[PATCH 0/6] ASoC updates

[PATCH 0/6] ASoC updates

[Mark Brown]

The following changes since commit 72f2b894455775b980a5ac7ae70ab560b3d3d247:
  Mark Brown (1):
        ASoC: Move uda134x_codec.h to uda134x.h

are available in the git repository at:

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

Arun KS (1):
      ASoC: Add support for omap2evm board

Hugo Villeneuve (2):
      ASoC: Add PCM3008 ALSA SoC driver
      ASoC: Add driver for the Lyrtech SFFSDR board

Mark Brown (3):
      ASoC: Remove unused snd_soc_machine_config declaration
      ASoC: s3c24xx_uda134x DAI accessor functions and static cleanup
      ASoC: Convert blackfin machines to use DAI accessor functions

 include/sound/soc.h                 |    1 -
 sound/soc/blackfin/bf5xx-ad73311.c  |    2 +-
 sound/soc/blackfin/bf5xx-ssm2602.c  |    6 +-
 sound/soc/codecs/Kconfig            |    4 +
 sound/soc/codecs/Makefile           |    2 +
 sound/soc/codecs/pcm3008.c          |  201 +++++++++++++++++++++++++++++++++++
 sound/soc/codecs/pcm3008.h          |   25 +++++
 sound/soc/davinci/Kconfig           |   10 ++
 sound/soc/davinci/Makefile          |    2 +
 sound/soc/davinci/davinci-sffsdr.c  |  156 +++++++++++++++++++++++++++
 sound/soc/omap/Kconfig              |    9 ++
 sound/soc/omap/Makefile             |    2 +
 sound/soc/omap/omap2evm.c           |  150 ++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c24xx_uda134x.c |   27 +++---
 14 files changed, 578 insertions(+), 19 deletions(-)
 create mode 100644 sound/soc/codecs/pcm3008.c
 create mode 100644 sound/soc/codecs/pcm3008.h
 create mode 100644 sound/soc/davinci/davinci-sffsdr.c
 create mode 100644 sound/soc/omap/omap2evm.c

[PATCH 1/6] ASoC: Remove unused snd_soc_machine_config declaration

[Mark Brown]

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h |    1 -
 1 files changed, 0 insertions(+), 1 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index fa1b99b..077dfe4 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -228,7 +228,6 @@ struct snd_soc_dai_mode;
 struct snd_soc_pcm_runtime;
 struct snd_soc_dai;
 struct snd_soc_codec;
-struct snd_soc_machine_config;
 struct soc_enum;
 struct snd_soc_ac97_ops;
 struct snd_soc_clock_info;
-- 
1.5.6.5

[PATCH 2/6] ASoC: Add PCM3008 ALSA SoC driver

[Mark Brown]

From: Hugo Villeneuve <hugo@hugovil.com>

The PCM3008 is a 16-bit stereo audio codec. It accepts
left-justified format for ADC, and right-justified format
for DAC. Independent power-down modes for ADC and DAC are
provided, as well as a digital de-emphasis filter (4 modes).

[Merged Makefile & Kconfig, changed asm/gpio.h to linux/gpio.h -- broonie]

Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig   |    4 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/pcm3008.c |  201 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/pcm3008.h |   25 ++++++
 4 files changed, 232 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/pcm3008.c
 create mode 100644 sound/soc/codecs/pcm3008.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 04f49f5..bf68052 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -5,6 +5,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AD73311 if I2C
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_CS4270 if I2C
+	select SND_SOC_PCM3008
 	select SND_SOC_SSM2602 if I2C
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -70,6 +71,9 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_L3
        tristate
 
+config SND_SOC_PCM3008
+       tristate
+
 config SND_SOC_SSM2602
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index de65723..9a20fdd 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-l3-objs := l3.o
+snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
@@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
+obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
new file mode 100644
index 0000000..2b26e1d
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.c
@@ -0,0 +1,201 @@
+/*
+ * ALSA Soc PCM3008 codec support
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on AC97 Soc codec, original copyright follow:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ *
+ *  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.
+ *
+ * Generic PCM3008 support.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "pcm3008.h"
+
+#define PCM3008_VERSION "0.2"
+
+#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |	\
+		       SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai pcm3008_dai = {
+	.name = "PCM3008 HiFi",
+	.type = SND_SOC_DAI_I2S,
+	.playback = {
+		.stream_name = "PCM3008 Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = PCM3008_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "PCM3008 Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = PCM3008_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+};
+EXPORT_SYMBOL_GPL(pcm3008_dai);
+
+static void pcm3008_gpio_free(struct pcm3008_setup_data *setup)
+{
+	gpio_free(setup->dem0_pin);
+	gpio_free(setup->dem1_pin);
+	gpio_free(setup->pdad_pin);
+	gpio_free(setup->pdda_pin);
+}
+
+static int pcm3008_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+	int ret = 0;
+
+	printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->codec)
+		return -ENOMEM;
+
+	codec = socdev->codec;
+	mutex_init(&codec->mutex);
+
+	codec->name = "PCM3008";
+	codec->owner = THIS_MODULE;
+	codec->dai = &pcm3008_dai;
+	codec->num_dai = 1;
+	codec->write = NULL;
+	codec->read = NULL;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	/* Register PCMs. */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "pcm3008: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* Register Card. */
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "pcm3008: failed to register card\n");
+		goto card_err;
+	}
+
+	/* DEM1  DEM0  DE-EMPHASIS_MODE
+	 * Low   Low   De-emphasis 44.1 kHz ON
+	 * Low   High  De-emphasis OFF
+	 * High  Low   De-emphasis 48 kHz ON
+	 * High  High  De-emphasis 32 kHz ON
+	 */
+
+	/* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */
+	ret = gpio_request(setup->dem0_pin, "codec_dem0");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->dem0_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */
+	ret = gpio_request(setup->dem1_pin, "codec_dem1");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->dem1_pin, 0);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure PDAD GPIO. */
+	ret = gpio_request(setup->pdad_pin, "codec_pdad");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->pdad_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure PDDA GPIO. */
+	ret = gpio_request(setup->pdda_pin, "codec_pdda");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->pdda_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	return ret;
+
+gpio_err:
+	pcm3008_gpio_free(setup);
+card_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	kfree(socdev->codec);
+
+	return ret;
+}
+
+static int pcm3008_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	if (!codec)
+		return 0;
+
+	pcm3008_gpio_free(setup);
+	snd_soc_free_pcms(socdev);
+	kfree(socdev->codec);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	gpio_set_value(setup->pdad_pin, 0);
+	gpio_set_value(setup->pdda_pin, 0);
+
+	return 0;
+}
+
+static int pcm3008_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	gpio_set_value(setup->pdad_pin, 1);
+	gpio_set_value(setup->pdda_pin, 1);
+
+	return 0;
+}
+#else
+#define pcm3008_soc_suspend NULL
+#define pcm3008_soc_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_pcm3008 = {
+	.probe = 	pcm3008_soc_probe,
+	.remove = 	pcm3008_soc_remove,
+	.suspend =	pcm3008_soc_suspend,
+	.resume =	pcm3008_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008);
+
+MODULE_DESCRIPTION("Soc PCM3008 driver");
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h
new file mode 100644
index 0000000..d04e87d
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.h
@@ -0,0 +1,25 @@
+/*
+ * PCM3008 ALSA SoC Layer
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * 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.
+ */
+
+#ifndef __LINUX_SND_SOC_PCM3008_H
+#define __LINUX_SND_SOC_PCM3008_H
+
+struct pcm3008_setup_data {
+	unsigned dem0_pin;
+	unsigned dem1_pin;
+	unsigned pdad_pin;
+	unsigned pdda_pin;
+};
+
+extern struct snd_soc_codec_device soc_codec_dev_pcm3008;
+extern struct snd_soc_dai pcm3008_dai;
+
+#endif
-- 
1.5.6.5

[PATCH 3/6] ASoC: Add driver for the Lyrtech SFFSDR board

[Mark Brown]

From: Hugo Villeneuve <hugo@hugovil.com>

The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an
FPGA that generates the bit clock and the master clock

[Downgraded the rate debug print to pr_debug() in hw_params, converted
asm/gpio.h to linux/gpio.h -- broonie]

Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/davinci/Kconfig          |   10 +++
 sound/soc/davinci/Makefile         |    2 +
 sound/soc/davinci/davinci-sffsdr.c |  156 ++++++++++++++++++++++++++++++++++++
 3 files changed, 168 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/davinci/davinci-sffsdr.c

diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 8f7e338..b502741 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -17,3 +17,13 @@ config SND_DAVINCI_SOC_EVM
 	help
 	  Say Y if you want to add support for SoC audio on TI
 	  DaVinci EVM platform.
+
+config SND_DAVINCI_SOC_SFFSDR
+	tristate "SoC Audio support for SFFSDR"
+	depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR
+	select SND_DAVINCI_SOC_I2S
+	select SND_SOC_PCM3008
+	select SFFSDR_FPGA
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Lyrtech SFFSDR board.
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index ca772e5..ca8bae1 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -7,5 +7,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
 
 # DAVINCI Machine Support
 snd-soc-evm-objs := davinci-evm.o
+snd-soc-sffsdr-objs := davinci-sffsdr.o
 
 obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
new file mode 100644
index 0000000..69a8a76
--- /dev/null
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -0,0 +1,156 @@
+/*
+ * ASoC driver for Lyrtech SFFSDR board.
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow:
+ * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.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/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/dma.h>
+#include <asm/plat-sffsdr/sffsdr-fpga.h>
+
+#include <mach/mcbsp.h>
+#include <mach/edma.h>
+
+#include "../codecs/pcm3008.h"
+#include "davinci-pcm.h"
+#include "davinci-i2s.h"
+
+static int sffsdr_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 *cpu_dai = rtd->dai->cpu_dai;
+	int fs;
+	int ret = 0;
+
+	/* Set cpu DAI configuration:
+	 * CLKX and CLKR are the inputs for the Sample Rate Generator.
+	 * FSX and FSR are outputs, driven by the sample Rate Generator. */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_RIGHT_J |
+				  SND_SOC_DAIFMT_CBM_CFS |
+				  SND_SOC_DAIFMT_IB_NF);
+	if (ret < 0)
+		return ret;
+
+	/* Fsref can be 32000, 44100 or 48000. */
+	fs = params_rate(params);
+
+	pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs);
+
+	return sffsdr_fpga_set_codec_fs(fs);
+}
+
+static struct snd_soc_ops sffsdr_ops = {
+	.hw_params = sffsdr_hw_params,
+};
+
+/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sffsdr_dai = {
+	.name = "PCM3008", /* Codec name */
+	.stream_name = "PCM3008 HiFi",
+	.cpu_dai = &davinci_i2s_dai,
+	.codec_dai = &pcm3008_dai,
+	.ops = &sffsdr_ops,
+};
+
+/* davinci-sffsdr audio machine driver */
+static struct snd_soc_machine snd_soc_machine_sffsdr = {
+	.name = "DaVinci SFFSDR",
+	.dai_link = &sffsdr_dai,
+	.num_links = 1,
+};
+
+/* sffsdr audio private data */
+static struct pcm3008_setup_data sffsdr_pcm3008_setup = {
+	.dem0_pin = GPIO(45),
+	.dem1_pin = GPIO(46),
+	.pdad_pin = GPIO(47),
+	.pdda_pin = GPIO(38),
+};
+
+/* sffsdr audio subsystem */
+static struct snd_soc_device sffsdr_snd_devdata = {
+	.machine = &snd_soc_machine_sffsdr,
+	.platform = &davinci_soc_platform,
+	.codec_dev = &soc_codec_dev_pcm3008,
+	.codec_data = &sffsdr_pcm3008_setup,
+};
+
+static struct resource sffsdr_snd_resources[] = {
+	{
+		.start = DAVINCI_MCBSP_BASE,
+		.end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+		.flags = IORESOURCE_MEM,
+	},
+};
+
+static struct evm_snd_platform_data sffsdr_snd_data = {
+	.tx_dma_ch	= DAVINCI_DMA_MCBSP_TX,
+	.rx_dma_ch	= DAVINCI_DMA_MCBSP_RX,
+};
+
+static struct platform_device *sffsdr_snd_device;
+
+static int __init sffsdr_init(void)
+{
+	int ret;
+
+	sffsdr_snd_device = platform_device_alloc("soc-audio", 0);
+	if (!sffsdr_snd_device) {
+		printk(KERN_ERR "platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata);
+	sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev;
+	sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data;
+
+	ret = platform_device_add_resources(sffsdr_snd_device,
+					    sffsdr_snd_resources,
+					    ARRAY_SIZE(sffsdr_snd_resources));
+	if (ret) {
+		printk(KERN_ERR "platform device add ressources failed\n");
+		goto error;
+	}
+
+	ret = platform_device_add(sffsdr_snd_device);
+	if (ret)
+		goto error;
+
+	return ret;
+
+error:
+	platform_device_put(sffsdr_snd_device);
+	return ret;
+}
+
+static void __exit sffsdr_exit(void)
+{
+	platform_device_unregister(sffsdr_snd_device);
+}
+
+module_init(sffsdr_init);
+module_exit(sffsdr_exit);
+
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver");
+MODULE_LICENSE("GPL");
-- 
1.5.6.5

[PATCH 4/6] ASoC: Add support for omap2evm board

[Mark Brown]

From: Arun KS <arunks@mistralsolutions.com>

This patch adds twl4030 audio support on omap2evm

Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/Kconfig    |    9 +++
 sound/soc/omap/Makefile   |    2 +
 sound/soc/omap/omap2evm.c |  150 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 161 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/omap/omap2evm.c

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index cf40e42..6c56277 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -30,4 +30,13 @@ config SND_OMAP_SOC_OVERO
 	help
 	  Say Y if you want to add support for SoC audio on the Gumstix Overo.
 
+config SND_OMAP_SOC_OMAP2EVM
+	tristate "SoC Audio support for OMAP2EVM board"
+	depends on SND_OMAP_SOC && MACH_OMAP2EVM
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the omap2evm board.
+
+
 
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index fefc9be..f5da3cc 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -9,8 +9,10 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 snd-soc-n810-objs := n810.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
+snd-soc-omap2evm-objs := omap2evm.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
+obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
 
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
new file mode 100644
index 0000000..c376213
--- /dev/null
+++ b/sound/soc/omap/omap2evm.c
@@ -0,0 +1,150 @@
+/*
+ * omap2evm.c  --  SoC audio machine driver for omap2evm board
+ *
+ * Author: Arun KS <arunks@mistralsolutions.com>
+ *
+ * Based on sound/soc/omap/overo.c by Steve Sakoman
+ *
+ * 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 program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap2evm_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;
+
+	/* 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) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		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) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap2evm_ops = {
+	.hw_params = omap2evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap2evm_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &omap2evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_omap2evm = {
+	.name = "omap2evm",
+	.dai_link = &omap2evm_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap2evm_snd_devdata = {
+	.machine = &snd_soc_machine_omap2evm,
+	.platform = &omap_soc_platform,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap2evm_snd_device;
+
+static int __init omap2evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap2evm()) {
+		pr_debug("Not omap2evm!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "omap2evm SoC init\n");
+
+	omap2evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap2evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata);
+	omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev;
+	*(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(omap2evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap2evm_snd_device);
+
+	return ret;
+}
+module_init(omap2evm_soc_init);
+
+static void __exit omap2evm_soc_exit(void)
+{
+	platform_device_unregister(omap2evm_snd_device);
+}
+module_exit(omap2evm_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC omap2evm");
+MODULE_LICENSE("GPL");
-- 
1.5.6.5

[PATCH 5/6] ASoC: s3c24xx_uda134x DAI accessor functions and static cleanup

[Mark Brown]

Missed these during review.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c24xx_uda134x.c |   27 +++++++++++++--------------
 1 files changed, 13 insertions(+), 14 deletions(-)

diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index 487b010..92d90f2 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -63,7 +63,7 @@ static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
 
 static struct platform_device *s3c24xx_uda134x_snd_device;
 
-int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
+static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
 {
 	int ret = 0;
 #ifdef ENFORCE_RATES
@@ -115,7 +115,7 @@ int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
 	return ret;
 }
 
-void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream)
+static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream)
 {
 	mutex_lock(&clk_lock);
 	pr_debug("%s %d\n", __func__, clk_users);
@@ -180,39 +180,38 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+	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;
 
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+	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;
 
-	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk,
-						SND_SOC_CLOCK_IN);
+	ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk,
+			SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
 
-	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
-						fs_mode);
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode);
 	if (ret < 0)
 		return ret;
 
-	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
-						S3C2410_IISMOD_32FS);
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+			S3C2410_IISMOD_32FS);
 	if (ret < 0)
 		return ret;
 
-	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
-					  S3C24XX_PRESCALE(div, div));
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+			S3C24XX_PRESCALE(div, div));
 	if (ret < 0)
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk,
-						SND_SOC_CLOCK_OUT);
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+			SND_SOC_CLOCK_OUT);
 	if (ret < 0)
 		return ret;
 
-- 
1.5.6.5

[PATCH 6/6] ASoC: Convert blackfin machines to use DAI accessor functions

[Mark Brown]

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/blackfin/bf5xx-ad73311.c |    2 +-
 sound/soc/blackfin/bf5xx-ssm2602.c |    6 +++---
 2 files changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 622c9b9..47da49b 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -168,7 +168,7 @@ static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream,
 		params_format(params));
 
 	/* set cpu DAI configuration */
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e15f67f..744a90e 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -92,17 +92,17 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
 	 */
 
 	/* set codec DAI configuration */
-	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+	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 = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+	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;
 
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
-- 
1.5.6.5

Re: [PATCH 0/6] ASoC updates

[Takashi Iwai]

At Wed, 19 Nov 2008 13:34:27 +0000,
Mark Brown wrote:
> 
> The following changes since commit 72f2b894455775b980a5ac7ae70ab560b3d3d247:
>   Mark Brown (1):
>         ASoC: Move uda134x_codec.h to uda134x.h
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Pulled now.  Thanks.


Takashi

[PATCH 0/6] ASoC updates

[Mark Brown]

The following changes since commit 3e3b5651631629482e0e1541f8c4298a435bb100:
  Mark Brown (1):
        ALSA: ASoC: Use codec digital mute when stopping playback

are available in the git repository at:

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

Alan Horstmann (1):
      ASoC: Fix WM8731 SPI write

Dmitry Baryshkov (4):
      ASoC: Support runtime selection of CPU in pxa2xx-i2s
      ALSA: Separate common pxa2xx-ac97 code
      ALSA: pxa2xx-ac97-lib: support building for several CPUs
      ALSA: Separate common pxa2xx-pcm code

Kenneth Kiraly (1):
      ASoC: Add WM8971 CODEC driver

 include/sound/pxa2xx-lib.h  |   45 ++
 sound/arm/Kconfig           |    9 +
 sound/arm/Makefile          |    4 +
 sound/arm/pxa2xx-ac97-lib.c |  384 ++++++++++++++++++
 sound/arm/pxa2xx-ac97.c     |  247 +-----------
 sound/arm/pxa2xx-pcm-lib.c  |  278 +++++++++++++
 sound/arm/pxa2xx-pcm.c      |  252 +-----------
 sound/arm/pxa2xx-pcm.h      |   13 +-
 sound/soc/codecs/Kconfig    |    4 +
 sound/soc/codecs/Makefile   |    2 +
 sound/soc/codecs/wm8731.c   |    5 +-
 sound/soc/codecs/wm8971.c   |  942 +++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8971.h   |   64 +++
 sound/soc/pxa/Kconfig       |    3 +
 sound/soc/pxa/pxa2xx-ac97.c |  272 +------------
 sound/soc/pxa/pxa2xx-i2s.c  |   20 +-
 sound/soc/pxa/pxa2xx-pcm.c  |  261 +------------
 sound/soc/pxa/pxa2xx-pcm.h  |   15 -
 18 files changed, 1802 insertions(+), 1018 deletions(-)
 create mode 100644 include/sound/pxa2xx-lib.h
 create mode 100644 sound/arm/pxa2xx-ac97-lib.c
 create mode 100644 sound/arm/pxa2xx-pcm-lib.c
 create mode 100644 sound/soc/codecs/wm8971.c
 create mode 100644 sound/soc/codecs/wm8971.h

Re: [PATCH 0/6] ASoC updates

[Takashi Iwai]

At Mon, 15 Sep 2008 14:24:19 +0100,
Mark Brown wrote:
> 
> The following changes since commit 3e3b5651631629482e0e1541f8c4298a435bb100:
>   Mark Brown (1):
>         ALSA: ASoC: Use codec digital mute when stopping playback
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, pulled in, pushed out.


Takashi