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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-06 01:02:30 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-06 01:02:30 +0000 |
commit | 76cb841cb886eef6b3bee341a2266c76578724ad (patch) | |
tree | f5892e5ba6cc11949952a6ce4ecbe6d516d6ce58 /sound/soc/codecs/stac9766.c | |
parent | Initial commit. (diff) | |
download | linux-76cb841cb886eef6b3bee341a2266c76578724ad.tar.xz linux-76cb841cb886eef6b3bee341a2266c76578724ad.zip |
Adding upstream version 4.19.249.upstream/4.19.249
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'sound/soc/codecs/stac9766.c')
-rw-r--r-- | sound/soc/codecs/stac9766.c | 342 |
1 files changed, 342 insertions, 0 deletions
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 000000000..f62101a62 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,342 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl@gmail.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define STAC9766_VENDOR_ID 0x83847666 +#define STAC9766_VENDOR_ID_MASK 0xffffffff + +#define AC97_STAC_DA_CONTROL 0x6A +#define AC97_STAC_ANALOG_SPECIAL 0x6E +#define AC97_STAC_STEREO_MIC 0x78 + +static const struct reg_default stac9766_reg_defaults[] = { + { 0x02, 0x8000 }, + { 0x04, 0x8000 }, + { 0x06, 0x8000 }, + { 0x0a, 0x0000 }, + { 0x0c, 0x8008 }, + { 0x0e, 0x8008 }, + { 0x10, 0x8808 }, + { 0x12, 0x8808 }, + { 0x14, 0x8808 }, + { 0x16, 0x8808 }, + { 0x18, 0x8808 }, + { 0x1a, 0x0000 }, + { 0x1c, 0x8000 }, + { 0x20, 0x0000 }, + { 0x22, 0x0000 }, + { 0x28, 0x0a05 }, + { 0x2c, 0xbb80 }, + { 0x32, 0xbb80 }, + { 0x3a, 0x2000 }, + { 0x3e, 0x0100 }, + { 0x4c, 0x0300 }, + { 0x4e, 0xffff }, + { 0x50, 0x0000 }, + { 0x52, 0x0000 }, + { 0x54, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6e, 0x1000 }, + { 0x72, 0x0000 }, + { 0x78, 0x0000 }, +}; + +static const struct regmap_config stac9766_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x78, + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = regmap_ac97_default_volatile, + + .reg_defaults = stac9766_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(stac9766_reg_defaults), +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", + "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum, + AC97_REC_SEL, 8, 0, stac9766_record_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum, + AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum, + AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum, + AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum, + AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum, + AC97_STAC_ANALOG_SPECIAL, 12, + stac9766_record_all_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum, + AC97_MIC, 6, stac9766_boost1); /* 0/10dB */ +static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum, + AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */ +static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum, + AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic); + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(master_tlv, -4650, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(record_tlv, 0, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(beep_tlv, -4500, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(mix_tlv, -3450, 150, 0); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, + master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, + master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg; + + /* enable variable rate audio, disable SPDIF output */ + snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x5, 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return snd_soc_component_write(component, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg; + + snd_soc_component_write(component, AC97_SPDIF, 0x2002); + + /* Enable VRA and SPDIF out */ + snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x5, 0x5); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return snd_soc_component_write(component, reg, runtime->rate); +} + +static int stac9766_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + snd_soc_component_write(component, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + snd_soc_component_write(component, AC97_POWERDOWN, 0xffff); + break; + } + return 0; +} + +static int stac9766_component_resume(struct snd_soc_component *component) +{ + struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); + + return snd_ac97_reset(ac97, true, STAC9766_VENDOR_ID, + STAC9766_VENDOR_ID_MASK); +} + +static const struct snd_soc_dai_ops stac9766_dai_ops_analog = { + .prepare = ac97_analog_prepare, +}; + +static const struct snd_soc_dai_ops stac9766_dai_ops_digital = { + .prepare = ac97_digital_prepare, +}; + +static struct snd_soc_dai_driver stac9766_dai[] = { +{ + .name = "stac9766-hifi-analog", + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766-hifi-IEC958", + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +} +}; + +static int stac9766_component_probe(struct snd_soc_component *component) +{ + struct snd_ac97 *ac97; + struct regmap *regmap; + int ret; + + ac97 = snd_soc_new_ac97_component(component, STAC9766_VENDOR_ID, + STAC9766_VENDOR_ID_MASK); + if (IS_ERR(ac97)) + return PTR_ERR(ac97); + + regmap = regmap_init_ac97(ac97, &stac9766_regmap_config); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + goto err_free_ac97; + } + + snd_soc_component_init_regmap(component, regmap); + snd_soc_component_set_drvdata(component, ac97); + + return 0; +err_free_ac97: + snd_soc_free_ac97_component(ac97); + return ret; +} + +static void stac9766_component_remove(struct snd_soc_component *component) +{ + struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); + + snd_soc_component_exit_regmap(component); + snd_soc_free_ac97_component(ac97); +} + +static const struct snd_soc_component_driver soc_component_dev_stac9766 = { + .controls = stac9766_snd_ac97_controls, + .num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls), + .set_bias_level = stac9766_set_bias_level, + .probe = stac9766_component_probe, + .remove = stac9766_component_remove, + .resume = stac9766_component_resume, + .suspend_bias_off = 1, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + +}; + +static int stac9766_probe(struct platform_device *pdev) +{ + return devm_snd_soc_register_component(&pdev->dev, + &soc_component_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} + +static struct platform_driver stac9766_codec_driver = { + .driver = { + .name = "stac9766-codec", + }, + + .probe = stac9766_probe, +}; + +module_platform_driver(stac9766_codec_driver); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); +MODULE_LICENSE("GPL"); |