[alsa-devel] New codec for STAC9766 used on Efika

Jon Smirl jonsmirl at gmail.com
Thu Apr 30 22:30:12 CEST 2009


Could you please review this new STAC9766 codec implementation. Some
of the controls aren't where they need to be. For example the mutes on
the mixer inputs. I'm not sure where they are supposed to go.

How should SPDIF support work? The hardware supports simultaneous use
of analog and SPDIF.  Do I want one DAI or two?  If I have two DAI
then ALSA needs to interleave the samples into a single DMA stream. So
if one stream isn't active it needs to be filled with silence.

Other options...
A single DAI with a switch
A single DAI with the same output stream on both analog and SPDIF.

I've tried to add the DAPM routes. I don't know of anyone using this
chip on a low power design.

Datasheet is here:
http://www.idt.com/products/getDoc.cfm?docID=13134007

I get a few errors from the current config:
Failed to add route All Analog Mixer->Pop Bypass Mux
dapm: STAC9766: configuring unknown pin OUT3
dapm: STAC9766: configuring unknown pin MONOOUT
Failed to add route HPOUTL->Headphone Jack
dapm: STAC9766: configuring unknown pin OUT3
dapm: STAC9766: configuring unknown pin MONOOUT
Failed to add route HPOUTL->Headphone Jack



/*
 * stac9766.c  --  ALSA Soc STAC9766 codec support
 *
 * Copyright 2008 Jon Smirl, Digispeaker
 * Author: Jon Smirl <jonsmirl at gmail.com>
 *
 *  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.
 *
 *  Features:-
 *
 *   o Support for AC97 Codec, S/PDIF
 *   o Support for DAPM
 */

#include <linux/init.h>
#include <linux/module.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/soc-of-simple.h>

#include "stac9766.h"

#define STAC9766_VERSION "0.10"

/*
 * STAC9766 register cache
 */
static const u16 stac9766_reg[] = {
	0x6A90, 0x8000, 0x8000, 0x8000, // 6
	0x0000, 0x0000, 0x8008, 0x8008, // e
	0x8808, 0x8808, 0x8808, 0x8808, // 16
	0x8808, 0x0000, 0x8000, 0x0000, // 1e
	0x0000, 0x0000, 0x0000, 0x000f, // 26
	0x0a05, 0x0400, 0xbb80, 0x0000, // 2e
	0x0000, 0xbb80, 0x0000, 0x0000, // 36
	0x0000, 0x2000, 0x0000, 0x0100, // 3e
	0x0000, 0x0000, 0x0080, 0x0000, // 46
	0x0000, 0x0000, 0x0003, 0xffff, // 4e
	0x0000, 0x0000, 0x0000, 0x0000, // 56
	0x4000, 0x0000, 0x0000, 0x0000, // 5e
	0x1201, 0xFFFF, 0xFFFF, 0x0000, // 66
	0x0000, 0x0000, 0x0000, 0x0000, // 6e
	0x0000, 0x0000, 0x0000, 0x0006, // 76
	0x0000, 0x0000, 0x0000, 0x0000, // 7e
};

static const char *stac9766_record_mux[] = {"Mic", "CD", "Video",
"AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"};
static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
static const char *stac9766_record_all_mux[] = {"All analog", "Analog
plus DAC"};
static const char *stac9766_3D_separation[] = {"Off", "Low", "Medium", "High"};

static const struct soc_enum stac9766_enum[] = {
	SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux), /* Record Mux 0 */
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux), /*
Mono Mux 1 */
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux), /*
Mic1/2 Mux 2 */
	SOC_ENUM_SINGLE(AC97_SENSE_INFO, 1, 2, stac9766_SPDIF_mux), /* SPDIF Mux 3 */
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux),
/* Pop Bypass Mux 4 */
	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
stac9766_record_all_mux), /* Record All Mux 5 */
	SOC_ENUM_SINGLE(AC97_3D_CONTROL, 2, 4, stac9766_3D_separation), /* 3D
Separation 7 */
};

static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
	SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
	SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
	SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
	SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
	SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
	SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
	SOC_SINGLE("Mixer PC Beep Volume", AC97_PC_BEEP, 1, 15, 1),
	SOC_SINGLE("PC Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
	SOC_SINGLE("Mixer Phone Volume", AC97_PHONE, 0, 31, 1),
	SOC_SINGLE("Mixer Mic Volume", AC97_MIC, 0, 31, 1),
	SOC_SINGLE("Mic Boost", AC97_MIC, 6, 1, 1),
	SOC_SINGLE("Mic Gain", AC97_STAC_ANALOG_SPECIAL, 2, 1, 1),
	SOC_SINGLE("Stereo Mic", AC97_STAC_STEREO_MIC, 0, 1, 1),
	SOC_DOUBLE("Mixer Line Volume", AC97_LINE, 8, 0, 31, 1),
	SOC_DOUBLE("Mixer CD Volume", AC97_CD, 8, 0, 31, 1),
	SOC_DOUBLE("Mixer Video Volume", AC97_VIDEO, 8, 0, 31, 1),
	SOC_DOUBLE("Mixer AUX Volume", AC97_AUX, 8, 0, 31, 1),
	SOC_DOUBLE("All Analog PCM Volume", AC97_PCM, 8, 0, 31, 1),
	SOC_DOUBLE("Record Gain", AC97_REC_GAIN, 8, 0, 31, 1),
	SOC_SINGLE("Record Gain Switch", AC97_REC_GAIN, 15, 1, 1),
	SOC_SINGLE("3D Effect Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
	SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
	SOC_ENUM("3D Separation", stac9766_enum[6]),
};

/* Mixer */
static const struct snd_kcontrol_new stac9766_main_mixer_controls[] = {
	SOC_DAPM_SINGLE("PC Beep Switch", AC97_PC_BEEP, 15, 1, 1),
	SOC_DAPM_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
	SOC_DAPM_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
	SOC_DAPM_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
	SOC_DAPM_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
	SOC_DAPM_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
	SOC_DAPM_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
};

/* Record All Mixer */
static const struct snd_kcontrol_new stac9766_record_all_mixer_controls[] = {
	SOC_DAPM_SINGLE("PCM Switch", AC97_PCM, 15, 1, 1),
	SOC_DAPM_SINGLE("Mixer", 0, 0, 0, 0),
};

/* Record Mux 0 */
static const struct snd_kcontrol_new stac9766_record_mux_controls =
	SOC_DAPM_ENUM("Route", stac9766_enum[0]);

/* Mono Mux 1 */
static const struct snd_kcontrol_new stac9766_mono_mux_controls =
	SOC_DAPM_ENUM("Route", stac9766_enum[1]);

/* Mic1/2 Mux 2 */
static const struct snd_kcontrol_new stac9766_mic_mux_controls =
	SOC_DAPM_ENUM("Route", stac9766_enum[2]);

/* SPDIF Mux 3 */
static const struct snd_kcontrol_new stac9766_spdif_mux_controls =
	SOC_DAPM_ENUM("Route", stac9766_enum[3]);

/* Pop Bypass 4 */
static const struct snd_kcontrol_new stac9766_popbypass_mux_controls =
	SOC_DAPM_ENUM("Route", stac9766_enum[4]);

/* Record All 5 */
static const struct snd_kcontrol_new stac9766_record_all_mux_controls =
	SOC_DAPM_ENUM("Route", stac9766_enum[5]);

static const struct snd_soc_dapm_widget stac9766_dapm_widgets[] = {
	SND_SOC_DAPM_INPUT("PCMOut"),
	SND_SOC_DAPM_INPUT("PCBEEP"),
	SND_SOC_DAPM_INPUT("Phone"),
	SND_SOC_DAPM_INPUT("MIC1"),
	SND_SOC_DAPM_INPUT("MIC2"),
	SND_SOC_DAPM_INPUT("Line"),
	SND_SOC_DAPM_INPUT("CD"),
	SND_SOC_DAPM_INPUT("AUX"),
	SND_SOC_DAPM_INPUT("Video"),
	SND_SOC_DAPM_DAC("DAC", "Analog Playback", AC97_POWERDOWN, 9, 1),
	SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0,
		&stac9766_spdif_mux_controls),
	SND_SOC_DAPM_MUX("Mic1/2 Mux", SND_SOC_NOPM, 0, 0,
		&stac9766_mic_mux_controls),
	SND_SOC_DAPM_MIXER("Mixer", SND_SOC_NOPM, 0, 1,
		&stac9766_main_mixer_controls[0], ARRAY_SIZE(stac9766_main_mixer_controls)),
	SND_SOC_DAPM_MUX("Record All Mux", SND_SOC_NOPM, 0, 0,
		&stac9766_record_all_mux_controls),
	SND_SOC_DAPM_MUX("Record Mux", SND_SOC_NOPM, 0, 0,
		&stac9766_record_mux_controls),
	SND_SOC_DAPM_MUX("Mono Mux", SND_SOC_NOPM, 0, 0,
		&stac9766_mono_mux_controls),
	SND_SOC_DAPM_MUX("Pop Bypass Mux", SND_SOC_NOPM, 0, 0,
		&stac9766_popbypass_mux_controls),
	SND_SOC_DAPM_MIXER("All Analog", SND_SOC_NOPM, 0, 1,
		&stac9766_record_all_mixer_controls[0],
ARRAY_SIZE(stac9766_record_all_mixer_controls)),
	SND_SOC_DAPM_ADC("ADC", "Analog Capture", AC97_POWERDOWN, 8, 1),
	SND_SOC_DAPM_OUTPUT("HP"),
	SND_SOC_DAPM_OUTPUT("LINE"),
	SND_SOC_DAPM_OUTPUT("MONO"),
	SND_SOC_DAPM_OUTPUT("PCMIn"),
	SND_SOC_DAPM_VMID("VMID"),
};

static const struct snd_soc_dapm_route audio_map[] = {
	{"HP", NULL, "Pop Bypass Mux"},
	{"LINE", NULL, "Pop Bypass Mux"},

	/* Mono Mux */
	{"MONO", NULL, "Pop Bypass Mux"},
	{"MONO", NULL, "Mic1/2 Mux"},

	/* Pop Bypass Mux */
	{"Pop Bypass Mux", NULL, "DAC"},
	{"Pop Bypass Mux", NULL, "All Analog Mixer"},

	/* Record Mux */
	{"ADC", NULL, "Mic1/2 Mux"},
	{"ADC", NULL, "CD"},
	{"ADC", NULL, "Video"},
	{"ADC", NULL, "AUX"},
	{"ADC", NULL, "Line"},
	{"ADC", NULL, "Record All Mux"},
	{"ADC", NULL, "Phone"},

	{"PCMIn", NULL, "ADC"},

	/* Record All Mux */
	{"Record All Mux", NULL, "Mixer"},
	{"Record All Mux", NULL, "All Analog Mixer"},

	/* All Analog Mixer */
	{"All Analog", NULL, "Mixer"},
	{"All Analog", "PCM Switch", "DAC"},

	/* Mixer */
	{"Mixer", "PC Beep Switch", "PCBEEP"},
	{"Mixer", "Phone Switch", "Phone"},
	{"Mixer", "Mic Switch", "Mic1/2 Mux"},
	{"Mixer", "Line Switch", "Line"},
	{"Mixer", "CD Switch", "CD"},
	{"Mixer", "AUX Switch", "AUX"},
	{"Mixer", "Video Switch", "Video"},

	{"DAC", NULL, "PCMOut"},

	/* Mic1/2 Mux */
	{"Mic1/2 Mux", NULL, "MIC1"},
	{"Mic1/2 Mux", NULL, "MIC2"},

	/* SPDIF Mux */
	{"SPDIF Mux", NULL, "PCMOut"},
	{"SPDIF Mux", NULL, "ADC"},

	{NULL, NULL, NULL},
};

static int stac9766_add_widgets(struct snd_soc_codec *codec)
{
	snd_soc_dapm_new_controls(codec, stac9766_dapm_widgets,
				  ARRAY_SIZE(stac9766_dapm_widgets));

	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));

	snd_soc_dapm_new_widgets(codec);
	return 0;

}

unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
	unsigned int reg)
{
	u16 val = 0, *cache = codec->reg_cache;

	if (reg / 2 > ARRAY_SIZE(stac9766_reg))
		return -EIO;

	if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || AC97_INT_PAGING ||
		reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) {

		val = soc_ac97_ops.read(codec->ac97, reg);
		return val;
	}
	return cache[reg / 2];
}

int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
	unsigned int val)
{
	u16 *cache = codec->reg_cache;

	if (reg / 2 > ARRAY_SIZE(stac9766_reg))
		return -EIO;

	soc_ac97_ops.write(codec->ac97, reg, val);
	cache[reg / 2] = val;
	return 0;
}

static int ac97_analog_prepare(struct snd_pcm_substream *substream,
			     struct snd_soc_dai *dai)
{
	struct snd_soc_codec *codec = dai->codec;
	struct snd_pcm_runtime *runtime = substream->runtime;
	unsigned short reg, vra;

	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);

	//vra |= 0x4;

	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);

	printk("AC97_EXTENDED_STATUS %x\n", vra);

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		reg = AC97_PCM_FRONT_DAC_RATE;
	else
		reg = AC97_PCM_LR_ADC_RATE;

	return stac9766_ac97_write(codec, reg, runtime->rate);
}

static int ac97_digital_prepare(struct snd_pcm_substream *substream,
			     struct snd_soc_dai *dai)
{
	printk("stac9766: ac97_digital_prepare\n");

	return 0;
}

static int stac9766_set_bias_level(struct snd_soc_codec *codec,
						enum snd_soc_bias_level level)
{
	switch (level) {
	case SND_SOC_BIAS_ON: /* full On */
		stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
		break;
	case SND_SOC_BIAS_PREPARE: /* partial On */
		stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
		break;
	case SND_SOC_BIAS_STANDBY: /* Off, with power */
		stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
		break;
	case SND_SOC_BIAS_OFF: /* Off, without power */
		/* disable everything including AC link */
		stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
		break;
	}
	codec->bias_level = level;
	return 0;
}

static int stac9766_codec_suspend(struct platform_device *pdev,
	pm_message_t state)
{
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_codec *codec = socdev->card->codec;

	stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
	return 0;
}

static int stac9766_codec_resume(struct platform_device *pdev)
{
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_codec *codec = socdev->card->codec;
	u16 id;

	/* give the codec an AC97 warm reset to start the link */
	codec->ac97->bus->ops->warm_reset(codec->ac97);
	id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
	if (id != 0x4c13) {
		printk(KERN_ERR "stac9766 failed to resume");
		return -EIO;
	}
	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
		stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);

	return 0;
}

static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
	.prepare	= ac97_analog_prepare,
};

static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
	.prepare	= ac97_digital_prepare,
};

struct snd_soc_dai stac9766_dai[] = {
{
	.name	= "stac9766 analog",
	.id	= 0,
	.ac97_control = 1,

	/* stream cababilities */
	.playback = {
		.stream_name	= "stac9766 analog",
		.channels_min	= 1,
		.channels_max	= 2,
		.rates		= SNDRV_PCM_RATE_8000_48000,
		.formats	= SNDRV_PCM_FMTBIT_S32_BE,
	},
	.capture = {
		.stream_name	= "stac9766 analog",
		.channels_min	= 1,
		.channels_max	= 2,
		.rates		= SNDRV_PCM_RATE_8000_48000,
		.formats	= SNDRV_PCM_FMTBIT_S32_BE,
	},
	/* alsa ops */
	.ops = &stac9766_dai_ops_analog,
},
{
	.name	= "stac9766 digital",
	.id	= 1,
	.ac97_control = 1,

	/* stream cababilities */
	.playback = {
		.stream_name	= "stac9766 digital",
		.channels_min	= 1,
		.channels_max	= 2,
		.rates		= SNDRV_PCM_RATE_32000 | \
			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
		.formats	= SNDRV_PCM_FMTBIT_IEC958_SUBFRAME,
	},
	/* alsa ops */
	.ops = &stac9766_dai_ops_digital,
}};
EXPORT_SYMBOL_GPL(stac9766_dai);

int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
{
	if (try_warm && soc_ac97_ops.warm_reset) {
		soc_ac97_ops.warm_reset(codec->ac97);
		if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
			return 1;
	}

	soc_ac97_ops.reset(codec->ac97);
	if (soc_ac97_ops.warm_reset)
		soc_ac97_ops.warm_reset(codec->ac97);
	if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
		return -EIO;
	return 0;
}

static int stac9766_codec_probe(struct platform_device *pdev)
{
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_codec *codec;
	int ret = 0;

	printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);

	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
				      GFP_KERNEL);
	if (socdev->card->codec == NULL)
		return -ENOMEM;
	codec = socdev->card->codec;
	mutex_init(&codec->mutex);

	codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL);
	if (codec->reg_cache == NULL) {
		ret = -ENOMEM;
		goto cache_err;
	}
	codec->reg_cache_size = sizeof(stac9766_reg);
	codec->reg_cache_step = 2;

	codec->name = "STAC9766";
	codec->owner = THIS_MODULE;
	codec->dai = stac9766_dai;
	codec->num_dai = ARRAY_SIZE(stac9766_dai);
	codec->write = stac9766_ac97_write;
	codec->read = stac9766_ac97_read;
	codec->set_bias_level = stac9766_set_bias_level;
	INIT_LIST_HEAD(&codec->dapm_widgets);
	INIT_LIST_HEAD(&codec->dapm_paths);

	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
	if (ret < 0)
		goto codec_err;

	/* register pcms */
	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
	if (ret < 0)
		goto pcm_err;

	/* do a cold reset for the controller and then try
	 * a warm reset followed by an optional cold reset for codec */
	stac9766_reset(codec, 0);
	ret = stac9766_reset(codec, 1);
	if (ret < 0) {
		printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
		goto reset_err;
	}

	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

	snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
				ARRAY_SIZE(stac9766_snd_ac97_controls));
	stac9766_add_widgets(codec);
	ret = snd_soc_init_card(socdev);
	if (ret < 0)
		goto reset_err;
	return 0;

reset_err:
	snd_soc_free_pcms(socdev);

pcm_err:
	snd_soc_free_ac97_codec(codec);

codec_err:
	kfree(codec->private_data);

cache_err:
	kfree(socdev->card->codec);
	socdev->card->codec = NULL;
	return ret;
}

static int stac9766_codec_remove(struct platform_device *pdev)
{
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_codec *codec = socdev->card->codec;

	if (codec == NULL)
		return 0;

	snd_soc_dapm_free(socdev);
	snd_soc_free_pcms(socdev);
	snd_soc_free_ac97_codec(codec);
	kfree(codec->reg_cache);
	kfree(codec);
	return 0;
}

struct snd_soc_codec_device soc_codec_dev_stac9766 = {
	.probe = 	stac9766_codec_probe,
	.remove = 	stac9766_codec_remove,
	.suspend =	stac9766_codec_suspend,
	.resume = 	stac9766_codec_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);

static int __init stac9766_probe(struct platform_device *pdev)
{
#if defined(CONFIG_SND_SOC_OF_SIMPLE)
	/* Tell the of_soc helper about this codec */
	of_snd_soc_register_codec(&soc_codec_dev_stac9766, pdev->dev.archdata.of_node,
									stac9766_dai, ARRAY_SIZE(stac9766_dai),
									pdev->dev.archdata.of_node);
#endif
	return 0;
}

static struct platform_driver stac9766_driver = {
	.probe	= stac9766_probe,
	.driver	= {
		.name	= "stac9766",
	},
};

static __init int stac9766_driver_init(void)
{
	snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai));
	return platform_driver_register(&stac9766_driver);
}

static __exit void stac9766_driver_exit(void)
{
}

module_init(stac9766_driver_init);
module_exit(stac9766_driver_exit);

MODULE_DESCRIPTION("ASoC stac9766 driver");
MODULE_AUTHOR("Jon Smirl <jonsmirl at gmail.com>");
MODULE_LICENSE("GPL");


-- 
Jon Smirl
jonsmirl at gmail.com


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