Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Fix S3/S4 problem on machines with VREF-pin mute-LED
  ALSA: hda_intel - revert a quirk that affect VIA chipsets
  ALSA: hda - Avoid touching mute-VREF pin for IDT codecs
  firmware: Sigma: Fix endianess issues
  firmware: Sigma: Skip header during CRC generation
  firmware: Sigma: Prevent out of bounds memory access
  ALSA: usb-audio - Support for Roland GAIA SH-01 Synthesizer
  ASoC: Supply dcs_codes for newer WM1811 revisions
  ASoC: Error out if we can't generate a LRCLK at all for WM8994
  ASoC: Correct name of Speyside Main Speaker widget
  ASoC: skip resume of soc-audio devices without codecs
  ASoC: cs42l51: Fix off-by-one for reg_cache_size
  ASoC: drop support for PlayPaq with WM8510
  ASoC: mpc8610: tell the CS4270 codec that it's the master
  ASoC: cs4720: use snd_soc_cache_sync()
  ASoC: SAMSUNG: Fix build error
  ASoC: max9877: Update register if either val or val2 is changed
  ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET

15 files changed, 74 insertions, 542 deletions

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 096507d2ca9a..7d98240def0b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f3658658548e..d8d2f9dccd9b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4441,7 +4441,9 @@ static int stac92xx_init(struct hda_codec *codec)
 		int pinctl, def_conf;
 
 		/* power on when no jack detection is available */
-		if (!spec->hp_detect) {
+		/* or when the VREF is used for controlling LED */
+		if (!spec->hp_detect ||
+		    (spec->gpio_led > 8 && spec->gpio_led == nid)) {
 			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
@@ -5055,20 +5057,6 @@ static int stac92xx_pre_resume(struct hda_codec *codec)
 	return 0;
 }
 
-static int stac92xx_post_suspend(struct hda_codec *codec)
-{
-	struct sigmatel_spec *spec = codec->spec;
-	if (spec->gpio_led > 8) {
-		/* with vref-out pin used for mute led control
-		 * codec AFG is prevented from D3 state, but on
-		 * system suspend it can (and should) be used
-		 */
-		snd_hda_codec_read(codec, codec->afg, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	}
-	return 0;
-}
-
 static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state)
 {
@@ -5668,8 +5656,6 @@ again:
 		} else {
 			codec->patch_ops.set_power_state =
 					stac92xx_set_power_state;
-			codec->patch_ops.post_suspend =
-					stac92xx_post_suspend;
 		}
 		codec->patch_ops.pre_resume = stac92xx_pre_resume;
 		codec->patch_ops.check_power_status =
@@ -5983,8 +5969,6 @@ again:
 		} else {
 			codec->patch_ops.set_power_state =
 					stac92xx_set_power_state;
-			codec->patch_ops.post_suspend =
-					stac92xx_post_suspend;
 		}
 		codec->patch_ops.pre_resume = stac92xx_pre_resume;
 		codec->patch_ops.check_power_status =
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index bee3c94f58b0..d1fcc816ce97 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -1,6 +1,6 @@
 config SND_ATMEL_SOC
 	tristate "SoC Audio for the Atmel System-on-Chip"
-	depends on ARCH_AT91 || AVR32
+	depends on ARCH_AT91
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the ATMEL SSC interface. You will also need
@@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731
 	  Say Y if you want to add support for SoC audio on WM8731-based
 	  AT91sam9g20 evaluation board.
 
-config SND_AT32_SOC_PLAYPAQ
-        tristate "SoC Audio support for PlayPaq with WM8510"
-        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
-        select SND_ATMEL_SOC_SSC
-        select SND_SOC_WM8510
-        help
-          Say Y or M here if you want to add support for SoC audio
-          on the LRS PlayPaq.
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
-        bool "Run CODEC on PlayPaq in slave mode"
-        depends on SND_AT32_SOC_PLAYPAQ
-        default n
-        help
-          Say Y if you want to run with the AT32 SSC generating the BCLK
-          and FRAME signals on the PlayPaq.  Unless you want to play
-          with the AT32 as the SSC master, you probably want to say N here,
-          as this will give you better sound quality.
-
 config SND_AT91_SOC_AFEB9260
 	tristate "SoC Audio support for AFEB9260 board"
 	depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index e7ea56bd5f82..a5c0bf19da78 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
 # AT91 Machine Support
 snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
 
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
 obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
deleted file mode 100644
index 73ae99ad4578..000000000000
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ /dev/null
@@ -1,473 +0,0 @@
-/* sound/soc/at32/playpaq_wm8510.c
- * ASoC machine driver for PlayPaq using WM8510 codec
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c
- *
- * NOTE: If you don't have the AT32 enhanced portmux configured (which
- * isn't currently in the mainline or Atmel patched kernel), you will
- * need to set the MCLK pin (PA30) to peripheral A in your board initialization
- * code.  Something like:
- *	at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0);
- *
- */
-
-/* #define DEBUG */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/errno.h>
-#include <linux/clk.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/at32ap700x.h>
-#include <mach/portmux.h>
-
-#include "../codecs/wm8510.h"
-#include "atmel-pcm.h"
-#include "atmel_ssc_dai.h"
-
-
-/*-------------------------------------------------------------------------*\
- * constants
-\*-------------------------------------------------------------------------*/
-#define MCLK_PIN		GPIO_PIN_PA(30)
-#define MCLK_PERIPH		GPIO_PERIPH_A
-
-
-/*-------------------------------------------------------------------------*\
- * data types
-\*-------------------------------------------------------------------------*/
-/* SSC clocking data */
-struct ssc_clock_data {
-	/* CMR div */
-	unsigned int cmr_div;
-
-	/* Frame period (as needed by xCMR.PERIOD) */
-	unsigned int period;
-
-	/* The SSC clock rate these settings where calculated for */
-	unsigned long ssc_rate;
-};
-
-
-/*-------------------------------------------------------------------------*\
- * module data
-\*-------------------------------------------------------------------------*/
-static struct clk *_gclk0;
-static struct clk *_pll0;
-
-#define CODEC_CLK (_gclk0)
-
-
-/*-------------------------------------------------------------------------*\
- * Sound SOC operations
-\*-------------------------------------------------------------------------*/
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
-	struct snd_pcm_hw_params *params,
-	struct snd_soc_dai *cpu_dai)
-{
-	struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
-	struct ssc_device *ssc = ssc_p->ssc;
-	struct ssc_clock_data cd;
-	unsigned int rate, width_bits, channels;
-	unsigned int bitrate, ssc_div;
-	unsigned actual_rate;
-
-
-	/*
-	 * Figure out required bitrate
-	 */
-	rate = params_rate(params);
-	channels = params_channels(params);
-	width_bits = snd_pcm_format_physical_width(params_format(params));
-	bitrate = rate * width_bits * channels;
-
-
-	/*
-	 * Figure out required SSC divider and period for required bitrate
-	 */
-	cd.ssc_rate = clk_get_rate(ssc->clk);
-	ssc_div = cd.ssc_rate / bitrate;
-	cd.cmr_div = ssc_div / 2;
-	if (ssc_div & 1) {
-		/* round cmr_div up */
-		cd.cmr_div++;
-	}
-	cd.period = width_bits - 1;
-
-
-	/*
-	 * Find actual rate, compare to requested rate
-	 */
-	actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
-	pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
-		 rate, actual_rate);
-
-
-	return cd;
-}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-
-static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
-				    struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
-	struct ssc_device *ssc = ssc_p->ssc;
-	unsigned int pll_out = 0, bclk = 0, mclk_div = 0;
-	int ret;
-
-
-	/* Due to difficulties with getting the correct clocks from the AT32's
-	 * PLL0, we're going to let the CODEC be in charge of all the clocks
-	 */
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-#else
-	struct ssc_clock_data cd;
-	const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBS_CFS);
-#endif
-
-	if (ssc == NULL) {
-		pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n");
-		return -EINVAL;
-	}
-
-
-	/*
-	 * Figure out PLL and BCLK dividers for WM8510
-	 */
-	switch (params_rate(params)) {
-	case 48000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_2;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 44100:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_2;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 22050:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_4;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 16000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_6;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 11025:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_8;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 8000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_12;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	default:
-		pr_warning("playpaq_wm8510: Unsupported sample rate %d\n",
-			   params_rate(params));
-		return -EINVAL;
-	}
-
-
-	/*
-	 * set CPU and CODEC DAI configuration
-	 */
-	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CODEC DAI format (%d)\n",
-			   ret);
-		return ret;
-	}
-	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CPU DAI format (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	/*
-	 * Set CPU clock configuration
-	 */
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai);
-	pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n",
-		 cd.cmr_div, cd.period);
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n",
-			   ret);
-		return ret;
-	}
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD,
-					  cd.period);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CPU transmit period (%d)\n",
-			   ret);
-		return ret;
-	}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-	/*
-	 * Set CODEC clock configuration
-	 */
-	pr_debug("playpaq_wm8510: "
-		 "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n",
-		 clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div);
-
-
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk);
-	if (ret < 0) {
-		pr_warning
-		    ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n",
-		     ret);
-		return ret;
-	}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-	ret = snd_soc_dai_set_pll(codec_dai, 0, 0,
-					 clk_get_rate(CODEC_CLK), pll_out);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	return 0;
-}
-
-
-
-static struct snd_soc_ops playpaq_wm8510_ops = {
-	.hw_params = playpaq_wm8510_hw_params,
-};
-
-
-
-static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
-	SND_SOC_DAPM_MIC("Int Mic", NULL),
-	SND_SOC_DAPM_SPK("Ext Spk", NULL),
-};
-
-
-
-static const struct snd_soc_dapm_route intercon[] = {
-	/* speaker connected to SPKOUT */
-	{"Ext Spk", NULL, "SPKOUTP"},
-	{"Ext Spk", NULL, "SPKOUTN"},
-
-	{"Mic Bias", NULL, "Int Mic"},
-	{"MICN", NULL, "Mic Bias"},
-	{"MICP", NULL, "Mic Bias"},
-};
-
-
-
-static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int i;
-
-	/*
-	 * Add DAPM widgets
-	 */
-	for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
-		snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
-
-
-
-	/*
-	 * Setup audio path interconnects
-	 */
-	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
-
-
-	/* always connected pins */
-	snd_soc_dapm_enable_pin(dapm, "Int Mic");
-	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
-
-
-
-	/* Make CSB show PLL rate */
-	snd_soc_dai_set_clkdiv(rtd->codec_dai, WM8510_OPCLKDIV,
-				       WM8510_OPCLKDIV_1 | 4);
-
-	return 0;
-}
-
-
-
-static struct snd_soc_dai_link playpaq_wm8510_dai = {
-	.name = "WM8510",
-	.stream_name = "WM8510 PCM",
-	.cpu_dai_name= "atmel-ssc-dai.0",
-	.platform_name = "atmel-pcm-audio",
-	.codec_name = "wm8510-codec.0-0x1a",
-	.codec_dai_name = "wm8510-hifi",
-	.init = playpaq_wm8510_init,
-	.ops = &playpaq_wm8510_ops,
-};
-
-
-
-static struct snd_soc_card snd_soc_playpaq = {
-	.name = "LRS_PlayPaq_WM8510",
-	.dai_link = &playpaq_wm8510_dai,
-	.num_links = 1,
-};
-
-static struct platform_device *playpaq_snd_device;
-
-
-static int __init playpaq_asoc_init(void)
-{
-	int ret = 0;
-
-	/*
-	 * Configure MCLK for WM8510
-	 */
-	_gclk0 = clk_get(NULL, "gclk0");
-	if (IS_ERR(_gclk0)) {
-		_gclk0 = NULL;
-		ret = PTR_ERR(_gclk0);
-		goto err_gclk0;
-	}
-	_pll0 = clk_get(NULL, "pll0");
-	if (IS_ERR(_pll0)) {
-		_pll0 = NULL;
-		ret = PTR_ERR(_pll0);
-		goto err_pll0;
-	}
-	ret = clk_set_parent(_gclk0, _pll0);
-	if (ret) {
-		pr_warning("snd-soc-playpaq: "
-			   "Failed to set PLL0 as parent for DAC clock\n");
-		goto err_set_clk;
-	}
-	clk_set_rate(CODEC_CLK, 12000000);
-	clk_enable(CODEC_CLK);
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
-	at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0);
-#endif
-
-
-	/*
-	 * Create and register platform device
-	 */
-	playpaq_snd_device = platform_device_alloc("soc-audio", 0);
-	if (playpaq_snd_device == NULL) {
-		ret = -ENOMEM;
-		goto err_device_alloc;
-	}
-
-	platform_set_drvdata(playpaq_snd_device, &snd_soc_playpaq);
-
-	ret = platform_device_add(playpaq_snd_device);
-	if (ret) {
-		pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n",
-			   ret);
-		goto err_device_add;
-	}
-
-	return 0;
-
-
-err_device_add:
-	if (playpaq_snd_device != NULL) {
-		platform_device_put(playpaq_snd_device);
-		playpaq_snd_device = NULL;
-	}
-err_device_alloc:
-err_set_clk:
-	if (_pll0 != NULL) {
-		clk_put(_pll0);
-		_pll0 = NULL;
-	}
-err_pll0:
-	if (_gclk0 != NULL) {
-		clk_put(_gclk0);
-		_gclk0 = NULL;
-	}
-	return ret;
-}
-
-
-static void __exit playpaq_asoc_exit(void)
-{
-	if (_gclk0 != NULL) {
-		clk_put(_gclk0);
-		_gclk0 = NULL;
-	}
-	if (_pll0 != NULL) {
-		clk_put(_pll0);
-		_pll0 = NULL;
-	}
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
-	at32_free_pin(MCLK_PIN);
-#endif
-
-	platform_device_unregister(playpaq_snd_device);
-	playpaq_snd_device = NULL;
-}
-
-module_init(playpaq_asoc_init);
-module_exit(playpaq_asoc_exit);
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 444747f0db26..dd7be0dbbc58 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -34,7 +34,7 @@
 
 #define AD1836_ADC_CTRL2               13
 #define AD1836_ADC_WORD_LEN_MASK       0x30
-#define AD1836_ADC_WORD_OFFSET         5
+#define AD1836_ADC_WORD_OFFSET         4
 #define AD1836_ADC_SERFMT_MASK         (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1f237ecec2a..73f46eb459f1 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
 static int cs4270_soc_resume(struct snd_soc_codec *codec)
 {
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
-	struct i2c_client *i2c_client = to_i2c_client(codec->dev);
 	int reg;
 
 	regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
@@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
 	ndelay(500);
 
 	/* first restore the entire register cache ... */
-	for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
-		u8 val = snd_soc_read(codec, reg);
-
-		if (i2c_smbus_write_byte_data(i2c_client, reg, val)) {
-			dev_err(codec->dev, "i2c write failed\n");
-			return -EIO;
-		}
-	}
+	snd_soc_cache_sync(codec);
 
 	/* ... then disable the power-down bits */
 	reg = snd_soc_read(codec, CS4270_PWRCTL);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 8c3c8205d19e..1ee66361f61b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
 	.probe =	cs42l51_probe,
-	.reg_cache_size = CS42L51_NUMREGS,
+	.reg_cache_size = CS42L51_NUMREGS + 1,
 	.reg_word_size = sizeof(u8),
 };
 
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 9e7e964a5fa3..dcf6f2a1600a 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
 	unsigned int mask = mc->max;
 	unsigned int val = (ucontrol->value.integer.value[0] & mask);
 	unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
-	unsigned int change = 1;
+	unsigned int change = 0;
 
-	if (((max9877_regs[reg] >> shift) & mask) == val)
-		change = 0;
+	if (((max9877_regs[reg] >> shift) & mask) != val)
+		change = 1;
 
-	if (((max9877_regs[reg2] >> shift) & mask) == val2)
-		change = 0;
+	if (((max9877_regs[reg2] >> shift) & mask) != val2)
+		change = 1;
 
 	if (change) {
 		max9877_regs[reg] &= ~(mask << shift);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 9c982e47eb99..6c2988549003 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 	bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT;
 
 	lrclk = bclk_rate / params_rate(params);
+	if (!lrclk) {
+		dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n",
+			bclk_rate);
+		return -EINVAL;
+	}
 	dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
 		lrclk, bclk_rate / lrclk);
 
@@ -3178,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		switch (wm8994->revision) {
 		case 0:
 		case 1:
+		case 2:
+		case 3:
 			wm8994->hubs.dcs_codes_l = -9;
 			wm8994->hubs.dcs_codes_r = -5;
 			break;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 31af405bda84..ae49f1c78c6d 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	}
 
 	if (strcasecmp(sprop, "i2s-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 
@@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 		}
 		machine_data->clk_frequency = be32_to_cpup(iprop);
 	} else if (strcasecmp(sprop, "i2s-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "lj-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "lj-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "rj-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "rj-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "ac97-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "ac97-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else {
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index f75e43997d5b..ad9ac42522e2 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -9,6 +9,7 @@
 
 #include "../codecs/wm8994.h"
 #include <sound/pcm_params.h>
+#include <linux/module.h>
 
  /*
   * Default CFG switch settings to use this driver:
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 85bf541a771d..4b8e35410eb1 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card)
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC");
-	snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input");
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a5d3685a5d38..a25fa63ce9a2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev)
 	struct snd_soc_card *card = dev_get_drvdata(dev);
 	int i, ac97_control = 0;
 
+	/* If the initialization of this soc device failed, there is no codec
+	 * associated with it. Just bail out in this case.
+	 */
+	if (list_empty(&card->codec_dev_list))
+		return 0;
+
 	/* AC97 devices might have other drivers hanging off them so
 	 * need to resume immediately.  Other drivers don't have that
 	 * problem and may take a substantial amount of time to resume
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index b61945f3af9e..32d2a21f2e3b 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1633,6 +1633,37 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* Roland GAIA SH-01 */
+	USB_DEVICE(0x0582, 0x0111),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Roland",
+		.product_name = "GAIA",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = &(const struct snd_usb_midi_endpoint_info) {
+				.out_cables = 0x0003,
+				.in_cables  = 0x0003
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
+{
 	USB_DEVICE(0x0582, 0x0113),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		/* .vendor_name = "BOSS", */