Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "A relative calm release at this time with a flat diffstat.  The only
  significant change in the ALSA core side is the support for more than
  32 card instances, configurable via kconfig.  Other than that, in both
  ASoC and other parts, mostly some improvements and fixes on the driver
  side.

   - hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes
   - hda: Haswell HDMI audio fixes, runtime PM improvements
   - hda: Intel BayTrail support, ALC5505 DSP support
   - es1968: MediaForte M56VAP support
   - usb-audio: Improved support for Yamaha/Roland devices
   - usb-audio: M2Tech hiFace, Audio Advantage Micro II support
   - hdspm: wordclock fixes
   - ASoC: Pending fixes for WM8962
   - ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500
   - ASoC: Generalisation of the Bluetooth and HDMI stub drivers
   - ASoC: SSM2518 and RT5640 codec drivers.
   - ASoC: Tegra CPUs with RT5640 machine driver
   - ASoC: AC'97 refactoring bug fixes
   - ASoC: ADAU1701 driver fixes
   - Clean up of *_set_drvdata() in a wide range of drivers"

* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (284 commits)
  ALSA: vmaster: Fix the regression of missing vmaster hook call
  ALSA: hda - Add Dell SSID to support Headset Mic recording
  ASoC: adau1701: remove control_data assignment
  ASoC: adau1701: more direct regmap usage
  ASoC: ac97: fixup multi-platform AC'97 module build failure
  ASoC: pxa2xx: fixup multi-platform AC'97 build failures
  ASoC: tegra20-ac97: Remove unused variable
  ASoC: tegra20-ac97: Remove duplicate error message
  ALSA: usb-audio: Add Audio Advantage Micro II
  ASoC: tas5086: fix Mid-Z implementation
  ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
  ALSA: Replace the magic number 44 with const
  ALSA: hda - Fix the max length of control name in generic parser
  ALSA: hda - Guess what, it's two more Dell headset mic quirks
  ALSA: hda - Yet another Dell headset mic quirk
  ALSA: hda - Add support for ALC5505 DSP power-save mode
  ASoC: mfld: Remove unused variable
  ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
  ALSA: usb-audio: claim autodetected PCM interfaces all at once
  ALSA: usb-audio: remove superfluous Roland quirks
  ...

9 files changed, 12 insertions, 413 deletions

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 4d2e46fae77c..b35809467547 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -130,26 +130,6 @@ config SND_PXA2XX_SOC_PALM27X
 	  Say Y if you want to add support for SoC audio on
 	  Palm T|X, T5, E2 or LifeDrive handheld computer.
 
-config SND_SOC_SAARB
-	tristate "SoC Audio support for Marvell Saarb"
-	depends on SND_PXA2XX_SOC && MACH_SAARB
-	select MFD_88PM860X
-	select SND_PXA_SOC_SSP
-	select SND_SOC_88PM860X
-	help
-	  Say Y if you want to add support for SoC audio on the
-	  Marvell Saarb reference platform.
-
-config SND_SOC_TAVOREVB3
-	tristate "SoC Audio support for Marvell Tavor EVB3"
-	depends on SND_PXA2XX_SOC && MACH_TAVOREVB3
-	select MFD_88PM860X
-	select SND_PXA_SOC_SSP
-	select SND_SOC_88PM860X
-	help
-	  Say Y if you want to add support for SoC audio on the
-	  Marvell Saarb reference platform.
-
 config SND_PXA910_SOC
 	tristate "SoC Audio for Marvell PXA910 chip"
 	depends on ARCH_MMP && SND
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index d8a265d2d5d7..2cff67b61dc3 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -23,8 +23,6 @@ snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 snd-soc-palm27x-objs := palm27x.o
-snd-soc-saarb-objs := saarb.o
-snd-soc-tavorevb3-objs := tavorevb3.o
 snd-soc-zylonite-objs := zylonite.o
 snd-soc-hx4700-objs := hx4700.o
 snd-soc-magician-objs := magician.o
@@ -48,8 +46,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
-obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o
-obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
 obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
 obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 349930015264..5d57e071cdf5 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -147,7 +147,7 @@ static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
 		vma->vm_end - vma->vm_start, vma->vm_page_prot);
 }
 
-struct snd_pcm_ops mmp_pcm_ops = {
+static struct snd_pcm_ops mmp_pcm_ops = {
 	.open		= mmp_pcm_open,
 	.close		= snd_dmaengine_pcm_close_release_chan,
 	.ioctl		= snd_pcm_lib_ioctl,
@@ -208,7 +208,7 @@ static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
+static int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_pcm_substream *substream;
 	struct snd_pcm *pcm = rtd->pcm;
@@ -229,7 +229,7 @@ err:
 	return ret;
 }
 
-struct snd_soc_platform_driver mmp_soc_platform = {
+static struct snd_soc_platform_driver mmp_soc_platform = {
 	.ops		= &mmp_pcm_ops,
 	.pcm_new	= mmp_pcm_new,
 	.pcm_free	= mmp_pcm_free_dma_buffers,
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index a64779980177..62142ce367c7 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -388,7 +388,7 @@ static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
 	.set_fmt	= mmp_sspa_set_dai_fmt,
 };
 
-struct snd_soc_dai_driver mmp_sspa_dai = {
+static struct snd_soc_dai_driver mmp_sspa_dai = {
 	.probe = mmp_sspa_probe,
 	.playback = {
 		.channels_min = 1,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 57ea8e6c5488..1475515712e6 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -41,13 +41,12 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
 	pxa2xx_ac97_finish_reset(ac97);
 }
 
-struct snd_ac97_bus_ops soc_ac97_ops = {
+static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.read	= pxa2xx_ac97_read,
 	.write	= pxa2xx_ac97_write,
 	.warm_reset	= pxa2xx_ac97_warm_reset,
 	.reset	= pxa2xx_ac97_cold_reset,
 };
-EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
 	.name			= "AC97 PCM Stereo out",
@@ -239,11 +238,17 @@ static const struct snd_soc_component_driver pxa_ac97_component = {
 
 static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
+	int ret;
+
 	if (pdev->id != -1) {
 		dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
 		return -ENXIO;
 	}
 
+	ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
+	if (ret != 0)
+		return ret;
+
 	/* Punt most of the init to the SoC probe; we may need the machine
 	 * driver to do interesting things with the clocking to get us up
 	 * and running.
@@ -255,6 +260,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
 {
 	snd_soc_unregister_component(&pdev->dev);
+	snd_soc_set_ac97_ops(NULL);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
index eda891e6f31b..a49c21ba3842 100644
--- a/sound/soc/pxa/pxa2xx-ac97.h
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -14,7 +14,4 @@
 #define PXA2XX_DAI_AC97_AUX		1
 #define PXA2XX_DAI_AC97_MIC		2
 
-/* platform data */
-extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
-
 #endif
diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c
deleted file mode 100644
index c34146b776b4..000000000000
--- a/sound/soc/pxa/saarb.c
+++ /dev/null
@@ -1,190 +0,0 @@
-/*
- * saarb.c -- SoC audio for saarb
- *
- * Copyright (C) 2010 Marvell International Ltd.
- * 	Haojian Zhuang <haojian.zhuang@marvell.com>
- *
- * 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.
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <linux/clk.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-
-#include "../codecs/88pm860x-codec.h"
-#include "pxa-ssp.h"
-
-static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd);
-
-static struct platform_device *saarb_snd_device;
-
-static struct snd_soc_jack hs_jack, mic_jack;
-
-static struct snd_soc_jack_pin hs_jack_pins[] = {
-	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
-};
-
-static struct snd_soc_jack_pin mic_jack_pins[] = {
-	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
-};
-
-/* saarb machine dapm widgets */
-static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headphone Stereophone", NULL),
-	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
-	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
-	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
-	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
-};
-
-/* saarb machine audio map */
-static const struct snd_soc_dapm_route saarb_audio_map[] = {
-	{"Headset Stereophone", NULL, "HS1"},
-	{"Headset Stereophone", NULL, "HS2"},
-
-	{"Ext Speaker", NULL, "LSP"},
-	{"Ext Speaker", NULL, "LSN"},
-
-	{"Lineout Out 1", NULL, "LINEOUT1"},
-	{"Lineout Out 2", NULL, "LINEOUT2"},
-
-	{"MIC1P", NULL, "Mic1 Bias"},
-	{"MIC1N", NULL, "Mic1 Bias"},
-	{"Mic1 Bias", NULL, "Ext Mic 1"},
-
-	{"MIC2P", NULL, "Mic1 Bias"},
-	{"MIC2N", NULL, "Mic1 Bias"},
-	{"Mic1 Bias", NULL, "Headset Mic 2"},
-
-	{"MIC3P", NULL, "Mic3 Bias"},
-	{"MIC3N", NULL, "Mic3 Bias"},
-	{"Mic3 Bias", NULL, "Ext Mic 3"},
-};
-
-static int saarb_i2s_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;
-	int width = snd_pcm_format_physical_width(params_format(params));
-	int ret;
-
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
-				     PM860X_CLK_DIR_OUT);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
-
-	return ret;
-}
-
-static struct snd_soc_ops saarb_i2s_ops = {
-	.hw_params	= saarb_i2s_hw_params,
-};
-
-static struct snd_soc_dai_link saarb_dai[] = {
-	{
-		.name		= "88PM860x I2S",
-		.stream_name	= "I2S Audio",
-		.cpu_dai_name	= "pxa-ssp-dai.1",
-		.codec_dai_name	= "88pm860x-i2s",
-		.platform_name	= "pxa-pcm-audio",
-		.codec_name	= "88pm860x-codec",
-		.init		= saarb_pm860x_init,
-		.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM,
-		.ops		= &saarb_i2s_ops,
-	},
-};
-
-static struct snd_soc_card snd_soc_card_saarb = {
-	.name = "Saarb",
-	.owner = THIS_MODULE,
-	.dai_link = saarb_dai,
-	.num_links = ARRAY_SIZE(saarb_dai),
-
-	.dapm_widgets = saarb_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets),
-	.dapm_routes = saarb_audio_map,
-	.num_dapm_routes = ARRAY_SIZE(saarb_audio_map),
-};
-
-static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	/* connected pins */
-	snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
-	snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
-	snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
-	snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
-	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
-
-	/* Headset jack detection */
-	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
-			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
-			&hs_jack);
-	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
-			      hs_jack_pins);
-	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
-			 &mic_jack);
-	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
-			      mic_jack_pins);
-
-	/* headphone, microphone detection & headset short detection */
-	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
-			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
-	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
-	return 0;
-}
-
-static int __init saarb_init(void)
-{
-	int ret;
-
-	if (!machine_is_saarb())
-		return -ENODEV;
-	saarb_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!saarb_snd_device)
-		return -ENOMEM;
-
-	platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb);
-
-	ret = platform_device_add(saarb_snd_device);
-	if (ret)
-		platform_device_put(saarb_snd_device);
-
-	return ret;
-}
-
-static void __exit saarb_exit(void)
-{
-	platform_device_unregister(saarb_snd_device);
-}
-
-module_init(saarb_init);
-module_exit(saarb_exit);
-
-MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
-MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c
deleted file mode 100644
index 8b5ab8f72726..000000000000
--- a/sound/soc/pxa/tavorevb3.c
+++ /dev/null
@@ -1,189 +0,0 @@
-/*
- * tavorevb3.c -- SoC audio for Tavor EVB3
- *
- * Copyright (C) 2010 Marvell International Ltd.
- * 	Haojian Zhuang <haojian.zhuang@marvell.com>
- *
- * 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.
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <linux/clk.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-
-#include "../codecs/88pm860x-codec.h"
-#include "pxa-ssp.h"
-
-static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd);
-
-static struct platform_device *evb3_snd_device;
-
-static struct snd_soc_jack hs_jack, mic_jack;
-
-static struct snd_soc_jack_pin hs_jack_pins[] = {
-	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
-};
-
-static struct snd_soc_jack_pin mic_jack_pins[] = {
-	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
-};
-
-/* tavorevb3 machine dapm widgets */
-static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
-	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
-	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
-	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
-	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
-	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
-};
-
-/* tavorevb3 machine audio map */
-static const struct snd_soc_dapm_route evb3_audio_map[] = {
-	{"Headset Stereophone", NULL, "HS1"},
-	{"Headset Stereophone", NULL, "HS2"},
-
-	{"Ext Speaker", NULL, "LSP"},
-	{"Ext Speaker", NULL, "LSN"},
-
-	{"Lineout Out 1", NULL, "LINEOUT1"},
-	{"Lineout Out 2", NULL, "LINEOUT2"},
-
-	{"MIC1P", NULL, "Mic1 Bias"},
-	{"MIC1N", NULL, "Mic1 Bias"},
-	{"Mic1 Bias", NULL, "Ext Mic 1"},
-
-	{"MIC2P", NULL, "Mic1 Bias"},
-	{"MIC2N", NULL, "Mic1 Bias"},
-	{"Mic1 Bias", NULL, "Headset Mic 2"},
-
-	{"MIC3P", NULL, "Mic3 Bias"},
-	{"MIC3N", NULL, "Mic3 Bias"},
-	{"Mic3 Bias", NULL, "Ext Mic 3"},
-};
-
-static int evb3_i2s_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;
-	int width = snd_pcm_format_physical_width(params_format(params));
-	int ret;
-
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
-				     PM860X_CLK_DIR_OUT);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
-	return ret;
-}
-
-static struct snd_soc_ops evb3_i2s_ops = {
-	.hw_params	= evb3_i2s_hw_params,
-};
-
-static struct snd_soc_dai_link evb3_dai[] = {
-	{
-		.name		= "88PM860x I2S",
-		.stream_name	= "I2S Audio",
-		.cpu_dai_name	= "pxa-ssp-dai.1",
-		.codec_dai_name	= "88pm860x-i2s",
-		.platform_name	= "pxa-pcm-audio",
-		.codec_name	= "88pm860x-codec",
-		.init		= evb3_pm860x_init,
-		.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM,
-		.ops		= &evb3_i2s_ops,
-	},
-};
-
-static struct snd_soc_card snd_soc_card_evb3 = {
-	.name = "Tavor EVB3",
-	.owner = THIS_MODULE,
-	.dai_link = evb3_dai,
-	.num_links = ARRAY_SIZE(evb3_dai),
-
-	.dapm_widgets = evb3_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(evb3_dapm_widgets),
-	.dapm_routes = evb3_audio_map,
-	.num_dapm_routes = ARRAY_SIZE(evb3_audio_map),
-};
-
-static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	/* connected pins */
-	snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
-	snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
-	snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
-	snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
-	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
-
-	/* Headset jack detection */
-	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
-			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
-			&hs_jack);
-	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
-			      hs_jack_pins);
-	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
-			 &mic_jack);
-	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
-			      mic_jack_pins);
-
-	/* headphone, microphone detection & headset short detection */
-	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
-			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
-	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
-	return 0;
-}
-
-static int __init tavorevb3_init(void)
-{
-	int ret;
-
-	if (!machine_is_tavorevb3())
-		return -ENODEV;
-	evb3_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!evb3_snd_device)
-		return -ENOMEM;
-
-	platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3);
-
-	ret = platform_device_add(evb3_snd_device);
-	if (ret)
-		platform_device_put(evb3_snd_device);
-
-	return ret;
-}
-
-static void __exit tavorevb3_exit(void)
-{
-	platform_device_unregister(evb3_snd_device);
-}
-
-module_init(tavorevb3_init);
-module_exit(tavorevb3_exit);
-
-MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
-MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index ceb656695b0f..db8aadf8932d 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -256,7 +256,6 @@ static struct snd_soc_card zylonite = {
 	.resume_pre = &zylonite_resume_pre,
 	.dai_link = zylonite_dai,
 	.num_links = ARRAY_SIZE(zylonite_dai),
-	.owner = THIS_MODULE,
 };
 
 static struct platform_device *zylonite_snd_ac97_device;