diff options
Diffstat (limited to 'sound/soc/codecs/stac9766.c')
| -rw-r--r-- | sound/soc/codecs/stac9766.c | 407 |
1 files changed, 407 insertions, 0 deletions
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 000000000..6464caf72 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,407 @@ +/* + * 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 <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "stac9766.h" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", + "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static 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 DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, + master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, + master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops->write(ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops->write(ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops->read(ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + vra &= ~0x4; /* disable SPDIF output */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops->reset(ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_resume(struct snd_soc_codec *codec) +{ + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + dev_err(codec->dev, "Failed to resume\n"); + return -EIO; + } + ac97->bus->ops->warm_reset(ac97); + id = soc_ac97_ops->read(ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + + return 0; +} + +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_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +} +}; + +static int stac9766_codec_probe(struct snd_soc_codec *codec) +{ + struct snd_ac97 *ac97; + int ret = 0; + + ac97 = snd_soc_new_ac97_codec(codec); + if (IS_ERR(ac97)) + return PTR_ERR(ac97); + + snd_soc_codec_set_drvdata(codec, ac97); + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + dev_err(codec->dev, "Failed to reset: AC97 link error\n"); + goto codec_err; + } + + return 0; + +codec_err: + snd_soc_free_ac97_codec(ac97); + return ret; +} + +static int stac9766_codec_remove(struct snd_soc_codec *codec) +{ + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + snd_soc_free_ac97_codec(ac97); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { + .controls = stac9766_snd_ac97_controls, + .num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls), + .write = stac9766_ac97_write, + .read = stac9766_ac97_read, + .set_bias_level = stac9766_set_bias_level, + .suspend_bias_off = true, + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .resume = stac9766_codec_resume, + .reg_cache_size = ARRAY_SIZE(stac9766_reg), + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = stac9766_reg, +}; + +static int stac9766_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} + +static int stac9766_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver stac9766_codec_driver = { + .driver = { + .name = "stac9766-codec", + }, + + .probe = stac9766_probe, + .remove = stac9766_remove, +}; + +module_platform_driver(stac9766_codec_driver); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); +MODULE_LICENSE("GPL"); |