ASoC updates

ASoC updates

[Mark Brown]

The following changes since commit 2e2ed79f64e935540b0d148f7e2c3ebefafe133e:
  Vladimir Barinov (1):
        ALSA: Correct Vladimir Barinov's e-mail address

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Of these only the patch from Rob is 2.6.27 material.

Arun KS (1):
      ASoC: Add destination and source port for DMA on OMAP1

Frank Mandarino (1):
      ASoC: Remove references to Endrelia ETI-B1 board

Jonas Bonn (2):
      ASoC: Add widgets before setting endpoints on GTA01
      ASoC: Drop device registration from GTA01 lm4857 driver

Rob Sims (1):
      ASoC: Set correct name for WM8753 rec mixer output

 sound/soc/at91/Kconfig             |   17 --
 sound/soc/at91/Makefile            |    5 -
 sound/soc/at91/eti_b1_wm8731.c     |  349 ------------------------------------
 sound/soc/codecs/wm8753.c          |    4 +-
 sound/soc/omap/omap-pcm.c          |    4 +-
 sound/soc/s3c24xx/neo1973_wm8753.c |   57 +-----
 6 files changed, 14 insertions(+), 422 deletions(-)
 delete mode 100644 sound/soc/at91/eti_b1_wm8731.c

[PATCH 1/5] ASoC: Add destination and source port for DMA on OMAP1

[Mark Brown]

From: Arun KS <arunks@mistralsolutions.com>

Adds destination and source port for dma in platform driver as
required by OMAP1

Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/omap-pcm.c |    4 +++-
 1 files changed, 3 insertions(+), 1 deletions(-)

diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 690bfea..e9084fd 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	prtd->dma_data = dma_data;
 	err = omap_request_dma(dma_data->dma_req, dma_data->name,
 			       omap_pcm_dma_irq, substream, &prtd->dma_ch);
-	if (!cpu_is_omap1510()) {
+	if (!err & !cpu_is_omap1510()) {
 		/*
 		 * Link channel with itself so DMA doesn't need any
 		 * reprogramming while looping the buffer
@@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 		dma_params.src_or_dst_synch	= OMAP_DMA_DST_SYNC;
 		dma_params.src_start		= runtime->dma_addr;
 		dma_params.dst_start		= dma_data->port_addr;
+		dma_params.dst_port		= OMAP_DMA_PORT_MPUI;
 	} else {
 		dma_params.src_amode		= OMAP_DMA_AMODE_CONSTANT;
 		dma_params.dst_amode		= OMAP_DMA_AMODE_POST_INC;
 		dma_params.src_or_dst_synch	= OMAP_DMA_SRC_SYNC;
 		dma_params.src_start		= dma_data->port_addr;
 		dma_params.dst_start		= runtime->dma_addr;
+		dma_params.src_port		= OMAP_DMA_PORT_MPUI;
 	}
 	/*
 	 * Set DMA transfer frame size equal to ALSA period size and frame
-- 
1.5.6.5

[PATCH 2/5] ASoC: Remove references to Endrelia ETI-B1 board

[Mark Brown]

From: Frank Mandarino <fmandarino@endrelia.com>

The ASoC machine drivers for this board were only provided as examples
for the new AT91 ASoC platform driver.  Since the ETI-B1 board is
proprietary and there are other AT91 ASoC machine drivers available,
it makes sense to remove these drivers.

Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/at91/Kconfig         |   17 --
 sound/soc/at91/Makefile        |    5 -
 sound/soc/at91/eti_b1_wm8731.c |  349 ----------------------------------------
 3 files changed, 0 insertions(+), 371 deletions(-)
 delete mode 100644 sound/soc/at91/eti_b1_wm8731.c

diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
index 9051865..85a8832 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 f23da17..b817f11 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/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
deleted file mode 100644
index 684781e..0000000
--- 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");
-- 
1.5.6.5

[PATCH 3/5] ASoC: Add widgets before setting endpoints on GTA01

[Mark Brown]

From: Jonas Bonn <jonas.bonn@gmail.com>

This prevents error messages at startup where the endpoints are being
set before the widgets/controls have even been added.

Signed-off-by: Jonas Bonn <jonas.bonn@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/neo1973_wm8753.c |    6 +++---
 1 files changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 41fc69c..f3ac449 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -517,13 +517,13 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "LINE1");
 	snd_soc_dapm_nc_pin(codec, "LINE2");
 
-	/* set endpoints to default mode */
-	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
-
 	/* Add neo1973 specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
 				  ARRAY_SIZE(wm8753_dapm_widgets));
 
+	/* set endpoints to default mode */
+	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+
 	/* add neo1973 specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
 		err = snd_ctl_add(codec->card,
-- 
1.5.6.5

[PATCH 4/5 2.6.27?] ASoC: Set correct name for WM8753 rec mixer output

[Mark Brown]

From: Rob Sims <openmoko-z@robsims.com>

Rob Sims wrote:

"I can't seem to turn on register 0x17, bit 3 in the sound chip, except
by codec_reg_write; the mixer lacks direct or indirect control.  It
seems there are two names for the output of the rec mixer:
Capture ST Mixer
Playback Mixer

Would the following do the trick?"

I confirm that this solves the audio problems I was having.

Signed-off-by: Jonas Bonn <jonas.bonn@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---

This is a bugfix which could be applied to 2.6.27 - without it the
WM8753 driver will fail to set up some audio paths.

 sound/soc/codecs/wm8753.c |    4 ++--
 1 files changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index b3e1b37..8c4df44 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -583,7 +583,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* out 4 */
 	{"Out4 Mux", "VREF", "VREF"},
-	{"Out4 Mux", "Capture ST", "Capture ST Mixer"},
+	{"Out4 Mux", "Capture ST", "Playback Mixer"},
 	{"Out4 Mux", "LOUT2", "LOUT2"},
 	{"Out 4", NULL, "Out4 Mux"},
 	{"OUT4", NULL, "Out 4"},
@@ -607,7 +607,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	/* Capture Right Mux */
 	{"Capture Right Mux", "PGA", "Right Capture Volume"},
 	{"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"},
-	{"Capture Right Mux", "Sidetone", "Capture ST Mixer"},
+	{"Capture Right Mux", "Sidetone", "Playback Mixer"},
 
 	/* Mono Capture mixer-mux */
 	{"Capture Right Mixer", "Stereo", "Capture Right Mux"},
-- 
1.5.6.5

[PATCH 5/5] ASoC: Drop device registration from GTA01 lm4857 driver

[Mark Brown]

From: Jonas Bonn <jonas.bonn@gmail.com>

Device registration should be handled at the machine level and not
in the driver code itself.  This patch removes the device registration
from the driver code in preparation for moving it to the machine
definition.

[Squashed down two parts to this patch for bisectability - there's also
a third part adding registration of the device to the out of tree GTA01
machine driver -- broonie]

Signed-off-by: Jonas Bonn <jonas.bonn@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/neo1973_wm8753.c |   51 ++++-------------------------------
 1 files changed, 6 insertions(+), 45 deletions(-)

diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index f3ac449..c602674 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -601,6 +601,8 @@ static int lm4857_i2c_probe(struct i2c_client *client,
 {
 	DBG("Entered %s\n", __func__);
 
+	i2c = client;
+
 	lm4857_write_regs();
 	return 0;
 }
@@ -609,6 +611,8 @@ static int lm4857_i2c_remove(struct i2c_client *client)
 {
 	DBG("Entered %s\n", __func__);
 
+	i2c = NULL;
+
 	return 0;
 }
 
@@ -666,48 +670,6 @@ static struct i2c_driver lm4857_i2c_driver = {
 };
 
 static struct platform_device *neo1973_snd_device;
-static struct i2c_client *lm4857_client;
-
-static int __init neo1973_add_lm4857_device(struct platform_device *pdev,
-					    int i2c_bus,
-					    unsigned short i2c_address)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&lm4857_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add lm4857 driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = i2c_address;
-	strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
-	}
-
-	lm4857_client = client;
-	return 0;
-
-err_driver:
-	i2c_del_driver(&lm4857_i2c_driver);
-	return -ENODEV;
-}
 
 static int __init neo1973_init(void)
 {
@@ -728,8 +690,8 @@ static int __init neo1973_init(void)
 		return ret;
 	}
 
-	ret = neo1973_add_lm4857_device(neo1973_snd_device,
-					0, 0x7C);
+	ret = i2c_add_driver(&lm4857_i2c_driver);
+
 	if (ret != 0)
 		platform_device_unregister(neo1973_snd_device);
 
@@ -740,7 +702,6 @@ static void __exit neo1973_exit(void)
 {
 	DBG("Entered %s\n", __func__);
 
-	i2c_unregister_device(lm4857_client);
 	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
 }
-- 
1.5.6.5

Re: ASoC updates

[Takashi Iwai]

At Thu, 2 Oct 2008 11:02:00 +0100,
Mark Brown wrote:
> 
> The following changes since commit 2e2ed79f64e935540b0d148f7e2c3ebefafe133e:
>   Vladimir Barinov (1):
>         ALSA: Correct Vladimir Barinov's e-mail address
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai
> 
> Of these only the patch from Rob is 2.6.27 material.

OK, then I split your commits to two parts.
Rob's patch went into asoc-fixes branch for the next (possible) pull
request, and others to usual topic/asoc branch.  The topic/asoc also
merged asoc-fixes branch, so all are in there as results, too.


Thanks,

Takashi

ASoC updates

[Mark Brown]

The following changes since commit f79f27b5e79b9ae1e39d8376fd2ed956e3e02430:
  Takashi Iwai (1):
        ALSA: ASoC - clean up Kconfig for TLV320AIC2

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Jarkko Nikula (3):
      ASoC: OMAP: Add multilink support to McBSP DAI driver
      ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
      ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver

Mark Brown (1):
      ASoC: Make TLV320AIC26 user-visible

 sound/soc/codecs/Kconfig    |    2 +-
 sound/soc/omap/omap-mcbsp.c |  176 ++++++++++++++++++++++++++++++-------------
 sound/soc/omap/omap-mcbsp.h |   16 +++-
 3 files changed, 134 insertions(+), 60 deletions(-)

Re: ASoC updates

[Takashi Iwai]

At Fri, 10 Oct 2008 15:51:27 +0100,
Mark Brown wrote:
> 
> The following changes since commit f79f27b5e79b9ae1e39d8376fd2ed956e3e02430:
>   Takashi Iwai (1):
>         ALSA: ASoC - clean up Kconfig for TLV320AIC2
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, pulled now.
There was a power cut yesterday ant it took over a day to recover,
and I finally can work again.

BTW, my git tree will be likely totally rebased.  But still it's
better to merge patches right now so that they won't go away.


thanks,

Takashi

[PATCHv2 1/1] Fix a Oops bug in omap soc driver.

[Stanley.Miao]

There will be a Oops or frequent underrun messages when playing music with
omap soc driver, this is because a data region is incorretly sized, other data
region will be overwriten when writing to this data region.

Signed-off-by: Stanley.Miao <stanley.miao@windriver.com>
---
 sound/soc/omap/omap-pcm.c |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index ce580a9..2adfbce 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -233,7 +233,7 @@ static int omap_pcm_open(struct snd_pcm_substream *substream)
 	if (ret < 0)
 		goto out;
 
-	prtd = kzalloc(sizeof(prtd), GFP_KERNEL);
+	prtd = kzalloc(sizeof(struct omap_runtime_data), GFP_KERNEL);
 	if (prtd == NULL) {
 		ret = -ENOMEM;
 		goto out;
-- 
1.5.6.3

Re: [PATCHv2 1/1] Fix a Oops bug in omap soc driver.

[Jarkko Nikula]

On Fri, 19 Dec 2008 22:08:22 +0800
"ext Stanley.Miao" <stanley.miao@windriver.com> wrote:

> There will be a Oops or frequent underrun messages when playing music with
> omap soc driver, this is because a data region is incorretly sized, other data
> region will be overwriten when writing to this data region.
> 
> Signed-off-by: Stanley.Miao <stanley.miao@windriver.com>
> ---
>  sound/soc/omap/omap-pcm.c |    2 +-
>  1 files changed, 1 insertions(+), 1 deletions(-)
> 
> diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
> index ce580a9..2adfbce 100644
> --- a/sound/soc/omap/omap-pcm.c
> +++ b/sound/soc/omap/omap-pcm.c
> @@ -233,7 +233,7 @@ static int omap_pcm_open(struct snd_pcm_substream *substream)
>  	if (ret < 0)
>  		goto out;
>  
> -	prtd = kzalloc(sizeof(prtd), GFP_KERNEL);
> +	prtd = kzalloc(sizeof(struct omap_runtime_data), GFP_KERNEL);

Oh my... Can you change this as

	prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);

like suggested by the CodingStyle and then you can have my ack.

Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>

Re: [alsa-devel] [PATCHv2 1/1] Fix a Oops bug in omap soc driver.

[Takashi Iwai]

At Fri, 19 Dec 2008 16:35:01 +0200,
Jarkko Nikula wrote:
> 
> To: "ext Stanley.Miao" <stanley.miao@windriver.com>
> Cc: alsa-devel@alsa-project.org, linux-omap@vger.kernel.org
> Subject: Re: [alsa-devel] [PATCHv2 1/1] Fix a Oops bug in omap soc driver.
> From: Jarkko Nikula <jarkko.nikula@nokia.com>
> Delivered-To: alsa-devel@alsa-project.org
> Date: Fri, 19 Dec 2008 16:35:01 +0200
> Message-Id: <20081219163501.0e235884.jarkko.nikula@nokia.com>
> In-Reply-To: <4edcd8463d981e4babef82167aa58131539c367f.1229601269.git.stanley.miao@windriver.com>
> Organization: Nokia-M/Helsinki
> Mime-Version: 1.0
> List-Unsubscribe: <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>, 	<mailto:alsa-devel-request@alsa-project.org?subject=unsubscribe>
> List-Archive: <http://mailman.alsa-project.org/pipermail/alsa-devel>
> List-Post: <mailto:alsa-devel@alsa-project.org>
> List-Help: <mailto:alsa-devel-request@alsa-project.org?subject=help>
> List-Subscribe: <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>, 	<mailto:alsa-devel-request@alsa-project.org?subject=subscribe>
> Content-Transfer-Encoding: 7bit
> 
> On Fri, 19 Dec 2008 22:08:22 +0800
> "ext Stanley.Miao" <stanley.miao@windriver.com> wrote:
> 
> > There will be a Oops or frequent underrun messages when playing music with
> > omap soc driver, this is because a data region is incorretly sized, other data
> > region will be overwriten when writing to this data region.
> > 
> > Signed-off-by: Stanley.Miao <stanley.miao@windriver.com>
> > ---
> >  sound/soc/omap/omap-pcm.c |    2 +-
> >  1 files changed, 1 insertions(+), 1 deletions(-)
> > 
> > diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
> > index ce580a9..2adfbce 100644
> > --- a/sound/soc/omap/omap-pcm.c
> > +++ b/sound/soc/omap/omap-pcm.c
> > @@ -233,7 +233,7 @@ static int omap_pcm_open(struct snd_pcm_substream *substream)
> >  	if (ret < 0)
> >  		goto out;
> >  
> > -	prtd = kzalloc(sizeof(prtd), GFP_KERNEL);
> > +	prtd = kzalloc(sizeof(struct omap_runtime_data), GFP_KERNEL);
> 
> Oh my... Can you change this as
> 
> 	prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
> 
> like suggested by the CodingStyle and then you can have my ack.
>
> Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>

Thanks, merged now.  I'm going to push the fixes to the upstream soon.


Takashi

ASoC updates

[Mark Brown]

The following changes since commit 70043058a6968917c00c8d4cff5bfc0e095831ec:
  Takashi Iwai (1):
        Merge branch 'fix/asoc' into topic/asoc

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 for-tiwai

Troy Kisky (8):
      ALSA: ASoC: DaVinci: davinvi-evm, make requests explicit
      ALSA: ASoC: DaVinci: davinci-i2s add comments to explain polarity
      ALSA: ASoC: DaVinci: davinci-i2s clean up
      ALSA: ASoC: DaVinci: davinci-i2s clean up
      ALSA: ASoC: DaVinci: document I2S limitations
      ALSA: ASoC: tlv320aic3x add dsp_a
      ALSA: ASoC: DaVinci: i2s, evm, pass same value to codec and cpu_dai
      ALSA: ASoc: DaVinci: davinci-evm use dsp_b mode

 sound/soc/codecs/tlv320aic3x.c  |    4 +
 sound/soc/codecs/tlv320aic3x.h  |    2 +
 sound/soc/davinci/davinci-evm.c |    8 +-
 sound/soc/davinci/davinci-i2s.c |  158 ++++++++++++++++++++++++--------------
 4 files changed, 110 insertions(+), 62 deletions(-)

Re: ASoC updates

[SUBHRANIL CHOUDHURY]

Dear all,
I am new to ALSA  programming.
Suppose if i have to add some hardware specific feature in the ALSA sound
driver and expose to the application via the ALSA library, how can do that?

Please reply if you have some inputs.
Anything is welcome.
Thanks in advance.

Regards,
Subhranil

Re: ASoC updates

[Sean McNamara]

Hi,

On Sat, Dec 20, 2008 at 9:11 AM, SUBHRANIL CHOUDHURY <subhra85@gmail.com> wrote:
> Dear all,
> I am new to ALSA  programming.
> Suppose if i have to add some hardware specific feature in the ALSA sound
> driver and expose to the application via the ALSA library, how can do that?

The generic interface to do that would be hwdep... in userspace, you
would have to modify libasound2 (alsa-lib) to correctly use the hwdep
interface, as documented here:
http://www.alsa-project.org/alsa-doc/alsa-lib/group___hw_dep.html

>
> Please reply if you have some inputs.

Please look through the documentation for 5 minutes first before
posting to the ML... the updated documentation for 1.0.18 is
maintained at
http://www.alsa-project.org/alsa-doc/alsa-lib/index.html

Thanks,

Sean

> Anything is welcome.
> Thanks in advance.
>
> Regards,
> Subhranil
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>

Re: ASoC updates

[Takashi Iwai]

At Sat, 20 Dec 2008 13:51:32 +0000,
Mark Brown wrote:
> 
> The following changes since commit 70043058a6968917c00c8d4cff5bfc0e095831ec:
>   Takashi Iwai (1):
>         Merge branch 'fix/asoc' into topic/asoc
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 for-tiwai

Thanks, pulled now.


BTW, I'll be off from Dec. 23 for a week.  Since it's not sure whether
I'll have an internet access there, please let me know until the next
Monday if you have urgent updates for 2.6.29.


Takashi

> 
> Troy Kisky (8):
>       ALSA: ASoC: DaVinci: davinvi-evm, make requests explicit
>       ALSA: ASoC: DaVinci: davinci-i2s add comments to explain polarity
>       ALSA: ASoC: DaVinci: davinci-i2s clean up
>       ALSA: ASoC: DaVinci: davinci-i2s clean up
>       ALSA: ASoC: DaVinci: document I2S limitations
>       ALSA: ASoC: tlv320aic3x add dsp_a
>       ALSA: ASoC: DaVinci: i2s, evm, pass same value to codec and cpu_dai
>       ALSA: ASoc: DaVinci: davinci-evm use dsp_b mode
> 
>  sound/soc/codecs/tlv320aic3x.c  |    4 +
>  sound/soc/codecs/tlv320aic3x.h  |    2 +
>  sound/soc/davinci/davinci-evm.c |    8 +-
>  sound/soc/davinci/davinci-i2s.c |  158 ++++++++++++++++++++++++--------------
>  4 files changed, 110 insertions(+), 62 deletions(-)
> 

ASoC updates

[Mark Brown]

The following changes since commit cad9d90ec3f6c497dce2b163046436b6ab5cf6b6:
  Takashi Iwai (1):
        Merge branch 'fixes' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai

David Brownell (1):
      ASoC: Clocking fixes for davinci-evm.c

Dmitry Baryshkov (1):
      pxa2xx-ac97: switch AC unit to correct state before probing

Grazvydas Ignotas (1):
      ASoC: Mark non-connected TWL4030 pins for pandora

Jarkko Nikula (1):
      ASoC: OMAP: Select OMAP pin multiplexing when using Nokia N810 ASoC drivers

Peter Ujfalusi (2):
      ASoC: New enum type: value_enum
      ASoC: TWL4030: Convert the bitfield enums to VALUE_ENUM type

 include/sound/soc-dapm.h        |   15 +++
 include/sound/soc.h             |   30 ++++++
 sound/arm/pxa2xx-ac97-lib.c     |   25 +++--
 sound/soc/codecs/twl4030.c      |  118 +++++++++++-------------
 sound/soc/davinci/davinci-evm.c |   19 +++-
 sound/soc/omap/Kconfig          |    1 +
 sound/soc/omap/omap3pandora.c   |   13 +++
 sound/soc/soc-core.c            |  107 +++++++++++++++++++++
 sound/soc/soc-dapm.c            |  197 ++++++++++++++++++++++++++++++++++++++-
 9 files changed, 447 insertions(+), 78 deletions(-)

Re: ASoC updates

[Takashi Iwai]

At Mon, 5 Jan 2009 17:56:34 +0000,
Mark Brown wrote:
> 
> The following changes since commit cad9d90ec3f6c497dce2b163046436b6ab5cf6b6:
>   Takashi Iwai (1):
>         Merge branch 'fixes' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai

Are all these 2.6.30 material or for 2.6.29?
For 2.6.30, I'll create another branch.  It won't be included in
linux-next, too, until rc1 is reached (Stephen requested so), but we
can keep it in our main branch, of course.


thanks,

Takashi


> David Brownell (1):
>       ASoC: Clocking fixes for davinci-evm.c
> 
> Dmitry Baryshkov (1):
>       pxa2xx-ac97: switch AC unit to correct state before probing
> 
> Grazvydas Ignotas (1):
>       ASoC: Mark non-connected TWL4030 pins for pandora
> 
> Jarkko Nikula (1):
>       ASoC: OMAP: Select OMAP pin multiplexing when using Nokia N810 ASoC drivers
> 
> Peter Ujfalusi (2):
>       ASoC: New enum type: value_enum
>       ASoC: TWL4030: Convert the bitfield enums to VALUE_ENUM type
> 
>  include/sound/soc-dapm.h        |   15 +++
>  include/sound/soc.h             |   30 ++++++
>  sound/arm/pxa2xx-ac97-lib.c     |   25 +++--
>  sound/soc/codecs/twl4030.c      |  118 +++++++++++-------------
>  sound/soc/davinci/davinci-evm.c |   19 +++-
>  sound/soc/omap/Kconfig          |    1 +
>  sound/soc/omap/omap3pandora.c   |   13 +++
>  sound/soc/soc-core.c            |  107 +++++++++++++++++++++
>  sound/soc/soc-dapm.c            |  197 ++++++++++++++++++++++++++++++++++++++-
>  9 files changed, 447 insertions(+), 78 deletions(-)
> 

Re: ASoC updates

[Mark Brown]

On Mon, Jan 05, 2009 at 09:53:21PM +0100, Takashi Iwai wrote:

> Are all these 2.6.30 material or for 2.6.29?

They're a bit on the edge, I'd expect them to be safe for 2.6.29 but the
conservative thing would be to punt to 2.6.30.  None of them are
critical.

> For 2.6.30, I'll create another branch.  It won't be included in
> linux-next, too, until rc1 is reached (Stephen requested so), but we
> can keep it in our main branch, of course.

How about branching off a fixes branch for 2.6.29 rather than a -next
branch?  That way we can keep using the same branch for further fixes
through the -rc series.  Doesn't make much difference either way,
though.

Re: ASoC updates

[Takashi Iwai]

At Mon, 5 Jan 2009 21:09:18 +0000,
Mark Brown wrote:
> 
> On Mon, Jan 05, 2009 at 09:53:21PM +0100, Takashi Iwai wrote:
> 
> > Are all these 2.6.30 material or for 2.6.29?
> 
> They're a bit on the edge, I'd expect them to be safe for 2.6.29 but the
> conservative thing would be to punt to 2.6.30.  None of them are
> critical.

OK, I'll include them in the next 2.6.29 pull request, which will
be sent today.

> > For 2.6.30, I'll create another branch.  It won't be included in
> > linux-next, too, until rc1 is reached (Stephen requested so), but we
> > can keep it in our main branch, of course.
> 
> How about branching off a fixes branch for 2.6.29 rather than a -next
> branch?  That way we can keep using the same branch for further fixes
> through the -rc series.  Doesn't make much difference either way,
> though.

Usually I make a fix-branch off like "fix/asoc" after rc1 or rc2,
i.e. the linux-next tree starts accepting the next-kernel stuff.
Until then, topic/* will keep the 2.6.29 stuff.

So my plan is to create next/asoc branch, switch to topic/asoc after
the merge window, and use fix/asoc branch for further fixing.


thanks,

Takashi

ASoC updates

[Mark Brown]

The following changes since commit 070504ade7a95a0f4395673717f3bb7d41793ca8:
  Mark Brown (1):
        ASoC: Fix L3 bus handling in Kconfig

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30

Mark Brown (1):
      ASoC: Push the codec runtime storage into the card structure

Peter Ujfalusi (5):
      ASoC: TWL4030: Syncronize the reg_cache for ANAMICL after the offset cancelation
      ASoC: TWL4030: Code clean up for codec power up and down
      ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use
      ASoC: TWL4030: Physical ADC and amplifier power switch change
      ASoC: TWL4030: Move the twl4030_power_up and _power_down function

Timur Tabi (1):
      ASoC: improve I2C initialization code in CS4270 driver

 include/sound/soc.h            |    3 +-
 sound/soc/codecs/ac97.c        |   20 ++--
 sound/soc/codecs/ad1980.c      |   12 +-
 sound/soc/codecs/ad73311.c     |    8 +-
 sound/soc/codecs/ak4535.c      |   14 +-
 sound/soc/codecs/cs4270.c      |  254 ++++++++++++++++++----------------------
 sound/soc/codecs/pcm3008.c     |   12 +-
 sound/soc/codecs/ssm2602.c     |   20 ++--
 sound/soc/codecs/tlv320aic23.c |   18 ++--
 sound/soc/codecs/tlv320aic26.c |    4 +-
 sound/soc/codecs/tlv320aic3x.c |   14 +-
 sound/soc/codecs/twl4030.c     |  224 +++++++++++++++++++-----------------
 sound/soc/codecs/uda134x.c     |   18 ++--
 sound/soc/codecs/uda1380.c     |   18 ++--
 sound/soc/codecs/wm8350.c      |   10 +-
 sound/soc/codecs/wm8510.c      |   16 ++--
 sound/soc/codecs/wm8580.c      |   10 +-
 sound/soc/codecs/wm8728.c      |   16 ++--
 sound/soc/codecs/wm8731.c      |   20 ++--
 sound/soc/codecs/wm8750.c      |   16 ++--
 sound/soc/codecs/wm8753.c      |   18 ++--
 sound/soc/codecs/wm8900.c      |    8 +-
 sound/soc/codecs/wm8903.c      |   18 ++--
 sound/soc/codecs/wm8971.c      |   14 +-
 sound/soc/codecs/wm8990.c      |   14 +-
 sound/soc/codecs/wm9705.c      |   15 ++-
 sound/soc/codecs/wm9712.c      |   21 ++--
 sound/soc/codecs/wm9713.c      |   17 ++--
 sound/soc/soc-core.c           |   37 +++---
 sound/soc/soc-dapm.c           |    6 +-
 sound/soc/soc-jack.c           |    4 +-
 31 files changed, 443 insertions(+), 456 deletions(-)

Re: ASoC updates

[Takashi Iwai]

At Tue, 27 Jan 2009 20:03:11 +0000,
Mark Brown wrote:
> 
> The following changes since commit 070504ade7a95a0f4395673717f3bb7d41793ca8:
>   Mark Brown (1):
>         ASoC: Fix L3 bus handling in Kconfig
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30

Pulled now.  Thanks!


Takashi