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

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@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@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@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@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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