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29e189c29d
The core will ensure that the device is in either STANDBY or OFF bias before suspending, restoring the bias in the driver is unneeded. Some drivers doing slightly more roundabout things have been left alone for now. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
442 lines
13 KiB
C
442 lines
13 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/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 <sound/soc-of-simple.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->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 platform_device *pdev,
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pm_message_t state)
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{
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
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struct snd_soc_codec *codec = socdev->card->codec;
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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 platform_device *pdev)
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{
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
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struct snd_soc_codec *codec = socdev->card->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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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 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 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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struct snd_soc_dai stac9766_dai[] = {
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{
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.name = "stac9766 analog",
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.id = 0,
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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 IEC958",
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.id = 1,
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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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EXPORT_SYMBOL_GPL(stac9766_dai);
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static int stac9766_codec_probe(struct platform_device *pdev)
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{
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
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struct snd_soc_codec *codec;
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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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socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
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if (socdev->card->codec == NULL)
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return -ENOMEM;
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codec = socdev->card->codec;
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mutex_init(&codec->mutex);
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codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg),
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GFP_KERNEL);
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if (codec->reg_cache == NULL) {
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ret = -ENOMEM;
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goto cache_err;
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}
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codec->reg_cache_size = sizeof(stac9766_reg);
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codec->reg_cache_step = 2;
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codec->name = "STAC9766";
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codec->owner = THIS_MODULE;
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codec->dai = stac9766_dai;
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codec->num_dai = ARRAY_SIZE(stac9766_dai);
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codec->write = stac9766_ac97_write;
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codec->read = stac9766_ac97_read;
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codec->set_bias_level = stac9766_set_bias_level;
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INIT_LIST_HEAD(&codec->dapm_widgets);
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INIT_LIST_HEAD(&codec->dapm_paths);
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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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/* register pcms */
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ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
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if (ret < 0)
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goto pcm_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 reset_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_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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reset_err:
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snd_soc_free_pcms(socdev);
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pcm_err:
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snd_soc_free_ac97_codec(codec);
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codec_err:
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kfree(snd_soc_codec_get_drvdata(codec));
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cache_err:
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kfree(socdev->card->codec);
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socdev->card->codec = NULL;
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return ret;
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}
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static int stac9766_codec_remove(struct platform_device *pdev)
|
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{
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|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
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struct snd_soc_codec *codec = socdev->card->codec;
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|
|
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if (codec == NULL)
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return 0;
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|
|
|
snd_soc_free_pcms(socdev);
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|
snd_soc_free_ac97_codec(codec);
|
|
kfree(codec->reg_cache);
|
|
kfree(codec);
|
|
return 0;
|
|
}
|
|
|
|
struct snd_soc_codec_device soc_codec_dev_stac9766 = {
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|
.probe = stac9766_codec_probe,
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|
.remove = stac9766_codec_remove,
|
|
.suspend = stac9766_codec_suspend,
|
|
.resume = stac9766_codec_resume,
|
|
};
|
|
EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);
|
|
|
|
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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