ASoC: Codec for STAC9766 used on the Efika
Jon Smirl [Sat, 23 May 2009 23:13:07 +0000 (19:13 -0400)]
Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

sound/soc/codecs/Kconfig
sound/soc/codecs/Makefile
sound/soc/codecs/stac9766.c [new file with mode: 0644]
sound/soc/codecs/stac9766.h [new file with mode: 0644]

index 7f78b65..cb07d9b 100644 (file)
@@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS
        select SND_SOC_CS4270 if I2C
        select SND_SOC_PCM3008
        select SND_SOC_SSM2602 if I2C
+       select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
        select SND_SOC_TLV320AIC23 if I2C
        select SND_SOC_TLV320AIC26 if SPI_MASTER
        select SND_SOC_TLV320AIC3X if I2C
@@ -93,6 +94,9 @@ config SND_SOC_PCM3008
 config SND_SOC_SSM2602
        tristate
 
+config SND_SOC_STAC9766
+       tristate
+
 config SND_SOC_TLV320AIC23
        tristate
 
index 70c55fa..46c007c 100644 (file)
@@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o
 snd-soc-l3-objs := l3.o
 snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270)  += snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_L3)       += snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_PCM3008)  += snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_SSM2602)  += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)      += snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)      += snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)      += snd-soc-tlv320aic3x.o
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
new file mode 100644 (file)
index 0000000..7740cd5
--- /dev/null
@@ -0,0 +1,470 @@
+/*
+ * 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/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 <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_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),
+};
+
+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;
+}
+
+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 */
+
+       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 ac97_digital_trigger(struct snd_pcm_substream *substream,
+                                                               int cmd, struct snd_soc_dai *dai)
+{
+       struct snd_soc_codec *codec = dai->codec;
+       unsigned short vra;
+
+       switch (cmd) {
+       case SNDRV_PCM_TRIGGER_STOP:
+               vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+               vra &= !0x04;
+               stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+               break;
+       }
+       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 */
+       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->bias_level = level;
+       return 0;
+}
+
+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 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, 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);
+
+       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,
+       .trigger = ac97_digital_trigger,
+};
+
+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 = 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 IEC958",
+       .id = 1,
+       .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,
+}};
+EXPORT_SYMBOL_GPL(stac9766_dai);
+
+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));
+
+       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_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_modinit(void)
+{
+       return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai));
+}
+module_init(stac9766_modinit);
+
+static void __exit stac9766_exit(void)
+{
+       snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai));
+}
+module_exit(stac9766_exit);
+
+MODULE_DESCRIPTION("ASoC stac9766 driver");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h
new file mode 100644 (file)
index 0000000..65642eb
--- /dev/null
@@ -0,0 +1,21 @@
+/*
+ * stac9766.h  --  STAC9766 Soc Audio driver
+ */
+
+#ifndef _STAC9766_H
+#define _STAC9766_H
+
+#define AC97_STAC_PAGE0 0x1000
+#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
+#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
+#define AC97_STAC_STEREO_MIC 0x78
+
+/* STAC9766 DAI ID's */
+#define STAC9766_DAI_AC97_ANALOG               0
+#define STAC9766_DAI_AC97_DIGITAL              1
+
+extern struct snd_soc_dai stac9766_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_stac9766;
+
+
+#endif