[linux-2.6-block.git] / sound / soc / codecs / stac9766.c

/*
 * 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/soc-dapm.h>
#include <sound/tlv.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_boost1[] = {"0dB", "10dB"};
static const char *stac9766_boost2[] = {"0dB", "20dB"};
static const char *stac9766_stereo_mic[] = {"Off", "On"};

static const struct soc_enum stac9766_record_enum =
	SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
static const struct soc_enum stac9766_mono_enum =
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
static const struct soc_enum stac9766_mic_enum =
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
static const struct soc_enum stac9766_SPDIF_enum =
	SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
static const struct soc_enum stac9766_popbypass_enum =
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
static const struct soc_enum stac9766_record_all_enum =
	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
			stac9766_record_all_mux);
static const struct soc_enum stac9766_boost1_enum =
	SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
static const struct soc_enum stac9766_boost2_enum =
	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
static const struct soc_enum stac9766_stereo_mic_enum =
	SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, 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,
				  pm_message_t state)
{
	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 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,
};

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;

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

	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_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 = sizeof(stac9766_reg),
	.reg_word_size = sizeof(u16),
	.reg_cache_step = 2,
};

static __devinit 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 __devexit 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 = __devexit_p(stac9766_remove),
};

static int __init stac9766_init(void)
{
	return platform_driver_register(&stac9766_codec_driver);
}
module_init(stac9766_init);

static void __exit stac9766_exit(void)
{
	platform_driver_unregister(&stac9766_codec_driver);
}
module_exit(stac9766_exit);

MODULE_DESCRIPTION("ASoC stac9766 driver");
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_LICENSE("GPL");
Commit	Line	Data
3c166c7f JS	1	/*
	2	* stac9766.c -- ALSA SoC STAC9766 codec support
	3	*
	4	* Copyright 2009 Jon Smirl, Digispeaker
	5	* Author: Jon Smirl <jonsmirl@gmail.com>
	6	*
	7	* This program is free software; you can redistribute it and/or modify it
	8	* under the terms of the GNU General Public License as published by the
	9	* Free Software Foundation; either version 2 of the License, or (at your
	10	* option) any later version.
	11	*
	12	* Features:-
	13	*
	14	* o Support for AC97 Codec, S/PDIF
	15	*/
	16
	17	#include <linux/init.h>
5a0e3ad6	18	#include <linux/slab.h>
3c166c7f JS	19	#include <linux/module.h>
	20	#include <linux/device.h>
	21	#include <sound/core.h>
	22	#include <sound/pcm.h>
	23	#include <sound/ac97_codec.h>
	24	#include <sound/initval.h>
	25	#include <sound/pcm_params.h>
	26	#include <sound/soc.h>
ce6120cc	27	#include <sound/soc-dapm.h>
3c166c7f	28	#include <sound/tlv.h>
3c166c7f JS	29
	30	#include "stac9766.h"
	31
	32	#define STAC9766_VERSION "0.10"
	33
	34	/*
	35	* STAC9766 register cache
	36	*/
	37	static const u16 stac9766_reg[] = {
	38	0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
	39	0x0000, 0x0000, 0x8008, 0x8008, /* e */
	40	0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
	41	0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
	42	0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
	43	0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
	44	0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
	45	0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
	46	0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
	47	0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
	48	0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
	49	0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
	50	0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
	51	0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
	52	0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
	53	0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
	54	};
	55
05e1efa2 MB	56	static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
05e1efa2 MB	57	"Line", "Stereo Mix", "Mono Mix", "Phone"};
3c166c7f JS	58	static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
	59	static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
	60	static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
	61	static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
05e1efa2 MB	62	static const char *stac9766_record_all_mux[] = {"All analog",
05e1efa2 MB	63	"Analog plus DAC"};
3c166c7f JS	64	static const char *stac9766_boost1[] = {"0dB", "10dB"};
	65	static const char *stac9766_boost2[] = {"0dB", "20dB"};
	66	static const char *stac9766_stereo_mic[] = {"Off", "On"};
	67
	68	static const struct soc_enum stac9766_record_enum =
	69	SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
	70	static const struct soc_enum stac9766_mono_enum =
	71	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
	72	static const struct soc_enum stac9766_mic_enum =
	73	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
	74	static const struct soc_enum stac9766_SPDIF_enum =
	75	SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
	76	static const struct soc_enum stac9766_popbypass_enum =
	77	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
	78	static const struct soc_enum stac9766_record_all_enum =
05e1efa2 MB	79	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
05e1efa2 MB	80	stac9766_record_all_mux);
3c166c7f JS	81	static const struct soc_enum stac9766_boost1_enum =
	82	SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
	83	static const struct soc_enum stac9766_boost2_enum =
	84	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
	85	static const struct soc_enum stac9766_stereo_mic_enum =
	86	SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
	87
	88	static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
	89	static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
	90	static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
	91	static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
	92
	93	static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
	94	SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
	95	SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
05e1efa2 MB	96	SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
05e1efa2 MB	97	master_tlv),
3c166c7f	98	SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
05e1efa2 MB	99	SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
05e1efa2 MB	100	master_tlv),
3c166c7f JS	101	SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
	102
	103	SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
	104	SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
	105
	106
	107	SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
	108	SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
	109	SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
	110	SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
	111	SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
	112
	113	SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
	114	SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
	115	SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
	116	SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
	117	SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
	118
	119	SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
	120	SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
	121	SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
	122	SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
	123	SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
	124	SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
	125	SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
	126	SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
	127
	128	SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
	129	SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
	130	SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
	131	SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
	132	SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
	133
	134	SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
	135	SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
	136	SOC_ENUM("Record All Mux", stac9766_record_all_enum),
	137	SOC_ENUM("Record Mux", stac9766_record_enum),
	138	SOC_ENUM("Mono Mux", stac9766_mono_enum),
	139	SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
	140	};
	141
05e1efa2 MB	142	static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
05e1efa2 MB	143	unsigned int val)
3c166c7f JS	144	{
	145	u16 *cache = codec->reg_cache;
	146
	147	if (reg > AC97_STAC_PAGE0) {
	148	stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
	149	soc_ac97_ops.write(codec->ac97, reg, val);
	150	stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
	151	return 0;
	152	}
33e319fb	153	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
3c166c7f JS	154	return -EIO;
	155
	156	soc_ac97_ops.write(codec->ac97, reg, val);
	157	cache[reg / 2] = val;
	158	return 0;
	159	}
	160
05e1efa2 MB	161	static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
05e1efa2 MB	162	unsigned int reg)
3c166c7f JS	163	{
	164	u16 val = 0, *cache = codec->reg_cache;
	165
	166	if (reg > AC97_STAC_PAGE0) {
	167	stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
	168	val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
	169	stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
	170	return val;
	171	}
33e319fb	172	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
3c166c7f JS	173	return -EIO;
	174
	175	if (reg == AC97_RESET \|\| reg == AC97_GPIO_STATUS \|\|
	176	reg == AC97_INT_PAGING \|\| reg == AC97_VENDOR_ID1 \|\|
	177	reg == AC97_VENDOR_ID2) {
	178
	179	val = soc_ac97_ops.read(codec->ac97, reg);
	180	return val;
	181	}
	182	return cache[reg / 2];
	183	}
	184
	185	static int ac97_analog_prepare(struct snd_pcm_substream *substream,
05e1efa2	186	struct snd_soc_dai *dai)
3c166c7f JS	187	{
	188	struct snd_soc_codec *codec = dai->codec;
	189	struct snd_pcm_runtime *runtime = substream->runtime;
	190	unsigned short reg, vra;
	191
	192	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
	193
	194	vra \|= 0x1; /* enable variable rate audio */
75b46c13	195	vra &= ~0x4; /* disable SPDIF output */
3c166c7f JS	196
	197	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
	198
	199	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
	200	reg = AC97_PCM_FRONT_DAC_RATE;
	201	else
	202	reg = AC97_PCM_LR_ADC_RATE;
	203
	204	return stac9766_ac97_write(codec, reg, runtime->rate);
	205	}
	206
	207	static int ac97_digital_prepare(struct snd_pcm_substream *substream,
05e1efa2	208	struct snd_soc_dai *dai)
3c166c7f JS	209	{
	210	struct snd_soc_codec *codec = dai->codec;
	211	struct snd_pcm_runtime *runtime = substream->runtime;
	212	unsigned short reg, vra;
	213
	214	stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
	215
	216	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
	217	vra \|= 0x5; /* Enable VRA and SPDIF out */
	218
	219	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
	220
	221	reg = AC97_PCM_FRONT_DAC_RATE;
	222
	223	return stac9766_ac97_write(codec, reg, runtime->rate);
	224	}
	225
3c166c7f	226	static int stac9766_set_bias_level(struct snd_soc_codec *codec,
05e1efa2	227	enum snd_soc_bias_level level)
3c166c7f JS	228	{
	229	switch (level) {
	230	case SND_SOC_BIAS_ON: /* full On */
	231	case SND_SOC_BIAS_PREPARE: /* partial On */
	232	case SND_SOC_BIAS_STANDBY: /* Off, with power */
	233	stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
	234	break;
	235	case SND_SOC_BIAS_OFF: /* Off, without power */
	236	/* disable everything including AC link */
	237	stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
	238	break;
	239	}
ce6120cc	240	codec->dapm.bias_level = level;
3c166c7f JS	241	return 0;
	242	}
	243
05e1efa2	244	static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
3c166c7f JS	245	{
	246	if (try_warm && soc_ac97_ops.warm_reset) {
	247	soc_ac97_ops.warm_reset(codec->ac97);
	248	if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
	249	return 1;
	250	}
	251
	252	soc_ac97_ops.reset(codec->ac97);
	253	if (soc_ac97_ops.warm_reset)
	254	soc_ac97_ops.warm_reset(codec->ac97);
	255	if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
	256	return -EIO;
	257	return 0;
	258	}
	259
f0fba2ad	260	static int stac9766_codec_suspend(struct snd_soc_codec *codec,
05e1efa2	261	pm_message_t state)
3c166c7f	262	{
3c166c7f JS	263	stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
	264	return 0;
	265	}
	266
f0fba2ad	267	static int stac9766_codec_resume(struct snd_soc_codec *codec)
3c166c7f	268	{
3c166c7f JS	269	u16 id, reset;
	270
	271	reset = 0;
	272	/* give the codec an AC97 warm reset to start the link */
	273	reset:
	274	if (reset > 5) {
	275	printk(KERN_ERR "stac9766 failed to resume");
	276	return -EIO;
	277	}
	278	codec->ac97->bus->ops->warm_reset(codec->ac97);
	279	id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
	280	if (id != 0x4c13) {
	281	stac9766_reset(codec, 0);
	282	reset++;
	283	goto reset;
	284	}
	285	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
	286
3c166c7f JS	287	return 0;
	288	}
	289
05e1efa2	290	static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
3c166c7f JS	291	.prepare = ac97_analog_prepare,
	292	};
	293
05e1efa2	294	static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
3c166c7f	295	.prepare = ac97_digital_prepare,
3c166c7f JS	296	};
3c166c7f JS	297
f0fba2ad	298	static struct snd_soc_dai_driver stac9766_dai[] = {
3c166c7f	299	{
f0fba2ad	300	.name = "stac9766-hifi-analog",
3c166c7f JS	301	.ac97_control = 1,
	302
	303	/* stream cababilities */
	304	.playback = {
	305	.stream_name = "stac9766 analog",
	306	.channels_min = 1,
	307	.channels_max = 2,
	308	.rates = SNDRV_PCM_RATE_8000_48000,
	309	.formats = SND_SOC_STD_AC97_FMTS,
	310	},
	311	.capture = {
	312	.stream_name = "stac9766 analog",
	313	.channels_min = 1,
	314	.channels_max = 2,
	315	.rates = SNDRV_PCM_RATE_8000_48000,
	316	.formats = SND_SOC_STD_AC97_FMTS,
	317	},
	318	/* alsa ops */
	319	.ops = &stac9766_dai_ops_analog,
	320	},
	321	{
f0fba2ad	322	.name = "stac9766-hifi-IEC958",
3c166c7f JS	323	.ac97_control = 1,
	324
	325	/* stream cababilities */
	326	.playback = {
	327	.stream_name = "stac9766 IEC958",
	328	.channels_min = 1,
	329	.channels_max = 2,
	330	.rates = SNDRV_PCM_RATE_32000 \| \
	331	SNDRV_PCM_RATE_44100 \| SNDRV_PCM_RATE_48000,
	332	.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
	333	},
	334	/* alsa ops */
	335	.ops = &stac9766_dai_ops_digital,
05e1efa2 MB	336	}
05e1efa2 MB	337	};
3c166c7f	338
f0fba2ad	339	static int stac9766_codec_probe(struct snd_soc_codec *codec)
3c166c7f	340	{
3c166c7f JS	341	int ret = 0;
	342
	343	printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
	344
3c166c7f JS	345	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
	346	if (ret < 0)
	347	goto codec_err;
	348
3c166c7f JS	349	/* do a cold reset for the controller and then try
	350	* a warm reset followed by an optional cold reset for codec */
	351	stac9766_reset(codec, 0);
	352	ret = stac9766_reset(codec, 1);
	353	if (ret < 0) {
	354	printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
f0fba2ad	355	goto codec_err;
3c166c7f JS	356	}
	357
	358	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
	359
05e1efa2 MB	360	snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
05e1efa2 MB	361	ARRAY_SIZE(stac9766_snd_ac97_controls));
3c166c7f	362
3c166c7f JS	363	return 0;
3c166c7f JS	364
3c166c7f	365	codec_err:
f0fba2ad	366	snd_soc_free_ac97_codec(codec);
3c166c7f JS	367	return ret;
	368	}
	369
f0fba2ad	370	static int stac9766_codec_remove(struct snd_soc_codec *codec)
3c166c7f	371	{
3c166c7f	372	snd_soc_free_ac97_codec(codec);
3c166c7f JS	373	return 0;
	374	}
	375
f0fba2ad LG	376	static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
	377	.write = stac9766_ac97_write,
	378	.read = stac9766_ac97_read,
	379	.set_bias_level = stac9766_set_bias_level,
3c166c7f JS	380	.probe = stac9766_codec_probe,
	381	.remove = stac9766_codec_remove,
	382	.suspend = stac9766_codec_suspend,
	383	.resume = stac9766_codec_resume,
f0fba2ad LG	384	.reg_cache_size = sizeof(stac9766_reg),
	385	.reg_word_size = sizeof(u16),
	386	.reg_cache_step = 2,
	387	};
	388
	389	static __devinit int stac9766_probe(struct platform_device *pdev)
	390	{
	391	return snd_soc_register_codec(&pdev->dev,
	392	&soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
	393	}
	394
	395	static int __devexit stac9766_remove(struct platform_device *pdev)
	396	{
	397	snd_soc_unregister_codec(&pdev->dev);
	398	return 0;
	399	}
	400
	401	static struct platform_driver stac9766_codec_driver = {
	402	.driver = {
	403	.name = "stac9766-codec",
	404	.owner = THIS_MODULE,
	405	},
	406
	407	.probe = stac9766_probe,
	408	.remove = __devexit_p(stac9766_remove),
3c166c7f	409	};
f0fba2ad LG	410
	411	static int __init stac9766_init(void)
	412	{
	413	return platform_driver_register(&stac9766_codec_driver);
	414	}
	415	module_init(stac9766_init);
	416
	417	static void __exit stac9766_exit(void)
	418	{
	419	platform_driver_unregister(&stac9766_codec_driver);
	420	}
	421	module_exit(stac9766_exit);
3c166c7f	422
3c166c7f JS	423	MODULE_DESCRIPTION("ASoC stac9766 driver");
	424	MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
	425	MODULE_LICENSE("GPL");