/* * ALSA SoC TLV320AIC23 codec driver * * Author: Arun KS, * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., * * Based on sound/soc/codecs/wm8731.c by Richard Purdie * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * * Notes: * The AIC23 is a driver for a low power stereo audio * codec tlv320aic23 * * The machine layer should disable unsupported inputs/outputs by * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include "tlv320aic23.h" /* * AIC23 register cache */ static const struct reg_default tlv320aic23_reg[] = { { 0, 0x0097 }, { 1, 0x0097 }, { 2, 0x00F9 }, { 3, 0x00F9 }, { 4, 0x001A }, { 5, 0x0004 }, { 6, 0x0007 }, { 7, 0x0001 }, { 8, 0x0020 }, { 9, 0x0000 }, }; const struct regmap_config tlv320aic23_regmap = { .reg_bits = 7, .val_bits = 9, .max_register = TLV320AIC23_RESET, .reg_defaults = tlv320aic23_reg, .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg), .cache_type = REGCACHE_RBTREE, }; static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; static SOC_ENUM_SINGLE_DECL(rec_src_enum, TLV320AIC23_ANLG, 2, rec_src_text); static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = SOC_DAPM_ENUM("Input Select", rec_src_enum); static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src, TLV320AIC23_ANLG, 2, rec_src_text); static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph, TLV320AIC23_DIGT, 1, deemph_text); static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0); static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u16 val, reg; val = (ucontrol->value.integer.value[0] & 0x07); /* linear conversion to userspace * 000 = -6db * 001 = -9db * 010 = -12db * 011 = -18db (Min) * 100 = 0db (Max) */ val = (val >= 4) ? 4 : (3 - val); reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0); snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); return 0; } static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u16 val; val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0); val = val >> 6; val = (val >= 4) ? 4 : (3 - val); ucontrol->value.integer.value[0] = val; return 0; } static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1), SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL, TLV320AIC23_RINVOL, 7, 1, 0), SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL, TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), SOC_SINGLE_EXT_TLV("Sidetone Volume", TLV320AIC23_ANLG, 6, 4, 0, snd_soc_tlv320aic23_get_volsw, snd_soc_tlv320aic23_put_volsw, sidetone_vol_tlv), SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; /* PGA Mixer controls for Line and Mic switch */ static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0), SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0), }; static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1), SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, &tlv320aic23_rec_src_mux_controls), SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1, &tlv320aic23_output_mixer_controls[0], ARRAY_SIZE(tlv320aic23_output_mixer_controls)), SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0), SND_SOC_DAPM_OUTPUT("LHPOUT"), SND_SOC_DAPM_OUTPUT("RHPOUT"), SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), SND_SOC_DAPM_INPUT("LLINEIN"), SND_SOC_DAPM_INPUT("RLINEIN"), SND_SOC_DAPM_INPUT("MICIN"), }; static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* Output Mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "Playback Switch", "DAC"}, {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, /* Outputs */ {"RHPOUT", NULL, "Output Mixer"}, {"LHPOUT", NULL, "Output Mixer"}, {"LOUT", NULL, "Output Mixer"}, {"ROUT", NULL, "Output Mixer"}, /* Inputs */ {"Line Input", "NULL", "LLINEIN"}, {"Line Input", "NULL", "RLINEIN"}, {"Mic Input", "NULL", "MICIN"}, /* input mux */ {"Capture Source", "Line", "Line Input"}, {"Capture Source", "Mic", "Mic Input"}, {"ADC", NULL, "Capture Source"}, }; /* AIC23 driver data */ struct aic23 { struct regmap *regmap; int mclk; int requested_adc; int requested_dac; }; /* * Common Crystals used * 11.2896 Mhz /128 = *88.2k /192 = 58.8k * 12.0000 Mhz /125 = *96k /136 = 88.235K * 12.2880 Mhz /128 = *96k /192 = 64k * 16.9344 Mhz /128 = 132.3k /192 = *88.2k * 18.4320 Mhz /128 = 144k /192 = *96k */ /* * Normal BOSR 0-256/2 = 128, 1-384/2 = 192 * USB BOSR 0-250/2 = 125, 1-272/2 = 136 */ static const int bosr_usb_divisor_table[] = { 128, 125, 192, 136 }; #define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7)) #define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) static const unsigned short sr_valid_mask[] = { LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ LOWER_GROUP, /* Usb, bosr - 0*/ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ UPPER_GROUP, /* Usb, bosr - 1*/ }; /* * Every divisor is a factor of 11*12 */ #define SR_MULT (11*12) #define A(x) (SR_MULT/x) static const unsigned char sr_adc_mult_table[] = { A(2), A(2), A(12), A(12), 0, 0, A(3), A(1), A(2), A(2), A(11), A(11), 0, 0, 0, A(1) }; static const unsigned char sr_dac_mult_table[] = { A(2), A(12), A(2), A(12), 0, 0, A(3), A(1), A(2), A(11), A(2), A(11), 0, 0, 0, A(1) }; static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, int dac, int dac_l, int dac_h, int need_dac) { if ((adc >= adc_l) && (adc <= adc_h) && (dac >= dac_l) && (dac <= dac_h)) { int diff_adc = need_adc - adc; int diff_dac = need_dac - dac; return abs(diff_adc) + abs(diff_dac); } return UINT_MAX; } static int find_rate(int mclk, u32 need_adc, u32 need_dac) { int i, j; int best_i = -1; int best_j = -1; int best_div = 0; unsigned best_score = UINT_MAX; int adc_l, adc_h, dac_l, dac_h; need_adc *= SR_MULT; need_dac *= SR_MULT; /* * rates given are +/- 1/32 */ adc_l = need_adc - (need_adc >> 5); adc_h = need_adc + (need_adc >> 5); dac_l = need_dac - (need_dac >> 5); dac_h = need_dac + (need_dac >> 5); for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) { int base = mclk / bosr_usb_divisor_table[i]; int mask = sr_valid_mask[i]; for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table); j++, mask >>= 1) { int adc; int dac; int score; if ((mask & 1) == 0) continue; adc = base * sr_adc_mult_table[j]; dac = base * sr_dac_mult_table[j]; score = get_score(adc, adc_l, adc_h, need_adc, dac, dac_l, dac_h, need_dac); if (best_score > score) { best_score = score; best_i = i; best_j = j; best_div = 0; } score = get_score((adc >> 1), adc_l, adc_h, need_adc, (dac >> 1), dac_l, dac_h, need_dac); /* prefer to have a /2 */ if ((score != UINT_MAX) && (best_score >= score)) { best_score = score; best_i = i; best_j = j; best_div = 1; } } } return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); } #ifdef DEBUG static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { int src = snd_soc_read(codec, TLV320AIC23_SRATE); int sr = (src >> 2) & 0x0f; int val = (mclk / bosr_usb_divisor_table[src & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; int dac = (val * sr_dac_mult_table[sr]) / SR_MULT; if (src & TLV320AIC23_CLKIN_HALF) { adc >>= 1; dac >>= 1; } *sample_rate_adc = adc; *sample_rate_dac = dac; } #endif static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, u32 sample_rate_adc, u32 sample_rate_dac) { /* Search for the right sample rate */ int data = find_rate(mclk, sample_rate_adc, sample_rate_dac); if (data < 0) { printk(KERN_ERR "%s:Invalid rate %u,%u requested\n", __func__, sample_rate_adc, sample_rate_dac); return -EINVAL; } snd_soc_write(codec, TLV320AIC23_SRATE, data); #ifdef DEBUG { u32 adc, dac; get_current_sample_rates(codec, mclk, &adc, &dac); printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", adc, dac, data); } #endif return 0; } static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; u16 iface_reg; int ret; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); u32 sample_rate_adc = aic23->requested_adc; u32 sample_rate_dac = aic23->requested_dac; u32 sample_rate = params_rate(params); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { aic23->requested_dac = sample_rate_dac = sample_rate; if (!sample_rate_adc) sample_rate_adc = sample_rate; } else { aic23->requested_adc = sample_rate_adc = sample_rate; if (!sample_rate_dac) sample_rate_dac = sample_rate; } ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc, sample_rate_dac); if (ret < 0) return ret; iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: iface_reg |= (0x01 << 2); break; case SNDRV_PCM_FORMAT_S24_LE: iface_reg |= (0x02 << 2); break; case SNDRV_PCM_FORMAT_S32_LE: iface_reg |= (0x03 << 2); break; } snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; /* set active */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); return 0; } static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ if (!codec->active) { udelay(50); snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aic23->requested_dac = 0; else aic23->requested_adc = 0; } static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 reg; reg = snd_soc_read(codec, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; else reg &= ~TLV320AIC23_DACM_MUTE; snd_soc_write(codec, TLV320AIC23_DIGT, reg); return 0; } static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; u16 iface_reg; iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: iface_reg |= TLV320AIC23_MS_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: iface_reg &= ~TLV320AIC23_MS_MASTER; break; default: return -EINVAL; } /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: iface_reg |= TLV320AIC23_FOR_I2S; break; case SND_SOC_DAIFMT_DSP_A: iface_reg |= TLV320AIC23_LRP_ON; case SND_SOC_DAIFMT_DSP_B: iface_reg |= TLV320AIC23_FOR_DSP; break; case SND_SOC_DAIFMT_RIGHT_J: break; case SND_SOC_DAIFMT_LEFT_J: iface_reg |= TLV320AIC23_FOR_LJUST; break; default: return -EINVAL; } snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct aic23 *aic23 = snd_soc_dai_get_drvdata(codec_dai); aic23->mclk = freq; return 0; } static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f; switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ TLV320AIC23_DAC_OFF); snd_soc_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ snd_soc_write(codec, TLV320AIC23_PWR, reg | TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff); break; } codec->dapm.bias_level = level; return 0; } #define AIC23_RATES SNDRV_PCM_RATE_8000_96000 #define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops tlv320aic23_dai_ops = { .prepare = tlv320aic23_pcm_prepare, .hw_params = tlv320aic23_hw_params, .shutdown = tlv320aic23_shutdown, .digital_mute = tlv320aic23_mute, .set_fmt = tlv320aic23_set_dai_fmt, .set_sysclk = tlv320aic23_set_dai_sysclk, }; static struct snd_soc_dai_driver tlv320aic23_dai = { .name = "tlv320aic23-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = AIC23_RATES, .formats = AIC23_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = AIC23_RATES, .formats = AIC23_FORMATS,}, .ops = &tlv320aic23_dai_ops, }; static int tlv320aic23_suspend(struct snd_soc_codec *codec) { tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int tlv320aic23_resume(struct snd_soc_codec *codec) { struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); regcache_mark_dirty(aic23->regmap); regcache_sync(aic23->regmap); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) { int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } /* Reset codec */ snd_soc_write(codec, TLV320AIC23_RESET, 0); /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ snd_soc_update_bits(codec, TLV320AIC23_LINVOL, TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); snd_soc_update_bits(codec, TLV320AIC23_RINVOL, TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); snd_soc_update_bits(codec, TLV320AIC23_ANLG, TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED, 0); /* Default output volume */ snd_soc_write(codec, TLV320AIC23_LCHNVOL, TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); snd_soc_write(codec, TLV320AIC23_RCHNVOL, TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); return 0; } static int tlv320aic23_remove(struct snd_soc_codec *codec) { tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .probe = tlv320aic23_codec_probe, .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, .set_bias_level = tlv320aic23_set_bias_level, .controls = tlv320aic23_snd_controls, .num_controls = ARRAY_SIZE(tlv320aic23_snd_controls), .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), .dapm_routes = tlv320aic23_intercon, .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; int tlv320aic23_probe(struct device *dev, struct regmap *regmap) { struct aic23 *aic23; if (IS_ERR(regmap)) return PTR_ERR(regmap); aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL); if (aic23 == NULL) return -ENOMEM; aic23->regmap = regmap; dev_set_drvdata(dev, aic23); return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); } MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS "); MODULE_LICENSE("GPL");