mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-12-18 20:16:54 +07:00
7361fbeaea
AC97 devices need to be initially reset before they can be used. Currently each driver does this on its own. Add support for resetting the device to core in snd_soc_new_ac97_codec(). If the caller supplies a device ID and device ID mask the function will reset the device and verify that it has the correct ID, if it does not a error is returned. This will allow to remove custom code with similar functionality from individual drivers. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
407 lines
12 KiB
C
407 lines
12 KiB
C
/*
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* stac9766.c -- ALSA SoC STAC9766 codec support
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*
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* Copyright 2009 Jon Smirl, Digispeaker
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* Author: Jon Smirl <jonsmirl@gmail.com>
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU General Public License as published by the
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* Free Software Foundation; either version 2 of the License, or (at your
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* option) any later version.
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*
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* Features:-
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*
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* o Support for AC97 Codec, S/PDIF
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*/
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#include <linux/init.h>
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#include <linux/slab.h>
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#include <linux/module.h>
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#include <linux/device.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/ac97_codec.h>
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#include <sound/initval.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include <sound/tlv.h>
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#include "stac9766.h"
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/*
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* STAC9766 register cache
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*/
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static const u16 stac9766_reg[] = {
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0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
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0x0000, 0x0000, 0x8008, 0x8008, /* e */
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0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
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0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
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0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
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0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
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0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
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0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
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0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
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0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
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0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
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0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
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0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
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0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
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0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
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0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
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};
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static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
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"Line", "Stereo Mix", "Mono Mix", "Phone"};
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static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
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static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
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static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
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static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
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static const char *stac9766_record_all_mux[] = {"All analog",
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"Analog plus DAC"};
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static const char *stac9766_boost1[] = {"0dB", "10dB"};
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static const char *stac9766_boost2[] = {"0dB", "20dB"};
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static const char *stac9766_stereo_mic[] = {"Off", "On"};
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static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum,
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AC97_REC_SEL, 8, 0, stac9766_record_mux);
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static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum,
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AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux);
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static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum,
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AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux);
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static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum,
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AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux);
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static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum,
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AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux);
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static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum,
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AC97_STAC_ANALOG_SPECIAL, 12,
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stac9766_record_all_mux);
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static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum,
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AC97_MIC, 6, stac9766_boost1); /* 0/10dB */
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static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum,
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AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */
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static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum,
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AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic);
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static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
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static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
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static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
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static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
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static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
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SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
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SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
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SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
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master_tlv),
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SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
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SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
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master_tlv),
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SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
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SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
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SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
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SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
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SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
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SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
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SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
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SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
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SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
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SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
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SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
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SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
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SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
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SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
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SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
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SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
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SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
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SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
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SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
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SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
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SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
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SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
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SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
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SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
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SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
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SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
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SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
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SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
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SOC_ENUM("Record All Mux", stac9766_record_all_enum),
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SOC_ENUM("Record Mux", stac9766_record_enum),
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SOC_ENUM("Mono Mux", stac9766_mono_enum),
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SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
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};
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static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
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unsigned int val)
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{
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struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
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u16 *cache = codec->reg_cache;
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if (reg > AC97_STAC_PAGE0) {
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stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
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soc_ac97_ops->write(ac97, reg, val);
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stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
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return 0;
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}
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if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
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return -EIO;
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soc_ac97_ops->write(ac97, reg, val);
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cache[reg / 2] = val;
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return 0;
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}
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static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
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unsigned int reg)
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{
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struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
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u16 val = 0, *cache = codec->reg_cache;
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if (reg > AC97_STAC_PAGE0) {
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stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
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val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0);
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stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
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return val;
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}
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if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
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return -EIO;
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if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
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reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
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reg == AC97_VENDOR_ID2) {
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val = soc_ac97_ops->read(ac97, reg);
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return val;
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}
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return cache[reg / 2];
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}
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static int ac97_analog_prepare(struct snd_pcm_substream *substream,
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struct snd_soc_dai *dai)
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{
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struct snd_soc_codec *codec = dai->codec;
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struct snd_pcm_runtime *runtime = substream->runtime;
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unsigned short reg, vra;
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vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
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vra |= 0x1; /* enable variable rate audio */
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vra &= ~0x4; /* disable SPDIF output */
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stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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reg = AC97_PCM_FRONT_DAC_RATE;
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else
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reg = AC97_PCM_LR_ADC_RATE;
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return stac9766_ac97_write(codec, reg, runtime->rate);
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}
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static int ac97_digital_prepare(struct snd_pcm_substream *substream,
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struct snd_soc_dai *dai)
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{
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struct snd_soc_codec *codec = dai->codec;
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struct snd_pcm_runtime *runtime = substream->runtime;
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unsigned short reg, vra;
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stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
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vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
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vra |= 0x5; /* Enable VRA and SPDIF out */
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stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
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reg = AC97_PCM_FRONT_DAC_RATE;
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return stac9766_ac97_write(codec, reg, runtime->rate);
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}
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static int stac9766_set_bias_level(struct snd_soc_codec *codec,
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enum snd_soc_bias_level level)
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{
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switch (level) {
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case SND_SOC_BIAS_ON: /* full On */
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case SND_SOC_BIAS_PREPARE: /* partial On */
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case SND_SOC_BIAS_STANDBY: /* Off, with power */
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stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
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break;
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case SND_SOC_BIAS_OFF: /* Off, without power */
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/* disable everything including AC link */
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stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
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break;
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}
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return 0;
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}
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static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
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{
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struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
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if (try_warm && soc_ac97_ops->warm_reset) {
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soc_ac97_ops->warm_reset(ac97);
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if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
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return 1;
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}
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soc_ac97_ops->reset(ac97);
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if (soc_ac97_ops->warm_reset)
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soc_ac97_ops->warm_reset(ac97);
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if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
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return -EIO;
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return 0;
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}
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static int stac9766_codec_resume(struct snd_soc_codec *codec)
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{
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struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
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u16 id, reset;
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reset = 0;
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/* give the codec an AC97 warm reset to start the link */
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reset:
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if (reset > 5) {
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dev_err(codec->dev, "Failed to resume\n");
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return -EIO;
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}
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ac97->bus->ops->warm_reset(ac97);
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id = soc_ac97_ops->read(ac97, AC97_VENDOR_ID2);
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if (id != 0x4c13) {
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stac9766_reset(codec, 0);
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reset++;
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goto reset;
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}
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return 0;
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}
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static const struct snd_soc_dai_ops stac9766_dai_ops_analog = {
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.prepare = ac97_analog_prepare,
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};
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static const struct snd_soc_dai_ops stac9766_dai_ops_digital = {
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.prepare = ac97_digital_prepare,
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};
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static struct snd_soc_dai_driver stac9766_dai[] = {
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{
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.name = "stac9766-hifi-analog",
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/* stream cababilities */
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.playback = {
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.stream_name = "stac9766 analog",
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.channels_min = 1,
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.channels_max = 2,
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.rates = SNDRV_PCM_RATE_8000_48000,
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.formats = SND_SOC_STD_AC97_FMTS,
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},
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.capture = {
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.stream_name = "stac9766 analog",
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.channels_min = 1,
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.channels_max = 2,
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.rates = SNDRV_PCM_RATE_8000_48000,
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.formats = SND_SOC_STD_AC97_FMTS,
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},
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/* alsa ops */
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.ops = &stac9766_dai_ops_analog,
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},
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{
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.name = "stac9766-hifi-IEC958",
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/* stream cababilities */
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.playback = {
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.stream_name = "stac9766 IEC958",
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.channels_min = 1,
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.channels_max = 2,
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.rates = SNDRV_PCM_RATE_32000 | \
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SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
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.formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
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},
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/* alsa ops */
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.ops = &stac9766_dai_ops_digital,
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}
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};
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static int stac9766_codec_probe(struct snd_soc_codec *codec)
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{
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struct snd_ac97 *ac97;
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int ret = 0;
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ac97 = snd_soc_new_ac97_codec(codec, 0, 0);
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if (IS_ERR(ac97))
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return PTR_ERR(ac97);
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snd_soc_codec_set_drvdata(codec, ac97);
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/* do a cold reset for the controller and then try
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* a warm reset followed by an optional cold reset for codec */
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stac9766_reset(codec, 0);
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ret = stac9766_reset(codec, 1);
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if (ret < 0) {
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dev_err(codec->dev, "Failed to reset: AC97 link error\n");
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goto codec_err;
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}
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return 0;
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codec_err:
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snd_soc_free_ac97_codec(ac97);
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return ret;
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}
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static int stac9766_codec_remove(struct snd_soc_codec *codec)
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{
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struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
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snd_soc_free_ac97_codec(ac97);
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return 0;
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}
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static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
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.controls = stac9766_snd_ac97_controls,
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.num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls),
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.write = stac9766_ac97_write,
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.read = stac9766_ac97_read,
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.set_bias_level = stac9766_set_bias_level,
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.suspend_bias_off = true,
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.probe = stac9766_codec_probe,
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.remove = stac9766_codec_remove,
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.resume = stac9766_codec_resume,
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.reg_cache_size = ARRAY_SIZE(stac9766_reg),
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.reg_word_size = sizeof(u16),
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.reg_cache_step = 2,
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.reg_cache_default = stac9766_reg,
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};
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static int stac9766_probe(struct platform_device *pdev)
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{
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return snd_soc_register_codec(&pdev->dev,
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&soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
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}
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static int stac9766_remove(struct platform_device *pdev)
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{
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snd_soc_unregister_codec(&pdev->dev);
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return 0;
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}
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static struct platform_driver stac9766_codec_driver = {
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.driver = {
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.name = "stac9766-codec",
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},
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.probe = stac9766_probe,
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.remove = stac9766_remove,
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};
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module_platform_driver(stac9766_codec_driver);
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MODULE_DESCRIPTION("ASoC stac9766 driver");
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MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
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MODULE_LICENSE("GPL");
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