[alsa-devel] [PATCH] ASoC: Add MAX9850 codec driver
Christian Glindkamp
christian.glindkamp at taskit.de
Mon Mar 7 18:04:20 CET 2011
Resend to list, sorry for the noise Dimitris
As all ready said, I can't really test these changes at the moment but
will at least compile test them. Additionally, alsa master currently
does not build here for the arm architecture. Further comments below.
On 2011-03-07 13:48, Dimitris Papastamos wrote:
> 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.
>
Have not seen this branch. I will use a newer revision of alsa master if
that is ok.
> > +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.
>
Do you mean something like that?
SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0),
SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0,
&max9850_mixer_controls[0],
ARRAY_SIZE(max9850_mixer_controls)),
SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0),
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0),
SND_SOC_DAPM_OUTPUT("OUTL"),
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("OUTR"),
SND_SOC_DAPM_OUTPUT("HPR"),
SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_INPUT("INL"),
SND_SOC_DAPM_INPUT("INR"),
> > +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?
>
Yes, it is a relatively primitive codec. You can only switch on/off line
in/out and the headphones. No routing is possible. Actually, what is
called "Output Mixer" here is in reality line out, but it needs to be
enabled for the headphone outputs to work.
> > +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?
>
I could wire them to be the supply for the "Output Mixer". Without it
being enabled, the codec does nothing anyway.
> > + 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.
>
Will look into it.
> > +#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?
>
Charge pump could also be a supply for the Output Mixer. Have to find
out how to toggle two bits at once via dapm (preferably without
resorting to callbacks).
> > + /* 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
> >
> > _______________________________________________
> > Alsa-devel mailing list
> > Alsa-devel at alsa-project.org
> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
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