dbad34eac2
Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap)
, it is not necessary to provide codec->control_data anymore.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
415 lines
12 KiB
C
415 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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#define STAC9766_VERSION "0.10"
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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 const struct soc_enum stac9766_record_enum =
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SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
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static const struct soc_enum stac9766_mono_enum =
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SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
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static const struct soc_enum stac9766_mic_enum =
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SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
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static const struct soc_enum stac9766_SPDIF_enum =
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SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
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static const struct soc_enum stac9766_popbypass_enum =
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SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
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static const struct soc_enum stac9766_record_all_enum =
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SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
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stac9766_record_all_mux);
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static const struct soc_enum stac9766_boost1_enum =
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SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
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static const struct soc_enum stac9766_boost2_enum =
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SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
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static const struct soc_enum stac9766_stereo_mic_enum =
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SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, 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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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(codec->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(codec->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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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(codec->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(codec->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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codec->dapm.bias_level = level;
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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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if (try_warm && soc_ac97_ops.warm_reset) {
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soc_ac97_ops.warm_reset(codec->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(codec->ac97);
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if (soc_ac97_ops.warm_reset)
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soc_ac97_ops.warm_reset(codec->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_suspend(struct snd_soc_codec *codec)
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{
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stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
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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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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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printk(KERN_ERR "stac9766 failed to resume");
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return -EIO;
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}
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codec->ac97->bus->ops->warm_reset(codec->ac97);
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id = soc_ac97_ops.read(codec->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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stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
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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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.ac97_control = 1,
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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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.ac97_control = 1,
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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_FORMAT_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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int ret = 0;
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printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
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ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
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if (ret < 0)
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goto codec_err;
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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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printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
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goto codec_err;
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}
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stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
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snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls,
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ARRAY_SIZE(stac9766_snd_ac97_controls));
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return 0;
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codec_err:
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snd_soc_free_ac97_codec(codec);
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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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snd_soc_free_ac97_codec(codec);
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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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.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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.probe = stac9766_codec_probe,
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.remove = stac9766_codec_remove,
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.suspend = stac9766_codec_suspend,
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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 __devinit 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 __devexit 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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.owner = THIS_MODULE,
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},
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.probe = stac9766_probe,
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.remove = __devexit_p(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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