[alsa-devel] [PATCH] ASoC: dsd1791: Introduce driver for TI DSD1791 stereo codec

Michael Williamson michael.williamson at criticallink.com
Mon Dec 19 19:53:30 CET 2011


This patch introduces a (spi) codec driver for the Texas Instruments
DSD1791 24 bit audio stereo DAC.

http://www.ti.com/product/dsd1791

Testing for basic operation using 16 and 24 bit I2S mode has been
performed using a MityDSP-L138 SOM and an Industrial I/O board
from Critical Link.

Signed-off-by: Michael Williamson <michael.williamson at criticallink.com>
---
This patch incorporates changes from the original RFC as a result of comments
received from Mark Brown, Leon Romanovsky, and Lars-Peter Clausen.  Thanks.

Summary of Changes:
	- Use devm_kzalloc()
	- use regmap cached I/O feature of framework
	- simplify code applying codec fmt configuration
	- clean up section attributes on local functions
	- make local functions/variables static where appropriate
	- remove set_sysclk method
	- remove unused tlv header file
	- remove chatter
	- strip "-codec" from driver name
	- fixup driver variable name for consistency
	- Add Left and Right volume control
	- Add digital mute

 sound/soc/codecs/Kconfig   |    4 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/dsd1791.c |  256 ++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 262 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/dsd1791.c

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4584514..95b7969 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
 	select SND_SOC_DFBMCS320
+	select SND_SOC_DSD1791 if SPI_MASTER
 	select SND_SOC_JZ4740_CODEC if SOC_JZ4740
 	select SND_SOC_LM4857 if I2C
 	select SND_SOC_MAX98088 if I2C
@@ -205,6 +206,9 @@ config SND_SOC_DA7210
 config SND_SOC_DFBMCS320
 	tristate
 
+config SND_SOC_DSD1791
+	tristate
+
 config SND_SOC_DMIC
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index a2c7842..d6b5f6a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -21,6 +21,7 @@ snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
 snd-soc-dfbmcs320-objs := dfbmcs320.o
 snd-soc-dmic-objs := dmic.o
+snd-soc-dsd1791-objs := dsd1791.o
 snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-max98095-objs := max98095.o
@@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_CS4271)	+= snd-soc-cs4271.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_DFBMCS320)	+= snd-soc-dfbmcs320.o
+obj-$(CONFIG_SND_SOC_DSD1791)	+= snd-soc-dsd1791.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
diff --git a/sound/soc/codecs/dsd1791.c b/sound/soc/codecs/dsd1791.c
new file mode 100644
index 0000000..41bbc61
--- /dev/null
+++ b/sound/soc/codecs/dsd1791.c
@@ -0,0 +1,256 @@
+/*
+ * ALSA SoC codec driver for Texas Instruments DSD1791.
+ *
+ * Author:      (C) Michael Williamson <michael.williamson at criticallink.com>
+ * Copyright:   (C) 2011 Critical Link, LLC
+ *
+ * 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/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/spi/spi.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#define DSD1791_REG_DIGATT_L	16
+#define DSD1791_REG_DIGATT_R	17
+#define DSD1791_REG_AUDFMT	18
+#define DSD1791_REG_SRST	20
+
+#define DSD1791_FMT_16RJ	(0<<4)
+#define DSD1791_FMT_20RJ	(1<<4)
+#define DSD1791_FMT_24RJ	(2<<4)
+#define DSD1791_FMT_24LJ	(3<<4)
+#define DSD1791_FMT_16I2S	(4<<4)
+#define DSD1791_FMT_24I2S	(5<<4)
+#define DSD1791_FMT_MASK	0x70
+
+/* DSD1791 register cache (16 through 23 are used) */
+static const u8 dsd1791_reg[] = {
+	[16] = 0xFF,
+	[17] = 0xFF,
+	[18] = 0x50,
+	[19] = 0x00,
+	[20] = 0x00,
+	[21] = 0x01,
+	[22] = 0x00,
+	[23] = 0x00,
+};
+
+struct dsd1791 {
+	struct spi_device *spi;
+	struct snd_soc_codec codec;
+	int dai_fmt;
+	int pcm_fmt;
+};
+
+static int dsd1791_set_format_word(struct dsd1791 *dsd1791,
+				   struct snd_soc_codec *codec)
+{
+	u8 fmt = 0;
+	u8 reg;
+
+	switch (dsd1791->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+
+	case SND_SOC_DAIFMT_I2S:
+		switch (dsd1791->pcm_fmt) {
+		case SNDRV_PCM_FORMAT_S16_LE:
+			fmt = DSD1791_FMT_16I2S;
+			break;
+		case SNDRV_PCM_FORMAT_S24_LE:
+			fmt = DSD1791_FMT_24I2S;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_RIGHT_J:
+		switch (dsd1791->pcm_fmt) {
+		case SNDRV_PCM_FORMAT_S16_LE:
+			fmt = DSD1791_FMT_16RJ;
+			break;
+		case SNDRV_PCM_FORMAT_S24_LE:
+			fmt = DSD1791_FMT_24RJ;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_LEFT_J:
+		switch (dsd1791->pcm_fmt) {
+		case SNDRV_PCM_FORMAT_S24_LE:
+			fmt = DSD1791_FMT_24LJ;
+		default:
+			return -EINVAL;
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+	reg = snd_soc_read(codec, DSD1791_REG_AUDFMT);
+	reg &= ~(DSD1791_FMT_MASK);
+	reg |= fmt;
+	return snd_soc_write(codec, DSD1791_REG_AUDFMT, reg);
+}
+
+static int dsd1791_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 reg;
+
+	reg = snd_soc_read(codec, DSD1791_REG_AUDFMT);
+	if (mute)
+		reg |= 1;
+	else
+		reg &= ~1;
+	return snd_soc_write(codec, DSD1791_REG_AUDFMT, reg);
+}
+
+static int dsd1791_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec);
+
+	dsd1791->pcm_fmt = params_format(params);
+
+	return dsd1791_set_format_word(dsd1791, codec);
+}
+
+static int dsd1791_set_fmt(struct snd_soc_dai *codec_dai,
+			     unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec);
+
+	dsd1791->dai_fmt = fmt;
+
+	return dsd1791_set_format_word(dsd1791, codec);
+}
+
+#define DSD1791_RATES	SNDRV_PCM_RATE_8000_192000
+#define DSD1791_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE |\
+			 SNDRV_PCM_FMTBIT_S24_LE)
+
+static const struct snd_soc_dai_ops dsd1791_dai_ops = {
+	.hw_params	= dsd1791_hw_params,
+	.set_fmt	= dsd1791_set_fmt,
+	.digital_mute	= dsd1791_mute,
+};
+
+static struct snd_soc_dai_driver dsd1791_dai = {
+	.name = "dsd1791",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = DSD1791_RATES,
+		.formats = DSD1791_FORMATS,
+	},
+	.ops = &dsd1791_dai_ops,
+};
+
+static const struct snd_kcontrol_new dsd1791_snd_controls[] = {
+	SOC_SINGLE("Left Playback Volume", DSD1791_REG_DIGATT_L, 0, 255, 0),
+	SOC_SINGLE("Right Playback Volume", DSD1791_REG_DIGATT_R, 0, 255, 0),
+};
+
+static int dsd1791_probe(struct snd_soc_codec *codec)
+{
+	u8 reg;
+	int ret;
+	struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec);
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_SPI);
+	if (ret) {
+		dev_err(codec->dev, "Failed to set Cache I/O: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_write(codec, DSD1791_REG_SRST, 0x40);
+	if (ret) {
+		dev_err(codec->dev, "Unable to reset device: %d\n", ret);
+		goto err;
+	}
+
+	/* default format after reset */
+	dsd1791->dai_fmt = SND_SOC_DAIFMT_I2S;
+	dsd1791->pcm_fmt = SNDRV_PCM_FORMAT_S24_LE;
+
+	/* enable attenuation control */
+	reg = snd_soc_read(codec, DSD1791_REG_AUDFMT);
+	reg |= 0x80;
+	snd_soc_write(codec, DSD1791_REG_AUDFMT, reg);
+
+	snd_soc_add_controls(codec, dsd1791_snd_controls,
+				ARRAY_SIZE(dsd1791_snd_controls));
+	return 0;
+err:
+	return ret;
+}
+
+struct snd_soc_codec_driver dsd1791_codec_driver = {
+	.probe			= dsd1791_probe,
+	.reg_cache_size		= ARRAY_SIZE(dsd1791_reg),
+	.reg_word_size		= sizeof(u8),
+	.reg_cache_default	= dsd1791_reg,
+};
+
+static int __devinit dsd1791_spi_probe(struct spi_device *spi)
+{
+	struct dsd1791 *dsd1791;
+
+	dsd1791 = devm_kzalloc(&spi->dev, sizeof *dsd1791, GFP_KERNEL);
+	if (!dsd1791)
+		return -ENOMEM;
+
+	spi_set_drvdata(spi, dsd1791);
+
+	return snd_soc_register_codec(&spi->dev,
+		&dsd1791_codec_driver, &dsd1791_dai, 1);
+};
+
+static int __devexit dsd1791_spi_remove(struct spi_device *spi)
+{
+	snd_soc_unregister_codec(&spi->dev);
+	return 0;
+}
+
+static struct spi_driver dsd1791_spi_driver = {
+	.driver = {
+		.name = "dsd1791",
+		.owner = THIS_MODULE,
+	},
+	.probe = dsd1791_spi_probe,
+	.remove = __devexit_p(dsd1791_spi_remove),
+};
+
+static int __init dsd1791_init(void)
+{
+	return spi_register_driver(&dsd1791_spi_driver);
+}
+module_init(dsd1791_init);
+
+static void __exit dsd1791_exit(void)
+{
+	spi_unregister_driver(&dsd1791_spi_driver);
+}
+module_exit(dsd1791_exit);
+
+MODULE_DESCRIPTION("ASoC DSD1791 codec driver");
+MODULE_AUTHOR("Michael Williamson");
+MODULE_LICENSE("GPL");
-- 
1.7.0.4



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