int snd_hda_suspend(struct hda_bus *bus)
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		if (hda_codec_is_power_on(codec))
 			hda_call_codec_suspend(codec);
-		else /* forcibly change the power to D3 even if not used */
-			hda_set_power_state(codec,
-					    codec->afg ? codec->afg : codec->mfg,
-					    AC_PWRST_D3);
 		if (codec->patch_ops.post_suspend)
 			codec->patch_ops.post_suspend(codec);
 	}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 9a9f372..56b4f74 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 b58b4b1..4c054f4 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_intel.c b/sound/pci/hda/hda_intel.c
index c19e71a..1f35052 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -783,11 +783,13 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 {
 	struct azx *chip = bus->private_data;
 	unsigned long timeout;
+	unsigned long loopcounter;
 	int do_poll = 0;
 
  again:
 	timeout = jiffies + msecs_to_jiffies(1000);
-	for (;;) {
+
+	for (loopcounter = 0;; loopcounter++) {
 		if (chip->polling_mode || do_poll) {
 			spin_lock_irq(&chip->reg_lock);
 			azx_update_rirb(chip);
@@ -803,7 +805,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 		if (time_after(jiffies, timeout))
 			break;
-		if (bus->needs_damn_long_delay)
+		if (bus->needs_damn_long_delay || loopcounter > 3000)
 			msleep(2); /* temporary workaround */
 		else {
 			udelay(10);
@@ -2351,6 +2353,17 @@ static void azx_power_notify(struct hda_bus *bus)
  * power management
  */
 
+static int snd_hda_codecs_inuse(struct hda_bus *bus)
+{
+	struct hda_codec *codec;
+
+	list_for_each_entry(codec, &bus->codec_list, list) {
+		if (snd_hda_codec_needs_resume(codec))
+			return 1;
+	}
+	return 0;
+}
+
 static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
@@ -2397,7 +2410,8 @@ static int azx_resume(struct pci_dev *pci)
 		return -EIO;
 	azx_init_pci(chip);
 
-	azx_init_chip(chip, 1);
+	if (snd_hda_codecs_inuse(chip->bus))
+		azx_init_chip(chip, 1);
 
 	snd_hda_resume(chip->bus);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 254ab52..e59e2f0 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 8c6523b..d906c5b 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 540cd13..83f345f 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 9917e55..7810913 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,
@@ -5358,7 +5405,10 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
 		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
+	SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G",
+		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	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),
@@ -5384,11 +5434,13 @@ 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),
 
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
+	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
@@ -5399,6 +5451,13 @@ 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 +5498,8 @@ 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);
@@ -5581,13 +5641,13 @@ static int patch_alc262(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
 	}
 #endif
-	alc_auto_parse_customize_define(codec);
-
 	alc_fix_pll_init(codec, 0x20, 0x0a, 10);
 
 	alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc262_parse_auto_config(codec);
 	if (err < 0)
@@ -6052,6 +6112,7 @@ 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 +6140,7 @@ 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),
@@ -6191,8 +6252,6 @@ static int patch_alc269(struct hda_codec *codec)
 
 	spec->mixer_nid = 0x0b;
 
-	alc_auto_parse_customize_define(codec);
-
 	err = alc_codec_rename_from_preset(codec);
 	if (err < 0)
 		goto error;
@@ -6225,6 +6284,8 @@ static int patch_alc269(struct hda_codec *codec)
 		       alc269_fixup_tbl, alc269_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc269_parse_auto_config(codec);
 	if (err < 0)
@@ -6296,7 +6357,7 @@ 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 +6775,7 @@ 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),
@@ -6801,8 +6862,6 @@ static int patch_alc662(struct hda_codec *codec)
 	/* handle multiple HPs as is */
 	spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 
-	alc_auto_parse_customize_define(codec);
-
 	alc_fix_pll_init(codec, 0x20, 0x04, 15);
 
 	err = alc_codec_rename_from_preset(codec);
@@ -6819,6 +6878,9 @@ static int patch_alc662(struct hda_codec *codec)
 	alc_pick_fixup(codec, alc662_fixup_models,
 		       alc662_fixup_tbl, alc662_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc662_parse_auto_config(codec);
 	if (err < 0)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 33a9946..2cb1e08 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4415,9 +4415,9 @@ static int stac92xx_init(struct hda_codec *codec)
 		def_conf = get_defcfg_connect(def_conf);
 		/* skip any ports that don't have jacks since presence
  		 * detection is useless */
-		if (def_conf != AC_JACK_PORT_COMPLEX) {
-			if (def_conf != AC_JACK_PORT_NONE)
-				stac_toggle_power_map(codec, nid, 1);
+		if (def_conf != AC_JACK_PORT_NONE &&
+		    !is_jack_detectable(codec, nid)) {
+			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
 		if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) {
@@ -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/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index b68cdec..0b2aea2 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5170,6 +5170,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
 	strcpy(hw->name, "HDSP hwdep interface");
 
 	hw->ops.ioctl = snd_hdsp_hwdep_ioctl;
+	hw->ops.ioctl_compat = snd_hdsp_hwdep_ioctl;
 
 	return 0;
 }
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index df3ac73..b39ad35 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 6508e8b..59d8efa 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 f8e10ce..b3e24f2 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 78979b3..3686417 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -568,22 +568,22 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
 			attn_tlv),
 
 	SOC_SINGLE_TLV("SPK-IP Mono Volume",
-			CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv),
+			CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv),
 	SOC_SINGLE_TLV("SPK-XSP Mono Volume",
-			CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv),
+			CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv),
 	SOC_SINGLE_TLV("SPK-ASP Mono Volume",
-			CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv),
+			CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv),
 	SOC_SINGLE_TLV("SPK-VSP Mono Volume",
-			CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv),
+			CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv),
 
 	SOC_SINGLE_TLV("ESL-IP Mono Volume",
-			CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv),
+			CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv),
 	SOC_SINGLE_TLV("ESL-XSP Mono Volume",
-			CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv),
+			CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv),
 	SOC_SINGLE_TLV("ESL-ASP Mono Volume",
-			CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv),
+			CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv),
 	SOC_SINGLE_TLV("ESL-VSP Mono Volume",
-			CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv),
+			CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv),
 
 	SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum),
 
@@ -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 d192626..8e92fb8 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 16d55f9..df1e07f 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 2d8c6b8..dc7509b 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 8c4c959..aa12c6b 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 7c49642..2de12eb 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_4,
+				    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 f13f288..6c028c4 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 1765a19..f237003 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 e00dd0b..deafbfa 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 a59bd35..5a649da 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 609abd5..d085837 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 7218507..79fbeea 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 378cc5b..74ed2df 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/sh/migor.c b/sound/soc/sh/migor.c
index 9d9ad8d..8526e1e 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -35,7 +35,7 @@ static unsigned long siumckb_recalc(struct clk *clk)
 	return codec_freq;
 }
 
-static struct clk_ops siumckb_clk_ops = {
+static struct sh_clk_ops siumckb_clk_ops = {
 	.recalc = siumckb_recalc,
 };
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e19c24a..c88d974 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 5cbd2d76..1bb6d4a 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 33509de..e533499 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 475428c..9ff2c60 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;
 }