[alsa-devel] [PATCH] ASoC: dsd1791: Introduce driver for TI DSD1791 stereo codec
Liam Girdwood
lrg at ti.com
Mon Dec 19 22:44:30 CET 2011
On Mon, 2011-12-19 at 13:53 -0500, Michael Williamson wrote:
> 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>
Looks mostly fine, just a few comments :-
> ---
> 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);
You could make the code flow easier by using snd_soc_update_bits() here
and in other places.
> +}
> +
> +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),
Best to use SOC_DOUBLE_R here and rename to "Master Playback Volume"
> +};
> +
> +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;
I dont think goto is required in either case here so easier just to
return.
> + }
> +
> + 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);
What about your private data ?
> + 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");
GPL v2 according to the commnts at the top.
Thanks
Liam
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