sound/soc/codecs/stac9766.c - maze/linux - Git at Google

 /*
  * stac9766.c  --  ALSA SoC STAC9766 codec support
  *
  * Copyright 2009 Jon Smirl, Digispeaker
  * Author: Jon Smirl <jonsmirl@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
  */

 #include <linux/init.h>
 #include <linux/slab.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/tlv.h>

 #include "stac9766.h"

 /*
  * 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_boost1[] = {"0dB", "10dB"};
 static const char *stac9766_boost2[] = {"0dB", "20dB"};
 static const char *stac9766_stereo_mic[] = {"Off", "On"};

 static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum,
 			    AC97_REC_SEL, 8, 0, stac9766_record_mux);
 static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum,
 			    AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux);
 static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum,
 			    AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux);
 static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum,
 			    AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux);
 static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum,
 			    AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux);
 static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum,
 			    AC97_STAC_ANALOG_SPECIAL, 12,
 			    stac9766_record_all_mux);
 static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum,
 			    AC97_MIC, 6, stac9766_boost1); /* 0/10dB */
 static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum,
 			    AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */
 static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum,
 			    AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic);

 static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
 static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
 static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
 static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);

 static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
 	SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
 	SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
 	SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
 		       master_tlv),
 	SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
 	SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
 		       master_tlv),
 	SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),

 	SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
 	SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),


 	SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
 	SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
 	SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
 	SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
 	SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),

 	SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
 	SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
 	SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
 	SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
 	SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),

 	SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
 	SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
 	SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
 	SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
 	SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
 	SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
 	SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
 	SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),

 	SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
 	SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
 	SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
 	SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
 	SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),

 	SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
 	SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
 	SOC_ENUM("Record All Mux", stac9766_record_all_enum),
 	SOC_ENUM("Record Mux", stac9766_record_enum),
 	SOC_ENUM("Mono Mux", stac9766_mono_enum),
 	SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
 };

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

 	if (reg > AC97_STAC_PAGE0) {
 		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
 		soc_ac97_ops->write(codec->ac97, reg, val);
 		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
 		return 0;
 	}
 	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
 		return -EIO;

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

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

 	if (reg > AC97_STAC_PAGE0) {
 		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
 		val = soc_ac97_ops->read(codec->ac97, reg - AC97_STAC_PAGE0);
 		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
 		return val;
 	}
 	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
 		return -EIO;

 	if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
 		reg == 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];
 }

 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 |= 0x1; /* enable variable rate audio */
 	vra &= ~0x4; /* disable SPDIF output */

 	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, 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)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned short reg, vra;

 	stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);

 	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
 	vra |= 0x5; /* Enable VRA and SPDIF out */

 	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);

 	reg = AC97_PCM_FRONT_DAC_RATE;

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

 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 */
 	case SND_SOC_BIAS_PREPARE: /* partial On */
 	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->dapm.bias_level = level;
 	return 0;
 }

 static 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_suspend(struct snd_soc_codec *codec)
 {
 	stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }

 static int stac9766_codec_resume(struct snd_soc_codec *codec)
 {
 	u16 id, reset;

 	reset = 0;
 	/* give the codec an AC97 warm reset to start the link */
 reset:
 	if (reset > 5) {
 		printk(KERN_ERR "stac9766 failed to resume");
 		return -EIO;
 	}
 	codec->ac97->bus->ops->warm_reset(codec->ac97);
 	id = soc_ac97_ops->read(codec->ac97, AC97_VENDOR_ID2);
 	if (id != 0x4c13) {
 		stac9766_reset(codec, 0);
 		reset++;
 		goto reset;
 	}
 	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

 	return 0;
 }

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

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

 static struct snd_soc_dai_driver stac9766_dai[] = {
 {
 	.name = "stac9766-hifi-analog",
 	.ac97_control = 1,

 	/* stream cababilities */
 	.playback = {
 		.stream_name = "stac9766 analog",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SND_SOC_STD_AC97_FMTS,
 	},
 	.capture = {
 		.stream_name = "stac9766 analog",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SND_SOC_STD_AC97_FMTS,
 	},
 	/* alsa ops */
 	.ops = &stac9766_dai_ops_analog,
 },
 {
 	.name = "stac9766-hifi-IEC958",
 	.ac97_control = 1,

 	/* stream cababilities */
 	.playback = {
 		.stream_name = "stac9766 IEC958",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_32000 | \
 			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
 	},
 	/* alsa ops */
 	.ops = &stac9766_dai_ops_digital,
 }
 };

 static int stac9766_codec_probe(struct snd_soc_codec *codec)
 {
 	int ret = 0;

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

 	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

 	snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls,
 			     ARRAY_SIZE(stac9766_snd_ac97_controls));

 	return 0;

 codec_err:
 	snd_soc_free_ac97_codec(codec);
 	return ret;
 }

 static int stac9766_codec_remove(struct snd_soc_codec *codec)
 {
 	snd_soc_free_ac97_codec(codec);
 	return 0;
 }

 static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
 	.write = stac9766_ac97_write,
 	.read = stac9766_ac97_read,
 	.set_bias_level = stac9766_set_bias_level,
 	.probe = stac9766_codec_probe,
 	.remove = stac9766_codec_remove,
 	.suspend = stac9766_codec_suspend,
 	.resume = stac9766_codec_resume,
 	.reg_cache_size = ARRAY_SIZE(stac9766_reg),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_step = 2,
 	.reg_cache_default = stac9766_reg,
 };

 static int stac9766_probe(struct platform_device *pdev)
 {
 	return snd_soc_register_codec(&pdev->dev,
 			&soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
 }

 static int stac9766_remove(struct platform_device *pdev)
 {
 	snd_soc_unregister_codec(&pdev->dev);
 	return 0;
 }

 static struct platform_driver stac9766_codec_driver = {
 	.driver = {
 			.name = "stac9766-codec",
 			.owner = THIS_MODULE,
 	},

 	.probe = stac9766_probe,
 	.remove = stac9766_remove,
 };

 module_platform_driver(stac9766_codec_driver);

 MODULE_DESCRIPTION("ASoC stac9766 driver");
 MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
 MODULE_LICENSE("GPL");
	/*
	* stac9766.c -- ALSA SoC STAC9766 codec support
	*
	* Copyright 2009 Jon Smirl, Digispeaker
	* Author: Jon Smirl <jonsmirl@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
	*/

	#include <linux/init.h>
	#include <linux/slab.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/tlv.h>

	#include "stac9766.h"

	/*
	* 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_boost1[] = {"0dB", "10dB"};
	static const char *stac9766_boost2[] = {"0dB", "20dB"};
	static const char *stac9766_stereo_mic[] = {"Off", "On"};

	static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum,
	AC97_REC_SEL, 8, 0, stac9766_record_mux);
	static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum,
	AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux);
	static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum,
	AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux);
	static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum,
	AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux);
	static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum,
	AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux);
	static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum,
	AC97_STAC_ANALOG_SPECIAL, 12,
	stac9766_record_all_mux);
	static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum,
	AC97_MIC, 6, stac9766_boost1); /* 0/10dB */
	static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum,
	AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */
	static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum,
	AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic);

	static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
	static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
	static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
	static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);

	static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
	SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
	SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
	SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
	master_tlv),
	SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
	SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
	master_tlv),
	SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),

	SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
	SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),


	SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
	SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
	SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
	SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
	SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),

	SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
	SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
	SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
	SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
	SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),

	SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
	SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
	SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
	SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),

	SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
	SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
	SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
	SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),

	SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
	SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
	SOC_ENUM("Record All Mux", stac9766_record_all_enum),
	SOC_ENUM("Record Mux", stac9766_record_enum),
	SOC_ENUM("Mono Mux", stac9766_mono_enum),
	SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
	};

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

	if (reg > AC97_STAC_PAGE0) {
	stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
	soc_ac97_ops->write(codec->ac97, reg, val);
	stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
	return 0;
	}
	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
	return -EIO;

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

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

	if (reg > AC97_STAC_PAGE0) {
	stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
	val = soc_ac97_ops->read(codec->ac97, reg - AC97_STAC_PAGE0);
	stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
	return val;
	}
	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
	return -EIO;

	if (reg == AC97_RESET \|\| reg == AC97_GPIO_STATUS \|\|
	reg == 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];
	}

	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 \|= 0x1; /* enable variable rate audio */
	vra &= ~0x4; /* disable SPDIF output */

	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, 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)
	{
	struct snd_soc_codec *codec = dai->codec;
	struct snd_pcm_runtime *runtime = substream->runtime;
	unsigned short reg, vra;

	stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);

	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
	vra \|= 0x5; /* Enable VRA and SPDIF out */

	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);

	reg = AC97_PCM_FRONT_DAC_RATE;

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

	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 */
	case SND_SOC_BIAS_PREPARE: /* partial On */
	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->dapm.bias_level = level;
	return 0;
	}

	static 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_suspend(struct snd_soc_codec *codec)
	{
	stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
	return 0;
	}

	static int stac9766_codec_resume(struct snd_soc_codec *codec)
	{
	u16 id, reset;

	reset = 0;
	/* give the codec an AC97 warm reset to start the link */
	reset:
	if (reset > 5) {
	printk(KERN_ERR "stac9766 failed to resume");
	return -EIO;
	}
	codec->ac97->bus->ops->warm_reset(codec->ac97);
	id = soc_ac97_ops->read(codec->ac97, AC97_VENDOR_ID2);
	if (id != 0x4c13) {
	stac9766_reset(codec, 0);
	reset++;
	goto reset;
	}
	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

	return 0;
	}

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

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

	static struct snd_soc_dai_driver stac9766_dai[] = {
	{
	.name = "stac9766-hifi-analog",
	.ac97_control = 1,

	/* stream cababilities */
	.playback = {
	.stream_name = "stac9766 analog",
	.channels_min = 1,
	.channels_max = 2,
	.rates = SNDRV_PCM_RATE_8000_48000,
	.formats = SND_SOC_STD_AC97_FMTS,
	},
	.capture = {
	.stream_name = "stac9766 analog",
	.channels_min = 1,
	.channels_max = 2,
	.rates = SNDRV_PCM_RATE_8000_48000,
	.formats = SND_SOC_STD_AC97_FMTS,
	},
	/* alsa ops */
	.ops = &stac9766_dai_ops_analog,
	},
	{
	.name = "stac9766-hifi-IEC958",
	.ac97_control = 1,

	/* stream cababilities */
	.playback = {
	.stream_name = "stac9766 IEC958",
	.channels_min = 1,
	.channels_max = 2,
	.rates = SNDRV_PCM_RATE_32000 \| \
	SNDRV_PCM_RATE_44100 \| SNDRV_PCM_RATE_48000,
	.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
	},
	/* alsa ops */
	.ops = &stac9766_dai_ops_digital,
	}
	};

	static int stac9766_codec_probe(struct snd_soc_codec *codec)
	{
	int ret = 0;

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

	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

	snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls,
	ARRAY_SIZE(stac9766_snd_ac97_controls));

	return 0;

	codec_err:
	snd_soc_free_ac97_codec(codec);
	return ret;
	}

	static int stac9766_codec_remove(struct snd_soc_codec *codec)
	{
	snd_soc_free_ac97_codec(codec);
	return 0;
	}

	static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
	.write = stac9766_ac97_write,
	.read = stac9766_ac97_read,
	.set_bias_level = stac9766_set_bias_level,
	.probe = stac9766_codec_probe,
	.remove = stac9766_codec_remove,
	.suspend = stac9766_codec_suspend,
	.resume = stac9766_codec_resume,
	.reg_cache_size = ARRAY_SIZE(stac9766_reg),
	.reg_word_size = sizeof(u16),
	.reg_cache_step = 2,
	.reg_cache_default = stac9766_reg,
	};

	static int stac9766_probe(struct platform_device *pdev)
	{
	return snd_soc_register_codec(&pdev->dev,
	&soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
	}

	static int stac9766_remove(struct platform_device *pdev)
	{
	snd_soc_unregister_codec(&pdev->dev);
	return 0;
	}

	static struct platform_driver stac9766_codec_driver = {
	.driver = {
	.name = "stac9766-codec",
	.owner = THIS_MODULE,
	},

	.probe = stac9766_probe,
	.remove = stac9766_remove,
	};

	module_platform_driver(stac9766_codec_driver);

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