diff --git a/[refs] b/[refs]
index 5553c2cd0b26..0b0a6a7535ce 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: b0791dda813c179e539b0fc1ecd3f5f30f2571e2
+refs/heads/master: 5807c3bf68eb489032ca8ff70b3d3c833fd8172b
diff --git a/trunk/Documentation/sound/alsa/HD-Audio-Models.txt b/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
index 03f7897c6414..d97d992ced14 100644
--- a/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -43,9 +43,7 @@ ALC680
 
 ALC882/883/885/888/889
 ======================
-  acer-aspire-4930g	Acer Aspire 4930G/5930G/6530G/6930G/7730G
-  acer-aspire-8930g	Acer Aspire 8330G/6935G
-  acer-aspire		Acer Aspire others
+  N/A
 
 ALC861/660
 ==========
diff --git a/trunk/include/sound/core.h b/trunk/include/sound/core.h
index bc056687f647..b6e0f57d451d 100644
--- a/trunk/include/sound/core.h
+++ b/trunk/include/sound/core.h
@@ -325,13 +325,6 @@ void release_and_free_resource(struct resource *res);
 
 /* --- */
 
-/* sound printk debug levels */
-enum {
-	SND_PR_ALWAYS,
-	SND_PR_DEBUG,
-	SND_PR_VERBOSE,
-};
-
 #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
 __printf(4, 5)
 void __snd_printk(unsigned int level, const char *file, int line,
@@ -361,8 +354,6 @@ void __snd_printk(unsigned int level, const char *file, int line,
  */
 #define snd_printd(fmt, args...) \
 	__snd_printk(1, __FILE__, __LINE__, fmt, ##args)
-#define _snd_printd(level, fmt, args...) \
-	__snd_printk(level, __FILE__, __LINE__, fmt, ##args)
 
 /**
  * snd_BUG - give a BUG warning message and stack trace
@@ -392,7 +383,6 @@ void __snd_printk(unsigned int level, const char *file, int line,
 #else /* !CONFIG_SND_DEBUG */
 
 #define snd_printd(fmt, args...)	do { } while (0)
-#define _snd_printd(level, fmt, args...) do { } while (0)
 #define snd_BUG()			do { } while (0)
 static inline int __snd_bug_on(int cond)
 {
diff --git a/trunk/sound/core/vmaster.c b/trunk/sound/core/vmaster.c
index 857586135d18..14a286a7bf2b 100644
--- a/trunk/sound/core/vmaster.c
+++ b/trunk/sound/core/vmaster.c
@@ -419,7 +419,6 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_master);
  * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control
  * @kcontrol: vmaster kctl element
  * @hook: the hook function
- * @private_data: the private_data pointer to be saved
  *
  * Adds the given hook to the vmaster control element so that it's called
  * at each time when the value is changed.
diff --git a/trunk/sound/isa/sscape.c b/trunk/sound/isa/sscape.c
index 8490f59709bb..b4a6aa960f4b 100644
--- a/trunk/sound/isa/sscape.c
+++ b/trunk/sound/isa/sscape.c
@@ -1019,15 +1019,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
 	irq_cfg = get_irq_config(sscape->type, irq[dev]);
 	if (irq_cfg == INVALID_IRQ) {
 		snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]);
-		err = -ENXIO;
-		goto _release_dma;
+		return -ENXIO;
 	}
 
 	mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]);
 	if (mpu_irq_cfg == INVALID_IRQ) {
 		snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]);
-		err = -ENXIO;
-		goto _release_dma;
+		return -ENXIO;
 	}
 
 	/*
diff --git a/trunk/sound/last.c b/trunk/sound/last.c
index 7ffc182e0844..bdd0857b8871 100644
--- a/trunk/sound/last.c
+++ b/trunk/sound/last.c
@@ -38,4 +38,4 @@ static int __init alsa_sound_last_init(void)
 	return 0;
 }
 
-late_initcall_sync(alsa_sound_last_init);
+__initcall(alsa_sound_last_init);
diff --git a/trunk/sound/oss/msnd_pinnacle.c b/trunk/sound/oss/msnd_pinnacle.c
index 536c4c0514d3..2c79d60a725f 100644
--- a/trunk/sound/oss/msnd_pinnacle.c
+++ b/trunk/sound/oss/msnd_pinnacle.c
@@ -1294,8 +1294,6 @@ static int __init calibrate_adc(WORD srate)
 
 static int upload_dsp_code(void)
 {
-	int ret = 0;
-
 	msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS);
 #ifndef HAVE_DSPCODEH
 	INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE);
@@ -1314,8 +1312,7 @@ static int upload_dsp_code(void)
 	memcpy_toio(dev.base, PERMCODE, PERMCODESIZE);
 	if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) {
 		printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n");
-		ret = -ENODEV;
-		goto out;
+		return -ENODEV;
 	}
 #ifdef HAVE_DSPCODEH
 	printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n");
@@ -1323,13 +1320,12 @@ static int upload_dsp_code(void)
 	printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n");
 #endif
 
-out:
 #ifndef HAVE_DSPCODEH
 	vfree(INITCODE);
 	vfree(PERMCODE);
 #endif
 
-	return ret;
+	return 0;
 }
 
 #ifdef MSND_CLASSIC
diff --git a/trunk/sound/pci/asihpi/hpi_internal.h b/trunk/sound/pci/asihpi/hpi_internal.h
index bc86cb726d79..8c63200cf339 100644
--- a/trunk/sound/pci/asihpi/hpi_internal.h
+++ b/trunk/sound/pci/asihpi/hpi_internal.h
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2012  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned.
 If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and
 HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle.
 */
-u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
+int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
 							   /**< memory handle */
 	u32 size, /**< Size in bytes to allocate */
 	struct pci_dev *p_os_reference
diff --git a/trunk/sound/pci/asihpi/hpios.c b/trunk/sound/pci/asihpi/hpios.c
index 5ef4fe964366..87f4385fe8c7 100644
--- a/trunk/sound/pci/asihpi/hpios.c
+++ b/trunk/sound/pci/asihpi/hpios.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2012  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -39,11 +39,11 @@ void hpios_delay_micro_seconds(u32 num_micro_sec)
 
 }
 
-/** Allocate an area of locked memory for bus master DMA operations.
+/** Allocated an area of locked memory for bus master DMA operations.
 
-If allocation fails, return 1, and *pMemArea.size = 0
+On error, return -ENOMEM, and *pMemArea.size = 0
 */
-u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
+int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
 	struct pci_dev *pdev)
 {
 	/*?? any benefit in using managed dmam_alloc_coherent? */
@@ -62,7 +62,7 @@ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
 		HPI_DEBUG_LOG(WARNING,
 			"failed to allocate %d bytes locked memory\n", size);
 		p_mem_area->size = 0;
-		return 1;
+		return -ENOMEM;
 	}
 }
 
diff --git a/trunk/sound/pci/echoaudio/echoaudio_dsp.c b/trunk/sound/pci/echoaudio/echoaudio_dsp.c
index d8c670c9d62c..64417a733220 100644
--- a/trunk/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/trunk/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->comm_page))
+	if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page))
 		return -EPERM;
 
 	/* See if the ASIC is present and working - only if the DSP is already loaded */
diff --git a/trunk/sound/pci/hda/hda_codec.c b/trunk/sound/pci/hda/hda_codec.c
index 841475cc13b6..7a8fcc4c15f8 100644
--- a/trunk/sound/pci/hda/hda_codec.c
+++ b/trunk/sound/pci/hda/hda_codec.c
@@ -5444,6 +5444,10 @@ 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/trunk/sound/pci/hda/hda_codec.h b/trunk/sound/pci/hda/hda_codec.h
index 56b4f74c0b13..9a9f372e1be4 100644
--- a/trunk/sound/pci/hda/hda_codec.h
+++ b/trunk/sound/pci/hda/hda_codec.h
@@ -851,9 +851,6 @@ struct hda_codec {
 	unsigned int pin_amp_workaround:1; /* pin out-amp takes index
 					    * (e.g. Conexant codecs)
 					    */
-	unsigned int single_adc_amp:1; /* adc in-amp takes no index
-					* (e.g. CX20549 codec)
-					*/
 	unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */
 	unsigned int pins_shutup:1;	/* pins are shut up */
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
diff --git a/trunk/sound/pci/hda/hda_eld.c b/trunk/sound/pci/hda/hda_eld.c
index 4c054f4486b9..b58b4b1687fa 100644
--- a/trunk/sound/pci/hda/hda_eld.c
+++ b/trunk/sound/pci/hda/hda_eld.c
@@ -418,7 +418,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
 	else
 		buf2[0] = '\0';
 
-	_snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:"
+	printk(KERN_INFO "HDMI: supports coding type %s:"
 			" channels = %d, rates =%s%s\n",
 			cea_audio_coding_type_names[a->format],
 			a->channels,
@@ -442,14 +442,14 @@ void snd_hdmi_show_eld(struct hdmi_eld *e)
 {
 	int i;
 
-	_snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n",
+	printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n",
 			e->monitor_name,
 			eld_connection_type_names[e->conn_type]);
 
 	if (e->spk_alloc) {
 		char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
 		snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
-		_snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf);
+		printk(KERN_INFO "HDMI: available speakers:%s\n", buf);
 	}
 
 	for (i = 0; i < e->sad_count; i++)
diff --git a/trunk/sound/pci/hda/hda_intel.c b/trunk/sound/pci/hda/hda_intel.c
index 1f350522bed4..c19e71a94e1b 100644
--- a/trunk/sound/pci/hda/hda_intel.c
+++ b/trunk/sound/pci/hda/hda_intel.c
@@ -783,13 +783,11 @@ 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 (loopcounter = 0;; loopcounter++) {
+	for (;;) {
 		if (chip->polling_mode || do_poll) {
 			spin_lock_irq(&chip->reg_lock);
 			azx_update_rirb(chip);
@@ -805,7 +803,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 		if (time_after(jiffies, timeout))
 			break;
-		if (bus->needs_damn_long_delay || loopcounter > 3000)
+		if (bus->needs_damn_long_delay)
 			msleep(2); /* temporary workaround */
 		else {
 			udelay(10);
@@ -2353,17 +2351,6 @@ 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);
@@ -2410,8 +2397,7 @@ static int azx_resume(struct pci_dev *pci)
 		return -EIO;
 	azx_init_pci(chip);
 
-	if (snd_hda_codecs_inuse(chip->bus))
-		azx_init_chip(chip, 1);
+	azx_init_chip(chip, 1);
 
 	snd_hda_resume(chip->bus);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
diff --git a/trunk/sound/pci/hda/hda_proc.c b/trunk/sound/pci/hda/hda_proc.c
index e59e2f059b6e..254ab5204603 100644
--- a/trunk/sound/pci/hda/hda_proc.c
+++ b/trunk/sound/pci/hda/hda_proc.c
@@ -651,16 +651,9 @@ static void print_codec_info(struct snd_info_entry *entry,
 			snd_iprintf(buffer, "  Amp-In caps: ");
 			print_amp_caps(buffer, codec, nid, HDA_INPUT);
 			snd_iprintf(buffer, "  Amp-In vals: ");
-			if (wid_type == AC_WID_PIN ||
-			    (codec->single_adc_amp &&
-			     wid_type == AC_WID_AUD_IN))
-				print_amp_vals(buffer, codec, nid, HDA_INPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       1);
-			else
-				print_amp_vals(buffer, codec, nid, HDA_INPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       conn_len);
+			print_amp_vals(buffer, codec, nid, HDA_INPUT,
+				       wid_caps & AC_WCAP_STEREO,
+				       wid_type == AC_WID_PIN ? 1 : conn_len);
 		}
 		if (wid_caps & AC_WCAP_OUT_AMP) {
 			snd_iprintf(buffer, "  Amp-Out caps: ");
diff --git a/trunk/sound/pci/hda/patch_conexant.c b/trunk/sound/pci/hda/patch_conexant.c
index d906c5b74cf0..8c6523bbc797 100644
--- a/trunk/sound/pci/hda/patch_conexant.c
+++ b/trunk/sound/pci/hda/patch_conexant.c
@@ -141,6 +141,7 @@ struct conexant_spec {
 	unsigned int hp_laptop:1;
 	unsigned int asus:1;
 	unsigned int pin_eapd_ctrls:1;
+	unsigned int single_adc_amp:1;
 
 	unsigned int adc_switching:1;
 
@@ -686,26 +687,27 @@ static const struct hda_channel_mode cxt5045_modes[1] = {
 static const struct hda_input_mux cxt5045_capture_source = {
 	.num_items = 2,
 	.items = {
-		{ "Internal Mic", 0x1 },
-		{ "Mic",          0x2 },
+		{ "IntMic", 0x1 },
+		{ "ExtMic", 0x2 },
 	}
 };
 
 static const struct hda_input_mux cxt5045_capture_source_benq = {
-	.num_items = 4,
+	.num_items = 5,
 	.items = {
-		{ "Internal Mic", 0x1 },
-		{ "Mic",          0x2 },
-		{ "Line",         0x3 },
-		{ "Mixer",        0x0 },
+		{ "IntMic", 0x1 },
+		{ "ExtMic", 0x2 },
+		{ "LineIn", 0x3 },
+		{ "CD",     0x4 },
+		{ "Mixer",  0x0 },
 	}
 };
 
 static const struct hda_input_mux cxt5045_capture_source_hp530 = {
 	.num_items = 2,
 	.items = {
-		{ "Mic",          0x1 },
-		{ "Internal Mic", 0x2 },
+		{ "ExtMic", 0x1 },
+		{ "IntMic", 0x2 },
 	}
 };
 
@@ -796,8 +798,10 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec,
 }
 
 static const struct snd_kcontrol_new cxt5045_mixers[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
@@ -818,15 +822,27 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = {
 };
 
 static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
+	HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
 
 	{}
 };
 
 static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
@@ -930,10 +946,10 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 	/* Output controls */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
 	
 	/* Modes for retasking pin widgets */
 	CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT),
@@ -944,16 +960,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 
 	/* Loopback mixer controls */
 
-	HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT),
-	HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT),
-	HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Input Source",
@@ -962,8 +978,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 		.put = conexant_mux_enum_put,
 	},
 	/* Audio input controls */
-	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -985,6 +1009,10 @@ static const struct hda_verb cxt5045_test_init_verbs[] = {
 	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
 
+	/* Start with output sum widgets muted and their output gains at min */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
 	/* Unmute retasking pin widget output buffers since the default
 	 * state appears to be output.  As the pin mode is changed by the
 	 * user the pin mode control will take care of enabling the pin's
@@ -999,11 +1027,11 @@ static const struct hda_verb cxt5045_test_init_verbs[] = {
 	/* Set ADC connection select to match default mixer setting (mic1
 	 * pin)
 	 */
-	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
 
 	/* Mute all inputs to mixer widget (even unconnected ones) */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */
@@ -1082,7 +1110,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
-	codec->single_adc_amp = 1;
+	codec->pin_amp_workaround = 1;
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
@@ -3971,14 +3999,9 @@ static void cx_auto_init_output(struct hda_codec *codec)
 	int i;
 
 	mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids);
-	for (i = 0; i < cfg->hp_outs; i++) {
-		unsigned int val = PIN_OUT;
-		if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) &
-		    AC_PINCAP_HP_DRV)
-			val |= AC_PINCTL_HP_EN;
+	for (i = 0; i < cfg->hp_outs; i++)
 		snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, val);
-	}
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
 	mute_outputs(codec, cfg->hp_outs, cfg->hp_pins);
 	mute_outputs(codec, cfg->line_outs, cfg->line_out_pins);
 	mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins);
@@ -4197,7 +4220,7 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
 		int idx = get_input_connection(codec, adc_nid, nid);
 		if (idx < 0)
 			continue;
-		if (codec->single_adc_amp)
+		if (spec->single_adc_amp)
 			idx = 0;
 		return cx_auto_add_volume_idx(codec, label, pfx,
 					      cidx, adc_nid, HDA_INPUT, idx);
@@ -4252,7 +4275,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
 		if (cidx < 0)
 			continue;
 		input_conn[i] = spec->imux_info[i].adc;
-		if (!codec->single_adc_amp)
+		if (!spec->single_adc_amp)
 			input_conn[i] |= cidx << 8;
 		if (i > 0 && input_conn[i] != input_conn[0])
 			multi_connection = 1;
@@ -4396,10 +4419,8 @@ static void apply_pin_fixup(struct hda_codec *codec,
 
 enum {
 	CXT_PINCFG_LENOVO_X200,
-	CXT_PINCFG_LENOVO_TP410,
 };
 
-/* ThinkPad X200 & co with cxt5051 */
 static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
 	{ 0x17, 0x21a11000 }, /* dock-mic */
@@ -4408,33 +4429,15 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
 	{}
 };
 
-/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */
-static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = {
-	{ 0x19, 0x042110ff }, /* HP (seq# overridden) */
-	{ 0x1a, 0x21a190f0 }, /* dock-mic */
-	{ 0x1c, 0x212140ff }, /* dock-HP */
-	{}
-};
-
 static const struct cxt_pincfg *cxt_pincfg_tbl[] = {
 	[CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200,
-	[CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410,
 };
 
-static const struct snd_pci_quirk cxt5051_fixups[] = {
+static const struct snd_pci_quirk cxt_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
 	{}
 };
 
-static const struct snd_pci_quirk cxt5066_fixups[] = {
-	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
-	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
-	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
-	SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
-	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
-	{}
-};
-
 /* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
  * can be created (bko#42825)
  */
@@ -4463,21 +4466,19 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->pin_amp_workaround = 1;
 
 	switch (codec->vendor_id) {
 	case 0x14f15045:
-		codec->single_adc_amp = 1;
+		spec->single_adc_amp = 1;
 		break;
 	case 0x14f15051:
 		add_cx5051_fake_mutes(codec);
-		codec->pin_amp_workaround = 1;
-		apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl);
 		break;
-	default:
-		codec->pin_amp_workaround = 1;
-		apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl);
 	}
 
+	apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
+
 	/* Show mute-led control only on HP laptops
 	 * This is a sort of white-list: on HP laptops, EAPD corresponds
 	 * only to the mute-LED without actualy amp function.  Meanwhile,
diff --git a/trunk/sound/pci/hda/patch_hdmi.c b/trunk/sound/pci/hda/patch_hdmi.c
index 83f345f3c961..540cd13f7f15 100644
--- a/trunk/sound/pci/hda/patch_hdmi.c
+++ b/trunk/sound/pci/hda/patch_hdmi.c
@@ -757,6 +757,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	struct hdmi_spec *spec = codec->spec;
 	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
 	int pin_nid;
+	int pd = !!(res & AC_UNSOL_RES_PD);
+	int eldv = !!(res & AC_UNSOL_RES_ELDV);
 	int pin_idx;
 	struct hda_jack_tbl *jack;
 
@@ -766,10 +768,9 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	pin_nid = jack->nid;
 	jack->jack_dirty = 1;
 
-	_snd_printd(SND_PR_VERBOSE,
+	printk(KERN_INFO
 		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid,
-		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
+		codec->addr, pin_nid, pd, eldv);
 
 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
 	if (pin_idx < 0)
@@ -991,7 +992,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 	if (eld->monitor_present)
 		eld_valid	= !!(present & AC_PINSENSE_ELDV);
 
-	_snd_printd(SND_PR_VERBOSE,
+	printk(KERN_INFO
 		"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
 		codec->addr, pin_nid, eld->monitor_present, eld_valid);
 
diff --git a/trunk/sound/pci/hda/patch_realtek.c b/trunk/sound/pci/hda/patch_realtek.c
index 7810913d07a0..9917e55d6f11 100644
--- a/trunk/sound/pci/hda/patch_realtek.c
+++ b/trunk/sound/pci/hda/patch_realtek.c
@@ -1445,13 +1445,6 @@ enum {
 	ALC_FIXUP_ACT_BUILD,
 };
 
-static void alc_apply_pincfgs(struct hda_codec *codec,
-			      const struct alc_pincfg *cfg)
-{
-	for (; cfg->nid; cfg++)
-		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-}
-
 static void alc_apply_fixup(struct hda_codec *codec, int action)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1485,7 +1478,9 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
 			snd_printdd(KERN_INFO "hda_codec: %s: "
 				    "Apply pincfg for %s\n",
 				    codec->chip_name, modelname);
-			alc_apply_pincfgs(codec, cfg);
+			for (; cfg->nid; cfg++)
+				snd_hda_codec_set_pincfg(codec, cfg->nid,
+							 cfg->val);
 			break;
 		case ALC_FIXUP_VERBS:
 			if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs)
@@ -3403,10 +3398,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 	for (;;) {
 		badness = fill_and_eval_dacs(codec, fill_hardwired,
 					     fill_mio_first);
-		if (badness < 0) {
-			kfree(best_cfg);
+		if (badness < 0)
 			return badness;
-		}
 		debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n",
 			      cfg->line_out_type, fill_hardwired, fill_mio_first,
 			      badness);
@@ -3441,7 +3434,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 			cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
 			fill_hardwired = true;
 			continue;
-		}
+		} 
 		if (cfg->hp_outs > 0 &&
 		    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
 			cfg->speaker_outs = cfg->line_outs;
@@ -3455,7 +3448,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 			cfg->line_out_type = AUTO_PIN_HP_OUT;
 			fill_hardwired = true;
 			continue;
-		}
+		} 
 		break;
 	}
 
@@ -4430,7 +4423,7 @@ static int alc_parse_auto_config(struct hda_codec *codec,
 static int alc880_parse_auto_config(struct hda_codec *codec)
 {
 	static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
-	static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+	static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; 
 	return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids);
 }
 
@@ -4866,7 +4859,6 @@ enum {
 	ALC260_FIXUP_GPIO1_TOGGLE,
 	ALC260_FIXUP_REPLACER,
 	ALC260_FIXUP_HP_B1900,
-	ALC260_FIXUP_KN1,
 };
 
 static void alc260_gpio1_automute(struct hda_codec *codec)
@@ -4894,36 +4886,6 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
 	}
 }
 
-static void alc260_fixup_kn1(struct hda_codec *codec,
-			     const struct alc_fixup *fix, int action)
-{
-	struct alc_spec *spec = codec->spec;
-	static const struct alc_pincfg pincfgs[] = {
-		{ 0x0f, 0x02214000 }, /* HP/speaker */
-		{ 0x12, 0x90a60160 }, /* int mic */
-		{ 0x13, 0x02a19000 }, /* ext mic */
-		{ 0x18, 0x01446000 }, /* SPDIF out */
-		/* disable bogus I/O pins */
-		{ 0x10, 0x411111f0 },
-		{ 0x11, 0x411111f0 },
-		{ 0x14, 0x411111f0 },
-		{ 0x15, 0x411111f0 },
-		{ 0x16, 0x411111f0 },
-		{ 0x17, 0x411111f0 },
-		{ 0x19, 0x411111f0 },
-		{ }
-	};
-
-	switch (action) {
-	case ALC_FIXUP_ACT_PRE_PROBE:
-		alc_apply_pincfgs(codec, pincfgs);
-		break;
-	case ALC_FIXUP_ACT_PROBE:
-		spec->init_amp = ALC_INIT_NONE;
-		break;
-	}
-}
-
 static const struct alc_fixup alc260_fixups[] = {
 	[ALC260_FIXUP_HP_DC5750] = {
 		.type = ALC_FIXUP_PINS,
@@ -4974,11 +4936,7 @@ static const struct alc_fixup alc260_fixups[] = {
 		.v.func = alc260_fixup_gpio1_toggle,
 		.chained = true,
 		.chain_id = ALC260_FIXUP_COEF,
-	},
-	[ALC260_FIXUP_KN1] = {
-		.type = ALC_FIXUP_FUNC,
-		.v.func = alc260_fixup_kn1,
-	},
+	}
 };
 
 static const struct snd_pci_quirk alc260_fixup_tbl[] = {
@@ -4988,7 +4946,6 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750),
 	SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900),
 	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1),
-	SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1),
 	SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER),
 	SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF),
 	{}
@@ -5312,9 +5269,7 @@ static const struct alc_fixup alc882_fixups[] = {
 			{ 0x16, 0x99130111 }, /* CLFE speaker */
 			{ 0x17, 0x99130112 }, /* surround speaker */
 			{ }
-		},
-		.chained = true,
-		.chain_id = ALC882_FIXUP_GPIO1,
+		}
 	},
 	[ALC882_FIXUP_ACER_ASPIRE_8930G] = {
 		.type = ALC_FIXUP_PINS,
@@ -5357,9 +5312,7 @@ static const struct alc_fixup alc882_fixups[] = {
 			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
 			{ 0x20, AC_VERB_SET_PROC_COEF, 0x3050 },
 			{ }
-		},
-		.chained = true,
-		.chain_id = ALC882_FIXUP_GPIO1,
+		}
 	},
 	[ALC885_FIXUP_MACPRO_GPIO] = {
 		.type = ALC_FIXUP_FUNC,
@@ -5405,10 +5358,7 @@ 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),
@@ -5434,13 +5384,11 @@ 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),
@@ -5451,13 +5399,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	{}
 };
 
-static const struct alc_model_fixup alc882_fixup_models[] = {
-	{.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"},
-	{.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"},
-	{.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"},
-	{}
-};
-
 /*
  * BIOS auto configuration
  */
@@ -5498,8 +5439,7 @@ static int patch_alc882(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
-		       alc882_fixups);
+	alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
 	alc_auto_parse_customize_define(codec);
@@ -5641,13 +5581,13 @@ 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)
@@ -6112,7 +6052,6 @@ static const struct alc_fixup alc269_fixups[] = {
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
-	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
@@ -6140,7 +6079,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	 * Basically the device should work as is without the fixup table.
 	 * If BIOS doesn't give a proper info, enable the corresponding
 	 * fixup entry.
-	 */
+	 */ 
 	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
 		      ALC269_FIXUP_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC),
@@ -6252,6 +6191,8 @@ 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;
@@ -6284,8 +6225,6 @@ 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)
@@ -6357,7 +6296,7 @@ static void alc_fixup_no_jack_detect(struct hda_codec *codec,
 {
 	if (action == ALC_FIXUP_ACT_PRE_PROBE)
 		codec->no_jack_detect = 1;
-}
+}	
 
 static const struct alc_fixup alc861_fixups[] = {
 	[ALC861_FIXUP_FSC_AMILO_PI1505] = {
@@ -6775,7 +6714,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	 * Basically the device should work as is without the fixup table.
 	 * If BIOS doesn't give a proper info, enable the corresponding
 	 * fixup entry.
-	 */
+	 */ 
 	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3),
 	SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1),
@@ -6862,6 +6801,8 @@ 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);
@@ -6878,9 +6819,6 @@ 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/trunk/sound/pci/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c
index 2cb1e08f962a..33a9946b492c 100644
--- a/trunk/sound/pci/hda/patch_sigmatel.c
+++ b/trunk/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_NONE &&
-		    !is_jack_detectable(codec, nid)) {
-			stac_toggle_power_map(codec, nid, 1);
+		if (def_conf != AC_JACK_PORT_COMPLEX) {
+			if (def_conf != AC_JACK_PORT_NONE)
+				stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
 		if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) {
@@ -5063,11 +5063,12 @@ static void stac92xx_update_led_status(struct hda_codec *codec, int enabled)
 	if (spec->gpio_led_polarity)
 		muted = !muted;
 
+	/*polarity defines *not* muted state level*/
 	if (!spec->vref_mute_led_nid) {
 		if (muted)
-			spec->gpio_data |= spec->gpio_led;
+			spec->gpio_data &= ~spec->gpio_led; /* orange */
 		else
-			spec->gpio_data &= ~spec->gpio_led;
+			spec->gpio_data |= spec->gpio_led; /* white */
 		stac_gpio_set(codec, spec->gpio_mask,
 				spec->gpio_dir, spec->gpio_data);
 	} else {
diff --git a/trunk/sound/pci/rme9652/hdsp.c b/trunk/sound/pci/rme9652/hdsp.c
index 0b2aea2ce172..b68cdec03b9e 100644
--- a/trunk/sound/pci/rme9652/hdsp.c
+++ b/trunk/sound/pci/rme9652/hdsp.c
@@ -5170,7 +5170,6 @@ 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/trunk/sound/soc/codecs/cs42l73.c b/trunk/sound/soc/codecs/cs42l73.c
index 07c44b71f096..3686417f5ea5 100644
--- a/trunk/sound/soc/codecs/cs42l73.c
+++ b/trunk/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),
 
diff --git a/trunk/sound/soc/omap/omap-pcm.c b/trunk/sound/soc/omap/omap-pcm.c
index 5a649da9122a..a59bd352d342 100644
--- a/trunk/sound/soc/omap/omap-pcm.c
+++ b/trunk/sound/soc/omap/omap-pcm.c
@@ -401,10 +401,6 @@ 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;
 }