Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
  ALSA: ASoC codec: remove unused #include <version.h>
  ALSA: ASoC: update email address for Liam Girdwood
  ALSA: hda: corrected invalid mixer values
  ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
  ALSA: ASoC: Add destination and source port for DMA on OMAP1
  ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
  ALSA: ASoC: Fix build of GTA01 audio driver
  ALSA: ASoC: Add widgets before setting endpoints on GTA01
  ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
  ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
  ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
  ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
  ALSA: ASoC: Make TLV320AIC26 user-visible
  ALSA: ASoC - clean up Kconfig for TLV320AIC2
  ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
  ALSA: ASoC: Implement WM8510 bias level control
  ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
  ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
  ALSA: ASoC: Add WM8510 SPI support
  ALSA: ASoC: Add WM8753 SPI support
  ...

55 files changed, 2036 insertions, 601 deletions

diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c
index b63839e8f9bd..456a1b4d7832 100644
--- a/sound/oss/ac97_codec.c
+++ b/sound/oss/ac97_codec.c
@@ -30,7 +30,7 @@
  **************************************************************************
  *
  * History
- * May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
  *	Removed non existant WM9700
  *	Added support for WM9705, WM9708, WM9709, WM9710, WM9711
  *	WM9712 and WM9717
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 6ce3cbe98a6a..6e831aff1bd0 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97)
 }
 
 /*
- * May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
  *  removed broken wolfson00 patch.
  *  added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717.
  */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c461baa83c2a..c59065513118 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -322,8 +322,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = {
 	0x1a, 0x1b
 };
 
-static hda_nid_t stac92hd71bxx_dmux_nids[1] = {
-	0x1c,
+static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
+	0x1c, 0x1d,
 };
 
 static hda_nid_t stac92hd71bxx_smux_nids[2] = {
@@ -861,20 +861,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* connect headphone jack to dac1 */
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
 	/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 };
 
-#define HD_DISABLE_PORTF 3
+#define HD_DISABLE_PORTF 2
 static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 	/* start of config #1 */
 
 	/* connect port 0f to audio mixer */
 	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
 	/* unmute right and left channels for node 0x0f */
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* start of config #2 */
@@ -883,10 +881,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* connect headphone jack to dac1 */
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* connect port 0d to audio mixer */
-	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2},
-	/* unmute dac0 input in audio mixer */
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
 	/* unmute right and left channels for nodes 0x0a, 0xd */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -1107,6 +1101,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 
 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	STAC_INPUT_SOURCE(2),
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
@@ -1119,8 +1114,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
 	*/
 
-	HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT),
+
+	HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT),
+
+	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),
+
+	HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -1649,7 +1653,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 
 static unsigned int ref92hd71bxx_pin_configs[11] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
-	0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
+	0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
 	0x90a000f0, 0x01452050, 0x01452050,
 };
 
@@ -3000,7 +3004,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
 
 /* labels for amp mux outputs */
 static const char *stac92xx_amp_labels[3] = {
-	"Front Microphone", "Microphone", "Line In"
+	"Front Microphone", "Microphone", "Line In",
 };
 
 /* create amp out controls mux on capable codecs */
@@ -4327,6 +4331,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = {
 #endif
 };
 
+static struct hda_input_mux stac92hd71bxx_dmux = {
+	.num_items = 4,
+	.items = {
+		{ "Analog Inputs", 0x00 },
+		{ "Mixer", 0x01 },
+		{ "Digital Mic 1", 0x02 },
+		{ "Digital Mic 2", 0x03 },
+	}
+};
+
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
@@ -4341,6 +4355,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 	spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
 	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
 	spec->pin_nids = stac92hd71bxx_pin_nids;
+	memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
+			sizeof(stac92hd71bxx_dmux));
 	spec->board_config = snd_hda_check_board_config(codec,
 							STAC_92HD71BXX_MODELS,
 							stac92hd71bxx_models,
@@ -4392,6 +4408,7 @@ again:
 		/* no output amps */
 		spec->num_pwrs = 0;
 		spec->mixer = stac92hd71bxx_analog_mixer;
+		spec->dinput_mux = &spec->private_dimux;
 
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
@@ -4409,12 +4426,13 @@ again:
 		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
+		spec->dinput_mux = &spec->private_dimux;
 		spec->mixer = stac92hd71bxx_analog_mixer;
 		spec->init = stac92hd71bxx_analog_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 	}
 
-	spec->aloopback_mask = 0x20;
+	spec->aloopback_mask = 0x50;
 	spec->aloopback_shift = 0;
 
 	if (spec->board_config > STAC_92HD71BXX_REF) {
@@ -4456,6 +4474,10 @@ again:
 	spec->multiout.num_dacs = 1;
 	spec->multiout.hp_nid = 0x11;
 	spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
+	if (spec->dinput_mux)
+		spec->private_dimux.num_items +=
+			spec->num_dmics -
+				(ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);
 
 	err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
 	if (!err) {
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
index 905186502e00..85a883299c2e 100644
--- a/sound/soc/at91/Kconfig
+++ b/sound/soc/at91/Kconfig
@@ -8,20 +8,3 @@ config SND_AT91_SOC
 
 config SND_AT91_SOC_SSC
 	tristate
-
-config SND_AT91_SOC_ETI_B1_WM8731
-	tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards"
-	depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
-	select SND_AT91_SOC_SSC
-	select SND_SOC_WM8731
-	help
-	  Say Y if you want to add support for SoC audio on WM8731-based
-	  Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.
-
-config SND_AT91_SOC_ETI_SLAVE
-	bool "Run codec in slave Mode on Endrelia boards"
-	depends on SND_AT91_SOC_ETI_B1_WM8731
-	default n
-	help
-	  Say Y if you want to run with the AT91 SSC generating the BCLK
-	  and LRC signals on Endrelia boards.
diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile
index f23da17cc328..b817f11df286 100644
--- a/sound/soc/at91/Makefile
+++ b/sound/soc/at91/Makefile
@@ -4,8 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o
 
 obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
 obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
-
-# AT91 Machine Support
-snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
-
-obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
index a5b1a79ebffb..1b61cc461261 100644
--- a/sound/soc/at91/at91-ssc.c
+++ b/sound/soc/at91/at91-ssc.c
@@ -5,7 +5,7 @@
  *         Endrelia Technologies Inc.
  *
  * Based on pxa2xx Platform drivers by
- * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
deleted file mode 100644
index 684781e4088b..000000000000
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ /dev/null
@@ -1,349 +0,0 @@
-/*
- * eti_b1_wm8731  --  SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
- *
- * Author:	Frank Mandarino <fmandarino@endrelia.com>
- *		Endrelia Technologies Inc.
- * Created:	Mar 29, 2006
- *
- * Based on corgi.c by:
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
- *          Richard Purdie <richard@openedhand.com>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- *
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.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/soc.h>
-#include <sound/soc-dapm.h>
-
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-
-#include "../codecs/wm8731.h"
-#include "at91-pcm.h"
-#include "at91-ssc.h"
-
-#if 0
-#define	DBG(x...)	printk(KERN_INFO "eti_b1_wm8731: " x)
-#else
-#define	DBG(x...)
-#endif
-
-static struct clk *pck1_clk;
-static struct clk *pllb_clk;
-
-
-static int eti_b1_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	int ret;
-
-	/* cpu clock is the AT91 master clock sent to the SSC */
-	ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
-		60000000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	/* codec system clock is supplied by PCK1, set to 12MHz */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
-		12000000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	/* Start PCK1 clock. */
-	clk_enable(pck1_clk);
-	DBG("pck1 started\n");
-
-	return 0;
-}
-
-static void eti_b1_shutdown(struct snd_pcm_substream *substream)
-{
-	/* Stop PCK1 clock. */
-	clk_disable(pck1_clk);
-	DBG("pck1 stopped\n");
-}
-
-static int eti_b1_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->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	int ret;
-
-#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
-	unsigned int rate;
-	int cmr_div, period;
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/*
-	 * The SSC clock dividers depend on the sample rate.  The CMR.DIV
-	 * field divides the system master clock MCK to drive the SSC TK
-	 * signal which provides the codec BCLK.  The TCMR.PERIOD and
-	 * RCMR.PERIOD fields further divide the BCLK signal to drive
-	 * the SSC TF and RF signals which provide the codec DACLRC and
-	 * ADCLRC clocks.
-	 *
-	 * The dividers were determined through trial and error, where a
-	 * CMR.DIV value is chosen such that the resulting BCLK value is
-	 * divisible, or almost divisible, by (2 * sample rate), and then
-	 * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
-	 */
-	rate = params_rate(params);
-
-	switch (rate) {
-	case 8000:
-		cmr_div = 25;	/* BCLK = 60MHz/(2*25) = 1.2MHz */
-		period = 74;	/* LRC = BCLK/(2*(74+1)) = 8000Hz */
-		break;
-	case 32000:
-		cmr_div = 7;	/* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
-		period = 66;	/* LRC = BCLK/(2*(66+1)) = 31982.942Hz */
-		break;
-	case 48000:
-		cmr_div = 13;	/* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */
-		period = 23;	/* LRC = BCLK/(2*(23+1)) = 48076.923Hz */
-		break;
-	default:
-		printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate);
-		return -EINVAL;
-	}
-
-	/* set the MCK divider for BCLK */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
-	if (ret < 0)
-		return ret;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		/* set the BCLK divider for DACLRC */
-		ret = snd_soc_dai_set_clkdiv(cpu_dai,
-						AT91SSC_TCMR_PERIOD, period);
-	} else {
-		/* set the BCLK divider for ADCLRC */
-		ret = snd_soc_dai_set_clkdiv(cpu_dai,
-						AT91SSC_RCMR_PERIOD, period);
-	}
-	if (ret < 0)
-		return ret;
-
-#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
-	/*
-	 * Codec in Master Mode.
-	 */
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
-
-	return 0;
-}
-
-static struct snd_soc_ops eti_b1_ops = {
-	.startup = eti_b1_startup,
-	.hw_params = eti_b1_hw_params,
-	.shutdown = eti_b1_shutdown,
-};
-
-
-static const struct snd_soc_dapm_widget eti_b1_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 LHPOUT */
-	{"Ext Spk", NULL, "LHPOUT"},
-
-	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
-	{"MICIN", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Int Mic"},
-};
-
-/*
- * Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
- */
-static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
-{
-	DBG("eti_b1_wm8731_init() called\n");
-
-	/* Add specific widgets */
-	snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets,
-				  ARRAY_SIZE(eti_b1_dapm_widgets));
-
-	/* Set up specific audio path interconnects */
-	snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon));
-
-	/* not connected */
-	snd_soc_dapm_disable_pin(codec, "RLINEIN");
-	snd_soc_dapm_disable_pin(codec, "LLINEIN");
-
-	/* always connected */
-	snd_soc_dapm_enable_pin(codec, "Int Mic");
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
-
-	snd_soc_dapm_sync(codec);
-
-	return 0;
-}
-
-static struct snd_soc_dai_link eti_b1_dai = {
-	.name = "WM8731",
-	.stream_name = "WM8731 PCM",
-	.cpu_dai = &at91_ssc_dai[1],
-	.codec_dai = &wm8731_dai,
-	.init = eti_b1_wm8731_init,
-	.ops = &eti_b1_ops,
-};
-
-static struct snd_soc_machine snd_soc_machine_eti_b1 = {
-	.name = "ETI_B1_WM8731",
-	.dai_link = &eti_b1_dai,
-	.num_links = 1,
-};
-
-static struct wm8731_setup_data eti_b1_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1a,
-};
-
-static struct snd_soc_device eti_b1_snd_devdata = {
-	.machine = &snd_soc_machine_eti_b1,
-	.platform = &at91_soc_platform,
-	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &eti_b1_wm8731_setup,
-};
-
-static struct platform_device *eti_b1_snd_device;
-
-static int __init eti_b1_init(void)
-{
-	int ret;
-	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
-
-	if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
-		DBG("SSC1 memory region is busy\n");
-		return -EBUSY;
-	}
-
-	ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K);
-	if (!ssc->base) {
-		DBG("SSC1 memory ioremap failed\n");
-		ret = -ENOMEM;
-		goto fail_release_mem;
-	}
-
-	ssc->pid = AT91RM9200_ID_SSC1;
-
-	eti_b1_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!eti_b1_snd_device) {
-		DBG("platform device allocation failed\n");
-		ret = -ENOMEM;
-		goto fail_io_unmap;
-	}
-
-	platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata);
-	eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev;
-
-	ret = platform_device_add(eti_b1_snd_device);
-	if (ret) {
-		DBG("platform device add failed\n");
-		platform_device_put(eti_b1_snd_device);
-		goto fail_io_unmap;
-	}
-
-	at91_set_A_periph(AT91_PIN_PB6, 0);	/* TF1 */
-	at91_set_A_periph(AT91_PIN_PB7, 0);	/* TK1 */
-	at91_set_A_periph(AT91_PIN_PB8, 0);	/* TD1 */
-	at91_set_A_periph(AT91_PIN_PB9, 0);	/* RD1 */
-/*	at91_set_A_periph(AT91_PIN_PB10, 0);*/	/* RK1 */
-	at91_set_A_periph(AT91_PIN_PB11, 0);	/* RF1 */
-
-	/*
-	 * Set PCK1 parent to PLLB and its rate to 12 Mhz.
-	 */
-	pllb_clk = clk_get(NULL, "pllb");
-	pck1_clk = clk_get(NULL, "pck1");
-
-	clk_set_parent(pck1_clk, pllb_clk);
-	clk_set_rate(pck1_clk, 12000000);
-
-	DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk));
-
-	/* assign the GPIO pin to PCK1 */
-	at91_set_B_periph(AT91_PIN_PA24, 0);
-
-#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
-	printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n");
-#else
-	printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n");
-#endif
-	return ret;
-
-fail_io_unmap:
-	iounmap(ssc->base);
-fail_release_mem:
-	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
-	return ret;
-}
-
-static void __exit eti_b1_exit(void)
-{
-	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
-
-	clk_put(pck1_clk);
-	clk_put(pllb_clk);
-
-	platform_device_unregister(eti_b1_snd_device);
-
-	iounmap(ssc->base);
-	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
-}
-
-module_init(eti_b1_init);
-module_exit(eti_b1_exit);
-
-/* Module information */
-MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
-MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index f98331d099e7..dc006206f622 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602
 	help
 	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
 
+config SND_BF5XX_SOC_AD73311
+	tristate "SoC AD73311 Audio support for Blackfin"
+	depends on SND_BF5XX_I2S
+	select SND_BF5XX_SOC_I2S
+	select SND_SOC_AD73311
+	help
+	  Say Y if you want to add support for AD73311 codec on Blackfin.
+
+config SND_BFIN_AD73311_SE
+	int "PF pin for AD73311L Chip Select"
+	depends on SND_BF5XX_SOC_AD73311
+	default 4
+	help
+	  Enter the GPIO used to control AD73311's SE pin. Acceptable
+	  values are 0 to 7
+
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
 	depends on BLACKFIN && SND_SOC
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 9ea8bd9e0ba3..97bb37a6359c 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
 # Blackfin Machine Support
 snd-ad1980-objs := bf5xx-ad1980.o
 snd-ssm2602-objs := bf5xx-ssm2602.o
-
+snd-ad73311-objs := bf5xx-ad73311.o
 
 obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
 obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
+obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 51f4907c4831..25e50d2ea1ec 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
 		sport->tx_pos += runtime->period_size;
 		if (sport->tx_pos >= runtime->buffer_size)
 			sport->tx_pos %= runtime->buffer_size;
+		sport->tx_delay_pos = sport->tx_pos;
 	} else {