[alsa-devel] [PATCH] ASoC: Add MAX9850 codec driver

Dimitris Papastamos dp at opensource.wolfsonmicro.com
Mon Mar 7 14:48:57 CET 2011


On Mon, Mar 07, 2011 at 01:45:13PM +0100, Christian Glindkamp wrote:
> This patch adds ASoC support for the MAX9850 codec with headphone
> amplifier.
> 
> Supported features:
> - Playback
> - 16, 20 and 24 bit audio
> - 8k - 48k sample rates
> - DAPM
> 
> Only 16 bit audio was tested while the codec was connected to an
> AT91SAM9G20 SSC in master mode.
> 
> Signed-off-by: Christian Glindkamp <christian.glindkamp at taskit.de>
> ---
> 
> I've all ready sent this patch some time ago, but it hung in the moderation
> queue. This is a slightly modified version. Unfortunately I do not have the
> hardware anymore to test suggested changes that alter function.
> 
>  sound/soc/codecs/Kconfig   |    4 +
>  sound/soc/codecs/Makefile  |    2 +
>  sound/soc/codecs/max9850.c |  358 ++++++++++++++++++++++++++++++++++++++++++++
>  sound/soc/codecs/max9850.h |   41 +++++
>  4 files changed, 405 insertions(+), 0 deletions(-)
>  create mode 100644 sound/soc/codecs/max9850.c
>  create mode 100644 sound/soc/codecs/max9850.h
> 
> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
> index e239345..51e9844 100644
> --- a/sound/soc/codecs/Kconfig
> +++ b/sound/soc/codecs/Kconfig
> @@ -31,6 +31,7 @@ config SND_SOC_ALL_CODECS
>  	select SND_SOC_DA7210 if I2C
>  	select SND_SOC_JZ4740_CODEC if SOC_JZ4740
>  	select SND_SOC_MAX98088 if I2C
> +	select SND_SOC_MAX9850 if I2C
>  	select SND_SOC_MAX9877 if I2C
>  	select SND_SOC_PCM3008
>  	select SND_SOC_SN95031 if INTEL_SCU_IPC
> @@ -179,6 +180,9 @@ config SND_SOC_DMIC
>  config SND_SOC_MAX98088
>         tristate
>  
> +config SND_SOC_MAX9850
> +	tristate
> +
>  config SND_SOC_PCM3008
>         tristate
>  
> diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
> index ae10507..f2efd1c 100644
> --- a/sound/soc/codecs/Makefile
> +++ b/sound/soc/codecs/Makefile
> @@ -18,6 +18,7 @@ snd-soc-da7210-objs := da7210.o
>  snd-soc-dmic-objs := dmic.o
>  snd-soc-l3-objs := l3.o
>  snd-soc-max98088-objs := max98088.o
> +snd-soc-max9850-objs := max9850.o
>  snd-soc-pcm3008-objs := pcm3008.o
>  snd-soc-alc5623-objs := alc5623.o
>  snd-soc-sn95031-objs := sn95031.o
> @@ -102,6 +103,7 @@ obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
>  obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
>  obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
>  obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
> +obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
>  obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
>  obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
>  obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
> diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
> new file mode 100644
> index 0000000..a8c1f95
> --- /dev/null
> +++ b/sound/soc/codecs/max9850.c
> @@ -0,0 +1,358 @@
> +/*
> + * max9850.c  --  codec driver for max9850
> + *
> + * Copyright (C) 2011 taskit GmbH
> + *
> + * Author: Christian Glindkamp <christian.glindkamp at taskit.de>
> + *
> + * Initial development of this code was funded by
> + * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/
> + *
> + * 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/init.h>
> +#include <linux/i2c.h>
> +#include <linux/slab.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/soc.h>
> +#include <sound/tlv.h>
> +
> +#include "max9850.h"
> +
> +struct max9850_priv {
> +	unsigned int sysclk;
> +};
> +
> +/* max9850 register cache */
> +static const u8 max9850_reg[MAX9850_CACHEREGNUM] = {
> +	0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
> +};
> +
> +/* these registers are not used at the moment but provided for the sake of
> + * completeness */
> +static int max9850_volatile_register(unsigned int reg)
> +{
> +	switch (reg) {
> +	case MAX9850_STATUSA:
> +	case MAX9850_STATUSB:
> +		return 1;
> +	default:
> +		return 0;
> +	}
> +}

This code doesn't seem to have been developed against for-2.6.39.  The
signature of the volatile_register callback has changed to include a
pointer to the snd_soc_codec structure.

> +static const unsigned int max9850_tlv[] = {
> +	TLV_DB_RANGE_HEAD(4),
> +	0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0),
> +	0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0),
> +	0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0),
> +	0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0),
> +};
> +
> +static const struct snd_kcontrol_new max9850_controls[] = {
> +SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv),
> +SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1),
> +SOC_SINGLE("Mono", MAX9850_GENERAL_PURPOSE, 2, 1, 0),
> +};

Mono Switch?

> +static const struct snd_kcontrol_new max9850_mixer_controls[] = {
> +	SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0),
> +};
> +
> +static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = {
> +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0),
> +SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0),
> +SND_SOC_DAPM_OUTPUT("OUTL"),
> +SND_SOC_DAPM_OUTPUT("OUTR"),
> +SND_SOC_DAPM_OUTPUT("HPL"),
> +SND_SOC_DAPM_OUTPUT("HPR"),
> +SND_SOC_DAPM_INPUT("INL"),
> +SND_SOC_DAPM_INPUT("INR"),
> +SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0),
> +SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
> +SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0,
> +		&max9850_mixer_controls[0],
> +		ARRAY_SIZE(max9850_mixer_controls)),
> +};

Consider grouping the input and output pins logically separately.

> +static const struct snd_soc_dapm_route intercon[] = {
> +	/* output mixer */
> +	{"Output Mixer", NULL, "DAC"},
> +	{"Output Mixer", "Line In Switch", "Line Input"},
> +
> +	/* outputs */
> +	{"Headphone Output", NULL, "Output Mixer"},
> +	{"HPL", NULL, "Headphone Output"},
> +	{"HPR", NULL, "Headphone Output"},
> +	{"OUTL", NULL, "Output Mixer"},
> +	{"OUTR", NULL, "Output Mixer"},
> +
> +	/* inputs */
> +	{"Line Input", NULL, "INL"},
> +	{"Line Input", NULL, "INR"},
> +
> +	/* supplies */
> +	{"DAC", NULL, "MCLK"},
> +};

Are all these really statically connected? 

> +static int max9850_hw_params(struct snd_pcm_substream *substream,
> +			     struct snd_pcm_hw_params *params,
> +			     struct snd_soc_dai *dai)
> +{
> +	struct snd_soc_codec *codec = dai->codec;
> +	struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
> +	u64 lrclk_div;
> +	u8 sf, da;
> +
> +	/* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */
> +	sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1;
> +	lrclk_div = (1 << 22);
> +	lrclk_div *= params_rate(params);
> +	lrclk_div *= sf;
> +	do_div(lrclk_div, max9850->sysclk);
> +
> +	snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f);
> +	snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff);
> +
> +	da = snd_soc_read(codec, MAX9850_DIGITAL_AUDIO);
> +	switch (params_format(params)) {
> +	case SNDRV_PCM_FORMAT_S16_LE:
> +		break;
> +	case SNDRV_PCM_FORMAT_S20_3LE:
> +		da |= 0x2;
> +		break;
> +	case SNDRV_PCM_FORMAT_S24_LE:
> +		da |= 0x3;
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +	snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da);
> +
> +	return 0;
> +}
> +
> +static int max9850_set_dai_sysclk(struct snd_soc_dai *codec_dai,
> +		int clk_id, unsigned int freq, int dir)
> +{
> +	struct snd_soc_codec *codec = codec_dai->codec;
> +	struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
> +
> +	/* calculate mclk -> iclk divider */
> +	if (freq <= 13000000)
> +		snd_soc_write(codec, MAX9850_CLOCK, 0x0);
> +	else if (freq <= 26000000)
> +		snd_soc_write(codec, MAX9850_CLOCK, 0x4);
> +	else if (freq <= 40000000)
> +		snd_soc_write(codec, MAX9850_CLOCK, 0x8);
> +	else
> +		return -EINVAL;
> +
> +	max9850->sysclk = freq;
> +	return 0;
> +}
> +
> +static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
> +{
> +	struct snd_soc_codec *codec = codec_dai->codec;
> +	u8 da = 0;
> +
> +	/* set master/slave audio interface */
> +	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
> +	case SND_SOC_DAIFMT_CBM_CFM:
> +		da |= MAX9850_MASTER;
> +		break;
> +	case SND_SOC_DAIFMT_CBS_CFS:
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	/* interface format */
> +	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
> +	case SND_SOC_DAIFMT_I2S:
> +		da |= MAX9850_DLY;
> +		break;
> +	case SND_SOC_DAIFMT_RIGHT_J:
> +		da |= MAX9850_RTJ;
> +		break;
> +	case SND_SOC_DAIFMT_LEFT_J:
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	/* clock inversion */
> +	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
> +	case SND_SOC_DAIFMT_NB_NF:
> +		break;
> +	case SND_SOC_DAIFMT_IB_IF:
> +		da |= MAX9850_BCINV | MAX9850_INV;
> +		break;
> +	case SND_SOC_DAIFMT_IB_NF:
> +		da |= MAX9850_BCINV;
> +		break;
> +	case SND_SOC_DAIFMT_NB_IF:
> +		da |= MAX9850_INV;
> +		break;
> +	default:
> +		return -EINVAL;
> +	}
> +
> +	/* set da */
> +	snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da);
> +
> +	return 0;
> +}
> +
> +static int max9850_set_bias_level(struct snd_soc_codec *codec,
> +				  enum snd_soc_bias_level level)
> +{
> +	switch (level) {
> +	case SND_SOC_BIAS_ON:
> +		break;
> +	case SND_SOC_BIAS_PREPARE:
> +		snd_soc_update_bits(codec, MAX9850_ENABLE, MAX9850_SHDN,
> +				MAX9850_SHDN);

Could possibly be handled by DAPM?

> +		break;
> +	case SND_SOC_BIAS_STANDBY:
> +		snd_soc_update_bits(codec, MAX9850_ENABLE, MAX9850_SHDN, 0);

Ditto.

> +		break;
> +	case SND_SOC_BIAS_OFF:
> +		break;
> +	}
> +	codec->dapm.bias_level = level;
> +	return 0;
> +}

I don't see any suspend/resume callbacks.  It'd be good if you could
provide default stubs that'd just set the bias level.  Also syncing the
cache when the bias level changes from BIAS_OFF to STANDBY would be a plus.

> +#define MAX9850_RATES SNDRV_PCM_RATE_8000_48000
> +
> +#define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
> +	SNDRV_PCM_FMTBIT_S24_LE)
> +
> +static struct snd_soc_dai_ops max9850_dai_ops = {
> +	.hw_params	= max9850_hw_params,
> +	.set_sysclk	= max9850_set_dai_sysclk,
> +	.set_fmt	= max9850_set_dai_fmt,
> +};
> +
> +static struct snd_soc_dai_driver max9850_dai = {
> +	.name = "max9850-hifi",
> +	.playback = {
> +		.stream_name = "Playback",
> +		.channels_min = 1,
> +		.channels_max = 2,
> +		.rates = MAX9850_RATES,
> +		.formats = MAX9850_FORMATS
> +	},
> +	.ops = &max9850_dai_ops,
> +};
> +
> +static int max9850_probe(struct snd_soc_codec *codec)
> +{
> +	struct snd_soc_dapm_context *dapm = &codec->dapm;
> +	int ret;
> +
> +	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
> +	if (ret < 0) {
> +		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
> +		return ret;
> +	}
> +
> +	/* enable zero-detect */
> +	snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1);
> +	/* enable charge pump, disable everything else */
> +	snd_soc_write(codec, MAX9850_ENABLE, 0x30);

DAPM?

> +	/* enable slew-rate control */
> +	snd_soc_update_bits(codec, MAX9850_VOLUME, 0x40, 0x40);
> +	/* set slew-rate 125ms */
> +	snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0);
> +
> +	snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets,
> +				  ARRAY_SIZE(max9850_dapm_widgets));
> +	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
> +
> +	snd_soc_add_controls(codec, max9850_controls,
> +			ARRAY_SIZE(max9850_controls));
> +
> +	return 0;
> +}
> +static int max9850_remove(struct snd_soc_codec *codec)
> +{
> +	return 0;
> +}

Setting the bias level to OFF would be preferable here.

> +static struct snd_soc_codec_driver soc_codec_dev_max9850 = {
> +	.probe =	max9850_probe,
> +	.remove =	max9850_remove,
> +	.set_bias_level = max9850_set_bias_level,
> +	.reg_cache_size = ARRAY_SIZE(max9850_reg),
> +	.reg_word_size = sizeof(u8),
> +	.reg_cache_default = max9850_reg,
> +	.volatile_register = max9850_volatile_register,
> +};
> +
> +static int __devinit max9850_i2c_probe(struct i2c_client *i2c,
> +		const struct i2c_device_id *id)
> +{
> +	struct max9850_priv *max9850;
> +	int ret;
> +
> +	max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL);
> +	if (max9850 == NULL)
> +		return -ENOMEM;
> +
> +	i2c_set_clientdata(i2c, max9850);
> +
> +	ret = snd_soc_register_codec(&i2c->dev,
> +			&soc_codec_dev_max9850, &max9850_dai, 1);
> +	if (ret < 0)
> +		kfree(max9850);
> +	return ret;
> +}
> +
> +static __devexit int max9850_i2c_remove(struct i2c_client *client)
> +{
> +	snd_soc_unregister_codec(&client->dev);
> +	kfree(i2c_get_clientdata(client));
> +	return 0;
> +}
> +
> +static const struct i2c_device_id max9850_i2c_id[] = {
> +	{ "max9850", 0 },
> +	{ }
> +};
> +MODULE_DEVICE_TABLE(i2c, max9850_i2c_id);
> +
> +static struct i2c_driver max9850_i2c_driver = {
> +	.driver = {
> +		.name = "max9850-codec",

Remove the `-codec'.

> +		.owner = THIS_MODULE,
> +	},
> +	.probe = max9850_i2c_probe,
> +	.remove = __devexit_p(max9850_i2c_remove),
> +	.id_table = max9850_i2c_id,
> +};
> +
> +static int __init max9850_init(void)
> +{
> +	return i2c_add_driver(&max9850_i2c_driver);
> +}
> +module_init(max9850_init);
> +
> +static void __exit max9850_exit(void)
> +{
> +	i2c_del_driver(&max9850_i2c_driver);
> +}
> +module_exit(max9850_exit);
> +
> +MODULE_AUTHOR("Christian Glindkamp <christian.glindkamp at taskit.de>");
> +MODULE_DESCRIPTION("ASoC MAX9850 codec driver");
> +MODULE_LICENSE("GPL");
> diff --git a/sound/soc/codecs/max9850.h b/sound/soc/codecs/max9850.h
> new file mode 100644
> index 0000000..5268575
> --- /dev/null
> +++ b/sound/soc/codecs/max9850.h
> @@ -0,0 +1,41 @@
> +/*
> + * max9850.h  --  codec driver for max9850
> + *
> + * Copyright (C) 2011 taskit GmbH
> + * Author: Christian Glindkamp <christian.glindkamp at taskit.de>
> + *
> + * 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.
> + *
> + */
> +
> +#ifndef _MAX9850_H
> +#define _MAX9850_H
> +
> +#define MAX9850_STATUSA			0x00
> +#define MAX9850_STATUSB			0x01
> +#define MAX9850_VOLUME			0x02
> +#define MAX9850_GENERAL_PURPOSE		0x03
> +#define MAX9850_INTERRUPT		0x04
> +#define MAX9850_ENABLE			0x05
> +#define MAX9850_CLOCK			0x06
> +#define MAX9850_CHARGE_PUMP		0x07
> +#define MAX9850_LRCLK_MSB		0x08
> +#define MAX9850_LRCLK_LSB		0x09
> +#define MAX9850_DIGITAL_AUDIO		0x0a
> +
> +#define MAX9850_CACHEREGNUM 11
> +
> +/* MAX9850_ENABLE */
> +#define MAX9850_SHDN			(1<<7)
> +
> +/* MAX9850_DIGITAL_AUDIO */
> +#define MAX9850_MASTER			(1<<7)
> +#define MAX9850_INV			(1<<6)
> +#define MAX9850_BCINV			(1<<5)
> +#define MAX9850_DLY			(1<<3)
> +#define MAX9850_RTJ			(1<<2)
> +
> +#endif
> -- 
> 1.7.2.3
> 
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