From ff06ac2a0489cfe913215d424667b52ad6c0fba1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 29 Jan 2018 04:12:48 +0000 Subject: ASoC: tlv320aic23: replace codec to component Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23-i2c.c | 7 --- sound/soc/codecs/tlv320aic23-spi.c | 7 --- sound/soc/codecs/tlv320aic23.c | 120 +++++++++++++++++++------------------ 3 files changed, 61 insertions(+), 73 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c index 78a94af65518..1d7c117316fb 100644 --- a/sound/soc/codecs/tlv320aic23-i2c.c +++ b/sound/soc/codecs/tlv320aic23-i2c.c @@ -31,12 +31,6 @@ static int tlv320aic23_i2c_probe(struct i2c_client *i2c, return tlv320aic23_probe(&i2c->dev, regmap); } -static int tlv320aic23_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; -} - static const struct i2c_device_id tlv320aic23_id[] = { {"tlv320aic23", 0}, {} @@ -56,7 +50,6 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic23_of_match), }, .probe = tlv320aic23_i2c_probe, - .remove = tlv320aic23_i2c_remove, .id_table = tlv320aic23_id, }; diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c index f801ae051658..d8c9ec1e9201 100644 --- a/sound/soc/codecs/tlv320aic23-spi.c +++ b/sound/soc/codecs/tlv320aic23-spi.c @@ -34,18 +34,11 @@ static int aic23_spi_probe(struct spi_device *spi) return tlv320aic23_probe(&spi->dev, regmap); } -static int aic23_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - static struct spi_driver aic23_spi = { .driver = { .name = "tlv320aic23", }, .probe = aic23_spi_probe, - .remove = aic23_spi_remove, }; module_spi_driver(aic23_spi); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 74909211c608..47480cb4d078 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -82,7 +82,7 @@ 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_soc_kcontrol_codec(kcontrol); + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); u16 val, reg; val = (ucontrol->value.integer.value[0] & 0x07); @@ -96,8 +96,8 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, */ val = (val >= 4) ? 4 : (3 - val); - reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0); - snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + reg = snd_soc_component_read32(component, TLV320AIC23_ANLG) & (~0x1C0); + snd_soc_component_write(component, TLV320AIC23_ANLG, reg | (val << 6)); return 0; } @@ -105,10 +105,10 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); u16 val; - val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0); + val = snd_soc_component_read32(component, TLV320AIC23_ANLG) & (0x1C0); val = val >> 6; val = (val >= 4) ? 4 : (3 - val); ucontrol->value.integer.value[0] = val; @@ -296,10 +296,10 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) } #ifdef DEBUG -static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, +static void get_current_sample_rates(struct snd_soc_component *component, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = snd_soc_read(codec, TLV320AIC23_SRATE); + int src = snd_soc_component_read32(component, 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; @@ -313,7 +313,7 @@ static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, } #endif -static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, +static int set_sample_rate_control(struct snd_soc_component *component, int mclk, u32 sample_rate_adc, u32 sample_rate_dac) { /* Search for the right sample rate */ @@ -323,11 +323,11 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, __func__, sample_rate_adc, sample_rate_dac); return -EINVAL; } - snd_soc_write(codec, TLV320AIC23_SRATE, data); + snd_soc_component_write(component, TLV320AIC23_SRATE, data); #ifdef DEBUG { u32 adc, dac; - get_current_sample_rates(codec, mclk, &adc, &dac); + get_current_sample_rates(component, mclk, &adc, &dac); printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", adc, dac, data); } @@ -339,10 +339,10 @@ 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; + struct snd_soc_component *component = dai->component; u16 iface_reg; int ret; - struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); + struct aic23 *aic23 = snd_soc_component_get_drvdata(component); u32 sample_rate_adc = aic23->requested_adc; u32 sample_rate_dac = aic23->requested_dac; u32 sample_rate = params_rate(params); @@ -356,12 +356,12 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, if (!sample_rate_dac) sample_rate_dac = sample_rate; } - ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc, + ret = set_sample_rate_control(component, aic23->mclk, sample_rate_adc, sample_rate_dac); if (ret < 0) return ret; - iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + iface_reg = snd_soc_component_read32(component, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); switch (params_width(params)) { case 16: @@ -376,7 +376,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg |= (0x03 << 2); break; } - snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_component_write(component, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -384,10 +384,10 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_codec *codec = dai->codec; + struct snd_soc_component *component = dai->component; /* set active */ - snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); + snd_soc_component_write(component, TLV320AIC23_ACTIVE, 0x0001); return 0; } @@ -395,13 +395,13 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, 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); + struct snd_soc_component *component = dai->component; + struct aic23 *aic23 = snd_soc_component_get_drvdata(component); /* deactivate */ - if (!snd_soc_codec_is_active(codec)) { + if (!snd_soc_component_is_active(component)) { udelay(50); - snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_component_write(component, TLV320AIC23_ACTIVE, 0x0); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aic23->requested_dac = 0; @@ -411,17 +411,17 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; + struct snd_soc_component *component = dai->component; u16 reg; - reg = snd_soc_read(codec, TLV320AIC23_DIGT); + reg = snd_soc_component_read32(component, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; else reg &= ~TLV320AIC23_DACM_MUTE; - snd_soc_write(codec, TLV320AIC23_DIGT, reg); + snd_soc_component_write(component, TLV320AIC23_DIGT, reg); return 0; } @@ -429,10 +429,10 @@ static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_component *component = codec_dai->component; u16 iface_reg; - iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + iface_reg = snd_soc_component_read32(component, TLV320AIC23_DIGT_FMT) & (~0x03); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -468,7 +468,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, } - snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_component_write(component, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -481,29 +481,29 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, +static int tlv320aic23_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f; + u16 reg = snd_soc_component_read32(component, 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); + snd_soc_component_write(component, 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, + snd_soc_component_write(component, 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); + snd_soc_component_write(component, TLV320AIC23_ACTIVE, 0x0); + snd_soc_component_write(component, TLV320AIC23_PWR, 0x1ff); break; } return 0; @@ -539,58 +539,59 @@ static struct snd_soc_dai_driver tlv320aic23_dai = { .ops = &tlv320aic23_dai_ops, }; -static int tlv320aic23_resume(struct snd_soc_codec *codec) +static int tlv320aic23_resume(struct snd_soc_component *component) { - struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); + struct aic23 *aic23 = snd_soc_component_get_drvdata(component); regcache_mark_dirty(aic23->regmap); regcache_sync(aic23->regmap); return 0; } -static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) +static int tlv320aic23_component_probe(struct snd_soc_component *component) { /* Reset codec */ - snd_soc_write(codec, TLV320AIC23_RESET, 0); + snd_soc_component_write(component, TLV320AIC23_RESET, 0); - snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + snd_soc_component_write(component, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ - snd_soc_update_bits(codec, TLV320AIC23_LINVOL, + snd_soc_component_update_bits(component, TLV320AIC23_LINVOL, TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - snd_soc_update_bits(codec, TLV320AIC23_RINVOL, + snd_soc_component_update_bits(component, TLV320AIC23_RINVOL, TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - snd_soc_update_bits(codec, TLV320AIC23_ANLG, + snd_soc_component_update_bits(component, TLV320AIC23_ANLG, TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED, 0); /* Default output volume */ - snd_soc_write(codec, TLV320AIC23_LCHNVOL, + snd_soc_component_write(component, TLV320AIC23_LCHNVOL, TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); - snd_soc_write(codec, TLV320AIC23_RCHNVOL, + snd_soc_component_write(component, TLV320AIC23_RCHNVOL, TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); - snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); + snd_soc_component_write(component, TLV320AIC23_ACTIVE, 0x1); return 0; } -static const struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { - .probe = tlv320aic23_codec_probe, - .resume = tlv320aic23_resume, - .set_bias_level = tlv320aic23_set_bias_level, - .suspend_bias_off = true, - - .component_driver = { - .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), - }, +static const struct snd_soc_component_driver soc_component_dev_tlv320aic23 = { + .probe = tlv320aic23_component_probe, + .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), + .suspend_bias_off = 1, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; int tlv320aic23_probe(struct device *dev, struct regmap *regmap) @@ -608,7 +609,8 @@ int tlv320aic23_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, aic23); - return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23, + return devm_snd_soc_register_component(dev, + &soc_component_dev_tlv320aic23, &tlv320aic23_dai, 1); } EXPORT_SYMBOL(tlv320aic23_probe); -- cgit v1.2.3 From 1514613a7a4480e40f40a2d41527292fece8b362 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 29 Jan 2018 04:13:13 +0000 Subject: ASoC: tlv320aic26: replace codec to component Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 83 ++++++++++++++++++++---------------------- 1 file changed, 39 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 89421caaeb70..b91b8d5f1ba3 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -30,7 +30,7 @@ MODULE_LICENSE("GPL"); struct aic26 { struct spi_device *spi; struct regmap *regmap; - struct snd_soc_codec *codec; + struct snd_soc_component *component; int master; int datfm; int mclk; @@ -64,8 +64,8 @@ static int aic26_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; - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct aic26 *aic26 = snd_soc_component_get_drvdata(component); int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; @@ -112,20 +112,20 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; - snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg); + snd_soc_component_write(component, AIC26_REG_PLL_PROG1, reg); reg = dval << 2; - snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); + snd_soc_component_write(component, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ if (aic26->master) reg = 0x0800; if (fsref == 48000) reg = 0x2000; - snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL3, 0xf800, reg); + snd_soc_component_update_bits(component, AIC26_REG_AUDIO_CTRL3, 0xf800, reg); /* Audio Control 1 (FSref divisor) */ reg = wlen | aic26->datfm | (divisor << 3) | divisor; - snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL1, 0xfff, reg); + snd_soc_component_update_bits(component, AIC26_REG_AUDIO_CTRL1, 0xfff, reg); return 0; } @@ -135,8 +135,8 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, */ static int aic26_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct aic26 *aic26 = snd_soc_component_get_drvdata(component); u16 reg; dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n", @@ -146,7 +146,7 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) reg = 0x8080; else reg = 0; - snd_soc_update_bits(codec, AIC26_REG_DAC_GAIN, 0x8000, reg); + snd_soc_component_update_bits(component, AIC26_REG_DAC_GAIN, 0x8000, reg); return 0; } @@ -154,8 +154,8 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) static int aic26_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct aic26 *aic26 = snd_soc_component_get_drvdata(component); dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i," " freq=%i, dir=%i)\n", @@ -171,8 +171,8 @@ static int aic26_set_sysclk(struct snd_soc_dai *codec_dai, static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct aic26 *aic26 = snd_soc_component_get_drvdata(component); dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n", codec_dai, fmt); @@ -265,7 +265,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = snd_soc_read(aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = snd_soc_component_read32(aic26->component, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -280,7 +280,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, { struct aic26 *aic26 = dev_get_drvdata(dev); - snd_soc_update_bits(aic26->codec, AIC26_REG_AUDIO_CTRL2, + snd_soc_component_update_bits(aic26->component, AIC26_REG_AUDIO_CTRL2, 0x8000, 0x800); return count; @@ -291,44 +291,46 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); /* --------------------------------------------------------------------- * SoC CODEC portion of driver: probe and release routines */ -static int aic26_probe(struct snd_soc_codec *codec) +static int aic26_probe(struct snd_soc_component *component) { - struct aic26 *aic26 = dev_get_drvdata(codec->dev); + struct aic26 *aic26 = dev_get_drvdata(component->dev); int ret, reg; - aic26->codec = codec; + aic26->component = component; /* Reset the codec to power on defaults */ - snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); + snd_soc_component_write(component, AIC26_REG_RESET, 0xBB00); /* Power up CODEC */ - snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0); + snd_soc_component_write(component, AIC26_REG_POWER_CTRL, 0); /* Audio Control 3 (master mode, fsref rate) */ - reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3); + reg = snd_soc_component_read32(component, AIC26_REG_AUDIO_CTRL3); reg &= ~0xf800; reg |= 0x0800; /* set master mode */ - snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_component_write(component, AIC26_REG_AUDIO_CTRL3, reg); /* Register the sysfs files for debugging */ /* Create SysFS files */ - ret = device_create_file(codec->dev, &dev_attr_keyclick); + ret = device_create_file(component->dev, &dev_attr_keyclick); if (ret) - dev_info(codec->dev, "error creating sysfs files\n"); + dev_info(component->dev, "error creating sysfs files\n"); return 0; } -static const struct snd_soc_codec_driver aic26_soc_codec_dev = { - .probe = aic26_probe, - .component_driver = { - .controls = aic26_snd_controls, - .num_controls = ARRAY_SIZE(aic26_snd_controls), - .dapm_widgets = tlv320aic26_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), - .dapm_routes = tlv320aic26_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes), - }, +static const struct snd_soc_component_driver aic26_soc_component_dev = { + .probe = aic26_probe, + .controls = aic26_snd_controls, + .num_controls = ARRAY_SIZE(aic26_snd_controls), + .dapm_widgets = tlv320aic26_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), + .dapm_routes = tlv320aic26_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static const struct regmap_config aic26_regmap = { @@ -361,23 +363,16 @@ static int aic26_spi_probe(struct spi_device *spi) dev_set_drvdata(&spi->dev, aic26); aic26->master = 1; - ret = snd_soc_register_codec(&spi->dev, - &aic26_soc_codec_dev, &aic26_dai, 1); + ret = devm_snd_soc_register_component(&spi->dev, + &aic26_soc_component_dev, &aic26_dai, 1); return ret; } -static int aic26_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - static struct spi_driver aic26_spi = { .driver = { .name = "tlv320aic26-codec", }, .probe = aic26_spi_probe, - .remove = aic26_spi_remove, }; module_spi_driver(aic26_spi); -- cgit v1.2.3 From f2e6f95b4b1381fe81cdbdb70d2c50a85b770d8b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 29 Jan 2018 04:15:27 +0000 Subject: ASoC: tlv320aic31xx: replace codec to component Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 240 +++++++++++++++++++-------------------- 1 file changed, 116 insertions(+), 124 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 858cb8be445f..d3cd924dc300 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -153,7 +153,7 @@ struct aic31xx_disable_nb { }; struct aic31xx_priv { - struct snd_soc_codec *codec; + struct snd_soc_component *component; u8 i2c_regs_status; struct device *dev; struct regmap *regmap; @@ -348,8 +348,8 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); unsigned int reg = AIC31XX_DACFLAG1; unsigned int mask; @@ -377,7 +377,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, reg = AIC31XX_ADCFLAG; break; default: - dev_err(codec->dev, "Unknown widget '%s' calling %s\n", + dev_err(component->dev, "Unknown widget '%s' calling %s\n", w->name, __func__); return -EINVAL; } @@ -388,7 +388,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100); default: - dev_dbg(codec->dev, + dev_dbg(component->dev, "Unhandled dapm widget event %d from %s\n", event, w->name); } @@ -444,23 +444,23 @@ static const struct snd_kcontrol_new aic31xx_dapm_spr_switch = static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); switch (event) { case SND_SOC_DAPM_POST_PMU: /* change mic bias voltage to user defined */ - snd_soc_update_bits(codec, AIC31XX_MICBIAS, + snd_soc_component_update_bits(component, AIC31XX_MICBIAS, AIC31XX_MICBIAS_MASK, aic31xx->micbias_vg << AIC31XX_MICBIAS_SHIFT); - dev_dbg(codec->dev, "%s: turned on\n", __func__); + dev_dbg(component->dev, "%s: turned on\n", __func__); break; case SND_SOC_DAPM_PRE_PMD: /* turn mic bias off */ - snd_soc_update_bits(codec, AIC31XX_MICBIAS, + snd_soc_component_update_bits(component, AIC31XX_MICBIAS, AIC31XX_MICBIAS_MASK, 0); - dev_dbg(codec->dev, "%s: turned off\n", __func__); + dev_dbg(component->dev, "%s: turned off\n", __func__); break; } return 0; @@ -670,34 +670,34 @@ aic310x_audio_map[] = { {"SPK", NULL, "SPK ClassD"}, }; -static int aic31xx_add_controls(struct snd_soc_codec *codec) +static int aic31xx_add_controls(struct snd_soc_component *component) { int ret = 0; - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); if (!(aic31xx->codec_type & DAC31XX_BIT)) - ret = snd_soc_add_codec_controls( - codec, aic31xx_snd_controls, + ret = snd_soc_add_component_controls( + component, aic31xx_snd_controls, ARRAY_SIZE(aic31xx_snd_controls)); if (ret) return ret; if (aic31xx->codec_type & AIC31XX_STEREO_CLASS_D_BIT) - ret = snd_soc_add_codec_controls( - codec, aic311x_snd_controls, + ret = snd_soc_add_component_controls( + component, aic311x_snd_controls, ARRAY_SIZE(aic311x_snd_controls)); else - ret = snd_soc_add_codec_controls( - codec, aic310x_snd_controls, + ret = snd_soc_add_component_controls( + component, aic310x_snd_controls, ARRAY_SIZE(aic310x_snd_controls)); return ret; } -static int aic31xx_add_widgets(struct snd_soc_codec *codec) +static int aic31xx_add_widgets(struct snd_soc_component *component) { - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); int ret = 0; if (aic31xx->codec_type & DAC31XX_BIT) { @@ -751,10 +751,10 @@ static int aic31xx_add_widgets(struct snd_soc_codec *codec) return 0; } -static int aic31xx_setup_pll(struct snd_soc_codec *codec, +static int aic31xx_setup_pll(struct snd_soc_component *component, struct snd_pcm_hw_params *params) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); int bclk_score = snd_soc_params_to_frame_size(params); int mclk_p; int bclk_n = 0; @@ -762,15 +762,15 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, int i; if (!aic31xx->sysclk || !aic31xx->p_div) { - dev_err(codec->dev, "Master clock not supplied\n"); + dev_err(component->dev, "Master clock not supplied\n"); return -EINVAL; } mclk_p = aic31xx->sysclk / aic31xx->p_div; /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ - snd_soc_update_bits(codec, AIC31XX_CLKMUX, + snd_soc_component_update_bits(component, AIC31XX_CLKMUX, AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL); - snd_soc_update_bits(codec, AIC31XX_IFACE2, + snd_soc_component_update_bits(component, AIC31XX_IFACE2, AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK); for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { @@ -789,14 +789,14 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, } if (match == -1) { - dev_err(codec->dev, + dev_err(component->dev, "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); /* See bellow for details how fix this. */ return -EINVAL; } if (bclk_score != 0) { - dev_warn(codec->dev, "Can not produce exact bitclock"); + dev_warn(component->dev, "Can not produce exact bitclock"); /* This is fine if using dsp format, but if using i2s there may be trouble. To fix the issue edit the aic31xx_divs table for your mclk and sample @@ -808,39 +808,39 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, i = match; /* PLL configuration */ - snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, + snd_soc_component_update_bits(component, AIC31XX_PLLPR, AIC31XX_PLL_MASK, (aic31xx->p_div << 4) | 0x01); - snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); + snd_soc_component_write(component, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); - snd_soc_write(codec, AIC31XX_PLLDMSB, + snd_soc_component_write(component, AIC31XX_PLLDMSB, aic31xx_divs[i].pll_d >> 8); - snd_soc_write(codec, AIC31XX_PLLDLSB, + snd_soc_component_write(component, AIC31XX_PLLDLSB, aic31xx_divs[i].pll_d & 0xff); /* DAC dividers configuration */ - snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK, + snd_soc_component_update_bits(component, AIC31XX_NDAC, AIC31XX_PLL_MASK, aic31xx_divs[i].ndac); - snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK, + snd_soc_component_update_bits(component, AIC31XX_MDAC, AIC31XX_PLL_MASK, aic31xx_divs[i].mdac); - snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8); - snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff); + snd_soc_component_write(component, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8); + snd_soc_component_write(component, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff); /* ADC dividers configuration. Write reset value 1 if not used. */ - snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK, + snd_soc_component_update_bits(component, AIC31XX_NADC, AIC31XX_PLL_MASK, aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1); - snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK, + snd_soc_component_update_bits(component, AIC31XX_MADC, AIC31XX_PLL_MASK, aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1); - snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); + snd_soc_component_write(component, AIC31XX_AOSR, aic31xx_divs[i].aosr); /* Bit clock divider configuration. */ - snd_soc_update_bits(codec, AIC31XX_BCLKN, + snd_soc_component_update_bits(component, AIC31XX_BCLKN, AIC31XX_PLL_MASK, bclk_n); aic31xx->rate_div_line = i; - dev_dbg(codec->dev, + dev_dbg(component->dev, "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, @@ -861,10 +861,10 @@ static int aic31xx_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; + struct snd_soc_component *component = dai->component; u8 data = 0; - dev_dbg(codec->dev, "## %s: width %d rate %d\n", + dev_dbg(component->dev, "## %s: width %d rate %d\n", __func__, params_width(params), params_rate(params)); @@ -884,28 +884,28 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream, AIC31XX_IFACE1_DATALEN_SHIFT); break; default: - dev_err(codec->dev, "%s: Unsupported width %d\n", + dev_err(component->dev, "%s: Unsupported width %d\n", __func__, params_width(params)); return -EINVAL; } - snd_soc_update_bits(codec, AIC31XX_IFACE1, + snd_soc_component_update_bits(component, AIC31XX_IFACE1, AIC31XX_IFACE1_DATALEN_MASK, data); - return aic31xx_setup_pll(codec, params); + return aic31xx_setup_pll(component, params); } static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) { - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_component *component = codec_dai->component; if (mute) { - snd_soc_update_bits(codec, AIC31XX_DACMUTE, + snd_soc_component_update_bits(component, AIC31XX_DACMUTE, AIC31XX_DACMUTE_MASK, AIC31XX_DACMUTE_MASK); } else { - snd_soc_update_bits(codec, AIC31XX_DACMUTE, + snd_soc_component_update_bits(component, AIC31XX_DACMUTE, AIC31XX_DACMUTE_MASK, 0x0); } @@ -915,12 +915,12 @@ static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_component *component = codec_dai->component; u8 iface_reg1 = 0; u8 iface_reg2 = 0; u8 dsp_a_val = 0; - dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); + dev_dbg(component->dev, "## %s: fmt = 0x%x\n", __func__, fmt); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -936,7 +936,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_CBS_CFS: break; default: - dev_err(codec->dev, "Invalid DAI master/slave interface\n"); + dev_err(component->dev, "Invalid DAI master/slave interface\n"); return -EINVAL; } @@ -948,7 +948,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, iface_reg2 |= AIC31XX_BCLKINV_MASK; break; default: - dev_err(codec->dev, "Invalid DAI clock signal polarity\n"); + dev_err(component->dev, "Invalid DAI clock signal polarity\n"); return -EINVAL; } @@ -977,18 +977,18 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, AIC31XX_IFACE1_DATATYPE_SHIFT); break; default: - dev_err(codec->dev, "Invalid DAI interface format\n"); + dev_err(component->dev, "Invalid DAI interface format\n"); return -EINVAL; } - snd_soc_update_bits(codec, AIC31XX_IFACE1, + snd_soc_component_update_bits(component, AIC31XX_IFACE1, AIC31XX_IFACE1_DATATYPE_MASK | AIC31XX_IFACE1_MASTER_MASK, iface_reg1); - snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET, + snd_soc_component_update_bits(component, AIC31XX_DATA_OFFSET, AIC31XX_DATA_OFFSET_MASK, dsp_a_val); - snd_soc_update_bits(codec, AIC31XX_IFACE2, + snd_soc_component_update_bits(component, AIC31XX_IFACE2, AIC31XX_BCLKINV_MASK, iface_reg2); @@ -998,11 +998,11 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); int i; - dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", + dev_dbg(component->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", __func__, clk_id, freq, dir); for (i = 1; i < 8; i++) @@ -1025,7 +1025,7 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, } /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ - snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK, + snd_soc_component_update_bits(component, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK, clk_id << AIC31XX_PLL_CLKIN_SHIFT); aic31xx->sysclk = freq; @@ -1071,42 +1071,42 @@ static int aic31xx_reset(struct aic31xx_priv *aic31xx) return ret; } -static void aic31xx_clk_on(struct snd_soc_codec *codec) +static void aic31xx_clk_on(struct snd_soc_component *component) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); u8 mask = AIC31XX_PM_MASK; u8 on = AIC31XX_PM_MASK; - dev_dbg(codec->dev, "codec clock -> on (rate %d)\n", + dev_dbg(component->dev, "codec clock -> on (rate %d)\n", aic31xx_divs[aic31xx->rate_div_line].rate); - snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on); + snd_soc_component_update_bits(component, AIC31XX_PLLPR, mask, on); mdelay(10); - snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on); - snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on); + snd_soc_component_update_bits(component, AIC31XX_NDAC, mask, on); + snd_soc_component_update_bits(component, AIC31XX_MDAC, mask, on); if (aic31xx_divs[aic31xx->rate_div_line].nadc) - snd_soc_update_bits(codec, AIC31XX_NADC, mask, on); + snd_soc_component_update_bits(component, AIC31XX_NADC, mask, on); if (aic31xx_divs[aic31xx->rate_div_line].madc) - snd_soc_update_bits(codec, AIC31XX_MADC, mask, on); - snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on); + snd_soc_component_update_bits(component, AIC31XX_MADC, mask, on); + snd_soc_component_update_bits(component, AIC31XX_BCLKN, mask, on); } -static void aic31xx_clk_off(struct snd_soc_codec *codec) +static void aic31xx_clk_off(struct snd_soc_component *component) { u8 mask = AIC31XX_PM_MASK; u8 off = 0; - dev_dbg(codec->dev, "codec clock -> off\n"); - snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off); - snd_soc_update_bits(codec, AIC31XX_MADC, mask, off); - snd_soc_update_bits(codec, AIC31XX_NADC, mask, off); - snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off); - snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off); - snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off); + dev_dbg(component->dev, "codec clock -> off\n"); + snd_soc_component_update_bits(component, AIC31XX_BCLKN, mask, off); + snd_soc_component_update_bits(component, AIC31XX_MADC, mask, off); + snd_soc_component_update_bits(component, AIC31XX_NADC, mask, off); + snd_soc_component_update_bits(component, AIC31XX_MDAC, mask, off); + snd_soc_component_update_bits(component, AIC31XX_NDAC, mask, off); + snd_soc_component_update_bits(component, AIC31XX_PLLPR, mask, off); } -static int aic31xx_power_on(struct snd_soc_codec *codec) +static int aic31xx_power_on(struct snd_soc_component *component) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); int ret; ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), @@ -1123,7 +1123,7 @@ static int aic31xx_power_on(struct snd_soc_codec *codec) ret = regcache_sync(aic31xx->regmap); if (ret) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to restore cache: %d\n", ret); regcache_cache_only(aic31xx->regmap, true); regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), @@ -1134,57 +1134,57 @@ static int aic31xx_power_on(struct snd_soc_codec *codec) return 0; } -static void aic31xx_power_off(struct snd_soc_codec *codec) +static void aic31xx_power_off(struct snd_soc_component *component) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); regcache_cache_only(aic31xx->regmap, true); regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), aic31xx->supplies); } -static int aic31xx_set_bias_level(struct snd_soc_codec *codec, +static int aic31xx_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { - dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, - snd_soc_codec_get_bias_level(codec), level); + dev_dbg(component->dev, "## %s: %d -> %d\n", __func__, + snd_soc_component_get_bias_level(component), level); switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) - aic31xx_clk_on(codec); + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_STANDBY) + aic31xx_clk_on(component); break; case SND_SOC_BIAS_STANDBY: - switch (snd_soc_codec_get_bias_level(codec)) { + switch (snd_soc_component_get_bias_level(component)) { case SND_SOC_BIAS_OFF: - aic31xx_power_on(codec); + aic31xx_power_on(component); break; case SND_SOC_BIAS_PREPARE: - aic31xx_clk_off(codec); + aic31xx_clk_off(component); break; default: BUG(); } break; case SND_SOC_BIAS_OFF: - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) - aic31xx_power_off(codec); + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_STANDBY) + aic31xx_power_off(component); break; } return 0; } -static int aic31xx_codec_probe(struct snd_soc_codec *codec) +static int aic31xx_codec_probe(struct snd_soc_component *component) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); int i, ret; dev_dbg(aic31xx->dev, "## %s\n", __func__); - aic31xx->codec = codec; + aic31xx->component = component; for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { aic31xx->disable_nb[i].nb.notifier_call = @@ -1193,7 +1193,7 @@ static int aic31xx_codec_probe(struct snd_soc_codec *codec) ret = regulator_register_notifier(aic31xx->supplies[i].consumer, &aic31xx->disable_nb[i].nb); if (ret) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to request regulator notifier: %d\n", ret); return ret; @@ -1203,43 +1203,42 @@ static int aic31xx_codec_probe(struct snd_soc_codec *codec) regcache_cache_only(aic31xx->regmap, true); regcache_mark_dirty(aic31xx->regmap); - ret = aic31xx_add_controls(codec); + ret = aic31xx_add_controls(component); if (ret) return ret; - ret = aic31xx_add_widgets(codec); + ret = aic31xx_add_widgets(component); if (ret) return ret; return 0; } -static int aic31xx_codec_remove(struct snd_soc_codec *codec) +static void aic31xx_codec_remove(struct snd_soc_component *component) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); int i; for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) regulator_unregister_notifier(aic31xx->supplies[i].consumer, &aic31xx->disable_nb[i].nb); - - return 0; } -static const struct snd_soc_codec_driver soc_codec_driver_aic31xx = { +static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, .remove = aic31xx_codec_remove, .set_bias_level = aic31xx_set_bias_level, - .suspend_bias_off = true, - - .component_driver = { - .controls = common31xx_snd_controls, - .num_controls = ARRAY_SIZE(common31xx_snd_controls), - .dapm_widgets = common31xx_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(common31xx_dapm_widgets), - .dapm_routes = common31xx_audio_map, - .num_dapm_routes = ARRAY_SIZE(common31xx_audio_map), - }, + .controls = common31xx_snd_controls, + .num_controls = ARRAY_SIZE(common31xx_snd_controls), + .dapm_widgets = common31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(common31xx_dapm_widgets), + .dapm_routes = common31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(common31xx_audio_map), + .suspend_bias_off = 1, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops aic31xx_dai_ops = { @@ -1375,23 +1374,17 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, } if (aic31xx->codec_type & DAC31XX_BIT) - return snd_soc_register_codec(&i2c->dev, + return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, dac31xx_dai_driver, ARRAY_SIZE(dac31xx_dai_driver)); else - return snd_soc_register_codec(&i2c->dev, + return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, ARRAY_SIZE(aic31xx_dai_driver)); } -static int aic31xx_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; -} - static const struct i2c_device_id aic31xx_i2c_id[] = { { "tlv320aic310x", AIC3100 }, { "tlv320aic311x", AIC3110 }, @@ -1412,7 +1405,6 @@ static struct i2c_driver aic31xx_i2c_driver = { .acpi_match_table = ACPI_PTR(aic31xx_acpi_match), }, .probe = aic31xx_i2c_probe, - .remove = aic31xx_i2c_remove, .id_table = aic31xx_i2c_id, }; module_i2c_driver(aic31xx_i2c_driver); -- cgit v1.2.3 From 064f6682f93898d55e6024a536dc1fad3843fa12 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 29 Jan 2018 04:39:54 +0000 Subject: ASoC: tfa9879: replace codec to component Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/tfa9879.c | 50 ++++++++++++++++++++-------------------------- 1 file changed, 22 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index e7ca764b5729..6d213c6d3920 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -30,8 +30,8 @@ static int tfa9879_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; - struct tfa9879_priv *tfa9879 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct tfa9879_priv *tfa9879 = snd_soc_component_get_drvdata(component); int fs; int i2s_set = 0; @@ -88,11 +88,11 @@ static int tfa9879_hw_params(struct snd_pcm_substream *substream, } if (tfa9879->lsb_justified) - snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1, + snd_soc_component_update_bits(component, TFA9879_SERIAL_INTERFACE_1, TFA9879_I2S_SET_MASK, i2s_set << TFA9879_I2S_SET_SHIFT); - snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1, + snd_soc_component_update_bits(component, TFA9879_SERIAL_INTERFACE_1, TFA9879_I2S_FS_MASK, fs << TFA9879_I2S_FS_SHIFT); return 0; @@ -100,9 +100,9 @@ static int tfa9879_hw_params(struct snd_pcm_substream *substream, static int tfa9879_digital_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; + struct snd_soc_component *component = dai->component; - snd_soc_update_bits(codec, TFA9879_MISC_CONTROL, + snd_soc_component_update_bits(component, TFA9879_MISC_CONTROL, TFA9879_S_MUTE_MASK, !!mute << TFA9879_S_MUTE_SHIFT); @@ -111,8 +111,8 @@ static int tfa9879_digital_mute(struct snd_soc_dai *dai, int mute) static int tfa9879_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - struct snd_soc_codec *codec = dai->codec; - struct tfa9879_priv *tfa9879 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct tfa9879_priv *tfa9879 = snd_soc_component_get_drvdata(component); int i2s_set; int sck_pol; @@ -151,10 +151,10 @@ static int tfa9879_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1, + snd_soc_component_update_bits(component, TFA9879_SERIAL_INTERFACE_1, TFA9879_SCK_POL_MASK, sck_pol << TFA9879_SCK_POL_SHIFT); - snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1, + snd_soc_component_update_bits(component, TFA9879_SERIAL_INTERFACE_1, TFA9879_I2S_SET_MASK, i2s_set << TFA9879_I2S_SET_SHIFT); return 0; @@ -230,15 +230,17 @@ static const struct snd_soc_dapm_route tfa9879_dapm_routes[] = { { "DAC", NULL, "POWER" }, }; -static const struct snd_soc_codec_driver tfa9879_codec = { - .component_driver = { - .controls = tfa9879_controls, - .num_controls = ARRAY_SIZE(tfa9879_controls), - .dapm_widgets = tfa9879_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tfa9879_dapm_widgets), - .dapm_routes = tfa9879_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(tfa9879_dapm_routes), - }, +static const struct snd_soc_component_driver tfa9879_component = { + .controls = tfa9879_controls, + .num_controls = ARRAY_SIZE(tfa9879_controls), + .dapm_widgets = tfa9879_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tfa9879_dapm_widgets), + .dapm_routes = tfa9879_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tfa9879_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static const struct regmap_config tfa9879_regmap = { @@ -295,17 +297,10 @@ static int tfa9879_i2c_probe(struct i2c_client *i2c, regmap_write(tfa9879->regmap, tfa9879_regs[i].reg, tfa9879_regs[i].def); - return snd_soc_register_codec(&i2c->dev, &tfa9879_codec, + return devm_snd_soc_register_component(&i2c->dev, &tfa9879_component, &tfa9879_dai, 1); } -static int tfa9879_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - - return 0; -} - static const struct i2c_device_id tfa9879_i2c_id[] = { { "tfa9879", 0 }, { } @@ -324,7 +319,6 @@ static struct i2c_driver tfa9879_i2c_driver = { .of_match_table = tfa9879_of_match, }, .probe = tfa9879_i2c_probe, - .remove = tfa9879_i2c_remove, .id_table = tfa9879_i2c_id, }; -- cgit v1.2.3 From d460b3f861e18b9c826abe178b2db57c6dc6b3e4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 14 Feb 2018 14:20:56 +0200 Subject: ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks In the reset state of the codec we do not have complete playback or capture routes. The audio playback/capture will not work due to missing clock signals on the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down. To make sure that even if all output/input is disconnected the codec is generating clocks, we need to have valid DAPM route in every case to power up the must needed parts of the codec. I have verified that switching DAC (during playback) or ADC (during capture) will stop the I2S clocks, so the only solution is to connect the 'Off' routes as well to output/input. The routes will be only added if the codec is clock master. In case the role changes runtime, the applied routes are removed. Tested on am43x-epos-evm with aic3111 codec in master mode. Signed-off-by: Peter Ujfalusi Reviewed-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 73 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 72 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3cd924dc300..7090342e8285 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -166,6 +166,7 @@ struct aic31xx_priv { unsigned int sysclk; u8 p_div; int rate_div_line; + bool master_dapm_route_applied; }; struct aic31xx_rate_divs { @@ -670,6 +671,29 @@ aic310x_audio_map[] = { {"SPK", NULL, "SPK ClassD"}, }; +/* + * Always connected DAPM routes for codec clock master modes. + * If the codec is the master on the I2S bus, we need to power on components + * to have valid DAC_CLK and also the DACs and ADC for playback/capture. + * Otherwise the codec will not generate clocks on the bus. + */ +static const struct snd_soc_dapm_route +common31xx_cm_audio_map[] = { + {"DAC Left Input", "Off", "DAC IN"}, + {"DAC Right Input", "Off", "DAC IN"}, + + {"HPL", NULL, "DAC Left"}, + {"HPR", NULL, "DAC Right"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_cm_audio_map[] = { + {"MIC1LP P-Terminal", "Off", "MIC1LP"}, + {"MIC1RP P-Terminal", "Off", "MIC1RP"}, + {"MIC1LM P-Terminal", "Off", "MIC1LM"}, + {"MIC1LM M-Terminal", "Off", "MIC1LM"}, +}; + static int aic31xx_add_controls(struct snd_soc_component *component) { int ret = 0; @@ -912,6 +936,53 @@ static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static int aic31xx_clock_master_routes(struct snd_soc_component *component, + unsigned int fmt) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + int ret; + + fmt &= SND_SOC_DAIFMT_MASTER_MASK; + if (fmt == SND_SOC_DAIFMT_CBS_CFS && + aic31xx->master_dapm_route_applied) { + /* + * Remove the DAPM route(s) for codec clock master modes, + * if applied + */ + ret = snd_soc_dapm_del_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_del_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = false; + } else if (fmt != SND_SOC_DAIFMT_CBS_CFS && + !aic31xx->master_dapm_route_applied) { + /* + * Add the needed DAPM route(s) for codec clock master modes, + * if it is not done already + */ + ret = snd_soc_dapm_add_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_add_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = true; + } + + return 0; +} + static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -992,7 +1063,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, AIC31XX_BCLKINV_MASK, iface_reg2); - return 0; + return aic31xx_clock_master_routes(component, fmt); } static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, -- cgit v1.2.3 From 7d41bc28e9fa3fed28691019ca0fe36ed68d7b86 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 19 Feb 2018 11:45:07 +0200 Subject: ASoC: tlv320aic31xx: Rename AIF_IN from 'DAC IN' to 'AIF IN' The audio interface is not really the DAC input. Use more generic name for it. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 7090342e8285..e5a1e2be17aa 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -468,7 +468,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, } static const struct snd_soc_dapm_widget common31xx_dapm_widgets[] = { - SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF IN", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("DAC Left Input", SND_SOC_NOPM, 0, 0, &ldac_in_control), @@ -584,12 +584,12 @@ static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { static const struct snd_soc_dapm_route common31xx_audio_map[] = { /* DAC Input Routing */ - {"DAC Left Input", "Left Data", "DAC IN"}, - {"DAC Left Input", "Right Data", "DAC IN"}, - {"DAC Left Input", "Mono", "DAC IN"}, - {"DAC Right Input", "Left Data", "DAC IN"}, - {"DAC Right Input", "Right Data", "DAC IN"}, - {"DAC Right Input", "Mono", "DAC IN"}, + {"DAC Left Input", "Left Data", "AIF IN"}, + {"DAC Left Input", "Right Data", "AIF IN"}, + {"DAC Left Input", "Mono", "AIF IN"}, + {"DAC Right Input", "Left Data", "AIF IN"}, + {"DAC Right Input", "Right Data", "AIF IN"}, + {"DAC Right Input", "Mono", "AIF IN"}, {"DAC Left", NULL, "DAC Left Input"}, {"DAC Right", NULL, "DAC Right Input"}, @@ -679,8 +679,8 @@ aic310x_audio_map[] = { */ static const struct snd_soc_dapm_route common31xx_cm_audio_map[] = { - {"DAC Left Input", "Off", "DAC IN"}, - {"DAC Right Input", "Off", "DAC IN"}, + {"DAC Left Input", "Off", "AIF IN"}, + {"DAC Right Input", "Off", "AIF IN"}, {"HPL", NULL, "DAC Left"}, {"HPR", NULL, "DAC Right"}, -- cgit v1.2.3 From a16be2a6e2ffa8ad5e24e96289f317e1b5b8c17a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 19 Feb 2018 11:45:08 +0200 Subject: ASoC: tlv320aic31xx: Do not force power on the DAC/ADC in clock master mode MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With P0 R29, Bit2 set the I2S clocks will be running when the DAC/ADC is powered down, but still the codec need to be powered up by needing at least one complete DAPM path for the stream. If the AIF is not needed (analog loopback for example) the I2S clocks will not run as they are not needed. Signed-off-by: Peter Ujfalusi Suggested-by: Stefan Müller-Klieser Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 32 ++++++++++++++++++++++---------- sound/soc/codecs/tlv320aic31xx.h | 1 + 2 files changed, 23 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index e5a1e2be17aa..bf92d36b8f8a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -501,6 +501,10 @@ static const struct snd_soc_dapm_widget common31xx_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + /* Keep BCLK/WCLK enabled even if DAC/ADC is powered down */ + SND_SOC_DAPM_SUPPLY("Activate I2S clocks", AIC31XX_IFACE2, 2, 0, + NULL, 0), + /* Outputs */ SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), @@ -553,6 +557,8 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, aic31xx_right_output_switches, ARRAY_SIZE(aic31xx_right_output_switches)), + + SND_SOC_DAPM_AIF_OUT("AIF OUT", "Capture", 0, SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { @@ -640,6 +646,8 @@ aic31xx_audio_map[] = { {"ADC", NULL, "MIC_GAIN_CTL"}, + {"AIF OUT", NULL, "ADC"}, + /* Left Output */ {"Output Left", "From Left DAC", "DAC Left"}, {"Output Left", "From MIC1LP", "MIC1LP"}, @@ -673,25 +681,29 @@ aic310x_audio_map[] = { /* * Always connected DAPM routes for codec clock master modes. - * If the codec is the master on the I2S bus, we need to power on components - * to have valid DAC_CLK and also the DACs and ADC for playback/capture. + * If the codec is the master on the I2S bus, we need to power up components + * to have valid DAC_CLK. + * + * In order to have the I2S clocks on the bus either the DACs/ADC need to be + * enabled, or the P0/R29/D2 (Keep bclk/wclk in power down) need to be set. + * * Otherwise the codec will not generate clocks on the bus. */ static const struct snd_soc_dapm_route common31xx_cm_audio_map[] = { - {"DAC Left Input", "Off", "AIF IN"}, - {"DAC Right Input", "Off", "AIF IN"}, + {"HPL", NULL, "AIF IN"}, + {"HPR", NULL, "AIF IN"}, - {"HPL", NULL, "DAC Left"}, - {"HPR", NULL, "DAC Right"}, + {"AIF IN", NULL, "Activate I2S clocks"}, }; static const struct snd_soc_dapm_route aic31xx_cm_audio_map[] = { - {"MIC1LP P-Terminal", "Off", "MIC1LP"}, - {"MIC1RP P-Terminal", "Off", "MIC1RP"}, - {"MIC1LM P-Terminal", "Off", "MIC1LM"}, - {"MIC1LM M-Terminal", "Off", "MIC1LM"}, + {"AIF OUT", NULL, "MIC1LP"}, + {"AIF OUT", NULL, "MIC1RP"}, + {"AIF OUT", NULL, "MIC1LM"}, + + {"AIF OUT", NULL, "Activate I2S clocks"}, }; static int aic31xx_add_controls(struct snd_soc_component *component) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 15ac7cba86fe..0b587585b38b 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -160,6 +160,7 @@ struct aic31xx_pdata { #define AIC31XX_DACMOD2BCLK 0x01 #define AIC31XX_ADC2BCLK 0x02 #define AIC31XX_ADCMOD2BCLK 0x03 +#define AIC31XX_KEEP_I2SCLK BIT(2) /* AIC31XX_ADCFLAG */ #define AIC31XX_ADCPWRSTATUS_MASK BIT(6) -- cgit v1.2.3 From 3d3db9432853b8b198722768ba0788b1bc586049 Mon Sep 17 00:00:00 2001 From: Matt Porter Date: Sun, 18 Mar 2018 13:22:38 -0400 Subject: ASoC: add tda7419 audio processor driver Component driver for the tda7419 audio processor. Signed-off-by: Matt Porter Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda7419.c | 654 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 662 insertions(+) create mode 100644 sound/soc/codecs/tda7419.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2b331f7266ab..1553cf2b9445 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -151,6 +151,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS571X if I2C select SND_SOC_TAS5720 if I2C select SND_SOC_TAS6424 if I2C + select SND_SOC_TDA7419 if I2C select SND_SOC_TFA9879 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER @@ -910,6 +911,11 @@ config SND_SOC_TAS6424 Enable support for Texas Instruments TAS6424 high-efficiency digital input quad-channel Class-D audio power amplifiers. +config SND_SOC_TDA7419 + tristate "ST TDA7419 audio processor" + depends on I2C + select REGMAP_I2C + config SND_SOC_TFA9879 tristate "NXP Semiconductors TFA9879 amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index da1571336f1e..6cf3c3b92cb5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -160,6 +160,7 @@ snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o snd-soc-tas5720-objs := tas5720.o snd-soc-tas6424-objs := tas6424.o +snd-soc-tda7419-objs := tda7419.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -405,6 +406,7 @@ obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o obj-$(CONFIG_SND_SOC_TAS6424) += snd-soc-tas6424.o +obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c new file mode 100644 index 000000000000..225c210ac38f --- /dev/null +++ b/sound/soc/codecs/tda7419.c @@ -0,0 +1,654 @@ +/* + * TDA7419 audio processor driver + * + * Copyright 2018 Konsulko Group + * + * Author: Matt Porter + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#define TDA7419_MAIN_SRC_REG 0x00 +#define TDA7419_LOUDNESS_REG 0x01 +#define TDA7419_MUTE_CLK_REG 0x02 +#define TDA7419_VOLUME_REG 0x03 +#define TDA7419_TREBLE_REG 0x04 +#define TDA7419_MIDDLE_REG 0x05 +#define TDA7419_BASS_REG 0x06 +#define TDA7419_SECOND_SRC_REG 0x07 +#define TDA7419_SUB_MID_BASS_REG 0x08 +#define TDA7419_MIXING_GAIN_REG 0x09 +#define TDA7419_ATTENUATOR_LF_REG 0x0a +#define TDA7419_ATTENUATOR_RF_REG 0x0b +#define TDA7419_ATTENUATOR_LR_REG 0x0c +#define TDA7419_ATTENUATOR_RR_REG 0x0d +#define TDA7419_MIXING_LEVEL_REG 0x0e +#define TDA7419_ATTENUATOR_SUB_REG 0x0f +#define TDA7419_SA_CLK_AC_REG 0x10 +#define TDA7419_TESTING_REG 0x11 + +#define TDA7419_MAIN_SRC_SEL 0 +#define TDA7419_MAIN_SRC_GAIN 3 +#define TDA7419_MAIN_SRC_AUTOZERO 7 + +#define TDA7419_LOUDNESS_ATTEN 0 +#define TDA7419_LOUDNESS_CENTER_FREQ 4 +#define TDA7419_LOUDNESS_BOOST 6 +#define TDA7419_LOUDNESS_SOFT_STEP 7 + +#define TDA7419_VOLUME_SOFT_STEP 7 + +#define TDA7419_SOFT_MUTE 0 +#define TDA7419_MUTE_INFLUENCE 1 +#define TDA7419_SOFT_MUTE_TIME 2 +#define TDA7419_SOFT_STEP_TIME 4 +#define TDA7419_CLK_FAST_MODE 7 + +#define TDA7419_TREBLE_CENTER_FREQ 5 +#define TDA7419_REF_OUT_SELECT 7 + +#define TDA7419_MIDDLE_Q_FACTOR 5 +#define TDA7419_MIDDLE_SOFT_STEP 7 + +#define TDA7419_BASS_Q_FACTOR 5 +#define TDA7419_BASS_SOFT_STEP 7 + +#define TDA7419_SECOND_SRC_SEL 0 +#define TDA7419_SECOND_SRC_GAIN 3 +#define TDA7419_REAR_SPKR_SRC 7 + +#define TDA7419_SUB_CUT_OFF_FREQ 0 +#define TDA7419_MIDDLE_CENTER_FREQ 2 +#define TDA7419_BASS_CENTER_FREQ 4 +#define TDA7419_BASS_DC_MODE 6 +#define TDA7419_SMOOTHING_FILTER 7 + +#define TDA7419_MIX_LF 0 +#define TDA7419_MIX_RF 1 +#define TDA7419_MIX_ENABLE 2 +#define TDA7419_SUB_ENABLE 3 +#define TDA7419_HPF_GAIN 4 + +#define TDA7419_SA_Q_FACTOR 0 +#define TDA7419_RESET_MODE 1 +#define TDA7419_SA_SOURCE 2 +#define TDA7419_SA_RUN 3 +#define TDA7419_RESET 4 +#define TDA7419_CLK_SOURCE 5 +#define TDA7419_COUPLING_MODE 6 + +struct tda7419_data { + struct regmap *regmap; +}; + +static bool tda7419_readable_reg(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct reg_default tda7419_regmap_defaults[] = { + { TDA7419_MAIN_SRC_REG, 0xfe }, + { TDA7419_LOUDNESS_REG, 0xfe }, + { TDA7419_MUTE_CLK_REG, 0xfe }, + { TDA7419_VOLUME_REG, 0xfe }, + { TDA7419_TREBLE_REG, 0xfe }, + { TDA7419_MIDDLE_REG, 0xfe }, + { TDA7419_BASS_REG, 0xfe }, + { TDA7419_SECOND_SRC_REG, 0xfe }, + { TDA7419_SUB_MID_BASS_REG, 0xfe }, + { TDA7419_MIXING_GAIN_REG, 0xfe }, + { TDA7419_ATTENUATOR_LF_REG, 0xfe }, + { TDA7419_ATTENUATOR_RF_REG, 0xfe }, + { TDA7419_ATTENUATOR_LR_REG, 0xfe }, + { TDA7419_ATTENUATOR_RR_REG, 0xfe }, + { TDA7419_MIXING_LEVEL_REG, 0xfe }, + { TDA7419_ATTENUATOR_SUB_REG, 0xfe }, + { TDA7419_SA_CLK_AC_REG, 0xfe }, + { TDA7419_TESTING_REG, 0xfe }, +}; + +static const struct regmap_config tda7419_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = TDA7419_TESTING_REG, + .cache_type = REGCACHE_RBTREE, + .readable_reg = tda7419_readable_reg, + .reg_defaults = tda7419_regmap_defaults, + .num_reg_defaults = ARRAY_SIZE(tda7419_regmap_defaults), +}; + +struct tda7419_vol_control { + int min, max; + unsigned int reg, rreg, mask, thresh; + unsigned int invert:1; +}; + +static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc) +{ + if (tvc->reg == tvc->rreg) + return 0; + + return 1; +} + +static int tda7419_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tda7419_vol_control *tvc = + (struct tda7419_vol_control *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = tda7419_vol_is_stereo(tvc) ? 2 : 1; + uinfo->value.integer.min = tvc->min; + uinfo->value.integer.max = tvc->max; + + return 0; +} + +static inline int tda7419_vol_get_value(int val, unsigned int mask, + int min, int thresh, + unsigned int invert) +{ + val &= mask; + if (val < thresh) { + if (invert) + val = 0 - val; + } else if (val > thresh) { + if (invert) + val = val - thresh; + else + val = thresh - val; + } + + if (val < min) + val = min; + + return val; +} + +static int tda7419_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct tda7419_vol_control *tvc = + (struct tda7419_vol_control *)kcontrol->private_value; + unsigned int reg = tvc->reg; + unsigned int rreg = tvc->rreg; + unsigned int mask = tvc->mask; + int min = tvc->min; + int thresh = tvc->thresh; + unsigned int invert = tvc->invert; + int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret < 0) + return ret; + ucontrol->value.integer.value[0] = + tda7419_vol_get_value(val, mask, min, thresh, invert); + + if (tda7419_vol_is_stereo(tvc)) { + ret = snd_soc_component_read(component, rreg, &val); + if (ret < 0) + return ret; + ucontrol->value.integer.value[1] = + tda7419_vol_get_value(val, mask, min, thresh, invert); + } + + return 0; +} + +static inline int tda7419_vol_put_value(int val, int thresh, + unsigned int invert) +{ + if (val < 0) { + if (invert) + val = abs(val); + else + val = thresh - val; + } else if ((val > 0) && invert) { + val += thresh; + } + + return val; +} + +static int tda7419_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_kcontrol_chip(kcontrol); + struct tda7419_vol_control *tvc = + (struct tda7419_vol_control *)kcontrol->private_value; + unsigned int reg = tvc->reg; + unsigned int rreg = tvc->rreg; + unsigned int mask = tvc->mask; + int thresh = tvc->thresh; + unsigned int invert = tvc->invert; + int val; + int ret; + + val = tda7419_vol_put_value(ucontrol->value.integer.value[0], + thresh, invert); + ret = snd_soc_component_update_bits(component, reg, + mask, val); + if (ret < 0) + return ret; + + if (tda7419_vol_is_stereo(tvc)) { + val = tda7419_vol_put_value(ucontrol->value.integer.value[1], + thresh, invert); + ret = snd_soc_component_update_bits(component, rreg, + mask, val); + } + + return ret; +} + +#define TDA7419_SINGLE_VALUE(xreg, xmask, xmin, xmax, xthresh, xinvert) \ + ((unsigned long)&(struct tda7419_vol_control) \ + {.reg = xreg, .rreg = xreg, .mask = xmask, .min = xmin, \ + .max = xmax, .thresh = xthresh, .invert = xinvert}) + +#define TDA7419_DOUBLE_R_VALUE(xregl, xregr, xmask, xmin, xmax, xthresh, \ + xinvert) \ + ((unsigned long)&(struct tda7419_vol_control) \ + {.reg = xregl, .rreg = xregr, .mask = xmask, .min = xmin, \ + .max = xmax, .thresh = xthresh, .invert = xinvert}) + +#define TDA7419_SINGLE_TLV(xname, xreg, xmask, xmin, xmax, xthresh, \ + xinvert, xtlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (xtlv_array), \ + .info = tda7419_vol_info, \ + .get = tda7419_vol_get, \ + .put = tda7419_vol_put, \ + .private_value = TDA7419_SINGLE_VALUE(xreg, xmask, xmin, \ + xmax, xthresh, xinvert), \ +} + +#define TDA7419_DOUBLE_R_TLV(xname, xregl, xregr, xmask, xmin, xmax, \ + xthresh, xinvert, xtlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (xtlv_array), \ + .info = tda7419_vol_info, \ + .get = tda7419_vol_get, \ + .put = tda7419_vol_put, \ + .private_value = TDA7419_DOUBLE_R_VALUE(xregl, xregr, xmask, \ + xmin, xmax, xthresh, \ + xinvert), \ +} + +static const char * const enum_src_sel[] = { + "QD", "SE1", "SE2", "SE3", "SE", "Mute", "Mute", "Mute"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_main_src_sel, + TDA7419_MAIN_SRC_REG, TDA7419_MAIN_SRC_SEL, enum_src_sel); +static const struct snd_kcontrol_new soc_mux_main_src_sel = + SOC_DAPM_ENUM("Main Source Select", soc_enum_main_src_sel); +static DECLARE_TLV_DB_SCALE(tlv_src_gain, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(tlv_loudness_atten, -1500, 100, 0); +static const char * const enum_loudness_center_freq[] = { + "Flat", "400 Hz", "800 Hz", "2400 Hz"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_loudness_center_freq, + TDA7419_LOUDNESS_REG, TDA7419_LOUDNESS_CENTER_FREQ, + enum_loudness_center_freq); +static const char * const enum_mute_influence[] = { + "Pin and IIC", "IIC"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_mute_influence, + TDA7419_MUTE_CLK_REG, TDA7419_MUTE_INFLUENCE, enum_mute_influence); +static const char * const enum_soft_mute_time[] = { + "0.48 ms", "0.96 ms", "123 ms", "123 ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_soft_mute_time, + TDA7419_MUTE_CLK_REG, TDA7419_SOFT_MUTE_TIME, enum_soft_mute_time); +static const char * const enum_soft_step_time[] = { + "0.160 ms", "0.321 ms", "0.642 ms", "1.28 ms", + "2.56 ms", "5.12 ms", "10.24 ms", "20.48 ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_soft_step_time, + TDA7419_MUTE_CLK_REG, TDA7419_SOFT_STEP_TIME, enum_soft_step_time); +static DECLARE_TLV_DB_SCALE(tlv_volume, -8000, 100, 1); +static const char * const enum_treble_center_freq[] = { + "10.0 kHz", "12.5 kHz", "15.0 kHz", "17.5 kHz"}; +static DECLARE_TLV_DB_SCALE(tlv_filter, -1500, 100, 0); +static SOC_ENUM_SINGLE_DECL(soc_enum_treble_center_freq, + TDA7419_TREBLE_REG, TDA7419_TREBLE_CENTER_FREQ, + enum_treble_center_freq); +static const char * const enum_ref_out_select[] = { + "External Vref (4 V)", "Internal Vref (3.3 V)"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_ref_out_select, + TDA7419_TREBLE_REG, TDA7419_REF_OUT_SELECT, enum_ref_out_select); +static const char * const enum_middle_q_factor[] = { + "0.5", "0.75", "1.0", "1.25"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_middle_q_factor, + TDA7419_MIDDLE_REG, TDA7419_MIDDLE_Q_FACTOR, enum_middle_q_factor); +static const char * const enum_bass_q_factor[] = { + "1.0", "1.25", "1.5", "2.0"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bass_q_factor, + TDA7419_BASS_REG, TDA7419_BASS_Q_FACTOR, enum_bass_q_factor); +static SOC_ENUM_SINGLE_DECL(soc_enum_second_src_sel, + TDA7419_SECOND_SRC_REG, TDA7419_SECOND_SRC_SEL, enum_src_sel); +static const struct snd_kcontrol_new soc_mux_second_src_sel = + SOC_DAPM_ENUM("Second Source Select", soc_enum_second_src_sel); +static const char * const enum_rear_spkr_src[] = { + "Main", "Second"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_rear_spkr_src, + TDA7419_SECOND_SRC_REG, TDA7419_REAR_SPKR_SRC, enum_rear_spkr_src); +static const struct snd_kcontrol_new soc_mux_rear_spkr_src = + SOC_DAPM_ENUM("Rear Speaker Source", soc_enum_rear_spkr_src); +static const char * const enum_sub_cut_off_freq[] = { + "Flat", "80 Hz", "120 Hz", "160 Hz"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_sub_cut_off_freq, + TDA7419_SUB_MID_BASS_REG, TDA7419_SUB_CUT_OFF_FREQ, + enum_sub_cut_off_freq); +static const char * const enum_middle_center_freq[] = { + "500 Hz", "1000 Hz", "1500 Hz", "2500 Hz"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_middle_center_freq, + TDA7419_SUB_MID_BASS_REG, TDA7419_MIDDLE_CENTER_FREQ, + enum_middle_center_freq); +static const char * const enum_bass_center_freq[] = { + "60 Hz", "80 Hz", "100 Hz", "200 Hz"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bass_center_freq, + TDA7419_SUB_MID_BASS_REG, TDA7419_BASS_CENTER_FREQ, + enum_bass_center_freq); +static const char * const enum_sa_q_factor[] = { + "3.5", "1.75" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_sa_q_factor, + TDA7419_SA_CLK_AC_REG, TDA7419_SA_Q_FACTOR, enum_sa_q_factor); +static const char * const enum_reset_mode[] = { + "IIC", "Auto" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_reset_mode, + TDA7419_SA_CLK_AC_REG, TDA7419_RESET_MODE, enum_reset_mode); +static const char * const enum_sa_src[] = { + "Bass", "In Gain" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_sa_src, + TDA7419_SA_CLK_AC_REG, TDA7419_SA_SOURCE, enum_sa_src); +static const char * const enum_clk_src[] = { + "Internal", "External" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_clk_src, + TDA7419_SA_CLK_AC_REG, TDA7419_CLK_SOURCE, enum_clk_src); +static const char * const enum_coupling_mode[] = { + "DC Coupling (without HPF)", "AC Coupling after In Gain", + "DC Coupling (with HPF)", "AC Coupling after Bass" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_coupling_mode, + TDA7419_SA_CLK_AC_REG, TDA7419_COUPLING_MODE, enum_coupling_mode); + +/* ASoC Controls */ +static struct snd_kcontrol_new tda7419_controls[] = { +SOC_SINGLE_TLV("Main Source Capture Volume", TDA7419_MAIN_SRC_REG, + TDA7419_MAIN_SRC_GAIN, 15, 0, tlv_src_gain), +SOC_SINGLE("Main Source AutoZero Switch", TDA7419_MAIN_SRC_REG, + TDA7419_MAIN_SRC_AUTOZERO, 1, 1), +SOC_SINGLE_TLV("Loudness Playback Volume", TDA7419_LOUDNESS_REG, + TDA7419_LOUDNESS_ATTEN, 15, 1, tlv_loudness_atten), +SOC_ENUM("Loudness Center Frequency", soc_enum_loudness_center_freq), +SOC_SINGLE("Loudness High Boost Switch", TDA7419_LOUDNESS_REG, + TDA7419_LOUDNESS_BOOST, 1, 1), +SOC_SINGLE("Loudness Soft Step Switch", TDA7419_LOUDNESS_REG, + TDA7419_LOUDNESS_SOFT_STEP, 1, 1), +SOC_SINGLE("Soft Mute Switch", TDA7419_MUTE_CLK_REG, TDA7419_SOFT_MUTE, 1, 1), +SOC_ENUM("Mute Influence", soc_enum_mute_influence), +SOC_ENUM("Soft Mute Time", soc_enum_soft_mute_time), +SOC_ENUM("Soft Step Time", soc_enum_soft_step_time), +SOC_SINGLE("Clock Fast Mode Switch", TDA7419_MUTE_CLK_REG, + TDA7419_CLK_FAST_MODE, 1, 1), +TDA7419_SINGLE_TLV("Master Playback Volume", TDA7419_VOLUME_REG, + 0x7f, -80, 15, 0x10, 0, tlv_volume), +SOC_SINGLE("Volume Soft Step Switch", TDA7419_VOLUME_REG, + TDA7419_VOLUME_SOFT_STEP, 1, 1), +TDA7419_SINGLE_TLV("Treble Playback Volume", TDA7419_TREBLE_REG, + 0x1f, -15, 15, 0x10, 1, tlv_filter), +SOC_ENUM("Treble Center Frequency", soc_enum_treble_center_freq), +SOC_ENUM("Reference Output Select", soc_enum_ref_out_select), +TDA7419_SINGLE_TLV("Middle Playback Volume", TDA7419_MIDDLE_REG, + 0x1f, -15, 15, 0x10, 1, tlv_filter), +SOC_ENUM("Middle Q Factor", soc_enum_middle_q_factor), +SOC_SINGLE("Middle Soft Step Switch", TDA7419_MIDDLE_REG, + TDA7419_MIDDLE_SOFT_STEP, 1, 1), +TDA7419_SINGLE_TLV("Bass Playback Volume", TDA7419_BASS_REG, + 0x1f, -15, 15, 0x10, 1, tlv_filter), +SOC_ENUM("Bass Q Factor", soc_enum_bass_q_factor), +SOC_SINGLE("Bass Soft Step Switch", TDA7419_BASS_REG, + TDA7419_BASS_SOFT_STEP, 1, 1), +SOC_SINGLE_TLV("Second Source Capture Volume", TDA7419_SECOND_SRC_REG, + TDA7419_SECOND_SRC_GAIN, 15, 0, tlv_src_gain), +SOC_ENUM("Subwoofer Cut-off Frequency", soc_enum_sub_cut_off_freq), +SOC_ENUM("Middle Center Frequency", soc_enum_middle_center_freq), +SOC_ENUM("Bass Center Frequency", soc_enum_bass_center_freq), +SOC_SINGLE("Bass DC Mode Switch", TDA7419_SUB_MID_BASS_REG, + TDA7419_BASS_DC_MODE, 1, 1), +SOC_SINGLE("Smoothing Filter Switch", TDA7419_SUB_MID_BASS_REG, + TDA7419_SMOOTHING_FILTER, 1, 1), +TDA7419_DOUBLE_R_TLV("Front Speaker Playback Volume", TDA7419_ATTENUATOR_LF_REG, + TDA7419_ATTENUATOR_RF_REG, 0x7f, -80, 15, 0x10, 0, + tlv_volume), +SOC_SINGLE("Left Front Soft Step Switch", TDA7419_ATTENUATOR_LF_REG, + TDA7419_VOLUME_SOFT_STEP, 1, 1), +SOC_SINGLE("Right Front Soft Step Switch", TDA7419_ATTENUATOR_RF_REG, + TDA7419_VOLUME_SOFT_STEP, 1, 1), +TDA7419_DOUBLE_R_TLV("Rear Speaker Playback Volume", TDA7419_ATTENUATOR_LR_REG, + TDA7419_ATTENUATOR_RR_REG, 0x7f, -80, 15, 0x10, 0, + tlv_volume), +SOC_SINGLE("Left Rear Soft Step Switch", TDA7419_ATTENUATOR_LR_REG, + TDA7419_VOLUME_SOFT_STEP, 1, 1), +SOC_SINGLE("Right Rear Soft Step Switch", TDA7419_ATTENUATOR_RR_REG, + TDA7419_VOLUME_SOFT_STEP, 1, 1), +TDA7419_SINGLE_TLV("Mixing Capture Volume", TDA7419_MIXING_LEVEL_REG, + 0x7f, -80, 15, 0x10, 0, tlv_volume), +SOC_SINGLE("Mixing Level Soft Step Switch", TDA7419_MIXING_LEVEL_REG, + TDA7419_VOLUME_SOFT_STEP, 1, 1), +TDA7419_SINGLE_TLV("Subwoofer Playback Volume", TDA7419_ATTENUATOR_SUB_REG, + 0x7f, -80, 15, 0x10, 0, tlv_volume), +SOC_SINGLE("Subwoofer Soft Step Switch", TDA7419_ATTENUATOR_SUB_REG, + TDA7419_VOLUME_SOFT_STEP, 1, 1), +SOC_ENUM("Spectrum Analyzer Q Factor", soc_enum_sa_q_factor), +SOC_ENUM("Spectrum Analyzer Reset Mode", soc_enum_reset_mode), +SOC_ENUM("Spectrum Analyzer Source", soc_enum_sa_src), +SOC_SINGLE("Spectrum Analyzer Run Switch", TDA7419_SA_CLK_AC_REG, + TDA7419_SA_RUN, 1, 1), +SOC_SINGLE("Spectrum Analyzer Reset Switch", TDA7419_SA_CLK_AC_REG, + TDA7419_RESET, 1, 1), +SOC_ENUM("Clock Source", soc_enum_clk_src), +SOC_ENUM("Coupling Mode", soc_enum_coupling_mode), +}; + +static const struct snd_kcontrol_new soc_mixer_lf_output_controls[] = { + SOC_DAPM_SINGLE("Mix to LF Speaker Switch", + TDA7419_MIXING_GAIN_REG, + TDA7419_MIX_LF, 1, 1), +}; + +static const struct snd_kcontrol_new soc_mixer_rf_output_controls[] = { + SOC_DAPM_SINGLE("Mix to RF Speaker Switch", + TDA7419_MIXING_GAIN_REG, + TDA7419_MIX_RF, 1, 1), +}; + +static const struct snd_kcontrol_new soc_mix_enable_switch_controls[] = { + SOC_DAPM_SINGLE("Switch", TDA7419_MIXING_GAIN_REG, + TDA7419_MIX_ENABLE, 1, 1), +}; + +static const struct snd_kcontrol_new soc_sub_enable_switch_controls[] = { + SOC_DAPM_SINGLE("Switch", TDA7419_MIXING_GAIN_REG, + TDA7419_MIX_ENABLE, 1, 1), +}; + +static const struct snd_soc_dapm_widget tda7419_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("SE3L"), + SND_SOC_DAPM_INPUT("SE3R"), + SND_SOC_DAPM_INPUT("SE2L"), + SND_SOC_DAPM_INPUT("SE2R"), + SND_SOC_DAPM_INPUT("SE1L"), + SND_SOC_DAPM_INPUT("SE1R"), + SND_SOC_DAPM_INPUT("DIFFL"), + SND_SOC_DAPM_INPUT("DIFFR"), + SND_SOC_DAPM_INPUT("MIX"), + + SND_SOC_DAPM_MUX("Main Source Select", SND_SOC_NOPM, + 0, 0, &soc_mux_main_src_sel), + SND_SOC_DAPM_MUX("Second Source Select", SND_SOC_NOPM, + 0, 0, &soc_mux_second_src_sel), + SND_SOC_DAPM_MUX("Rear Speaker Source", SND_SOC_NOPM, + 0, 0, &soc_mux_rear_spkr_src), + + SND_SOC_DAPM_SWITCH("Mix Enable", SND_SOC_NOPM, + 0, 0, &soc_mix_enable_switch_controls[0]), + SND_SOC_DAPM_MIXER_NAMED_CTL("LF Output Mixer", SND_SOC_NOPM, + 0, 0, &soc_mixer_lf_output_controls[0], + ARRAY_SIZE(soc_mixer_lf_output_controls)), + SND_SOC_DAPM_MIXER_NAMED_CTL("RF Output Mixer", SND_SOC_NOPM, + 0, 0, &soc_mixer_rf_output_controls[0], + ARRAY_SIZE(soc_mixer_rf_output_controls)), + + SND_SOC_DAPM_SWITCH("Subwoofer Enable", + SND_SOC_NOPM, 0, 0, + &soc_sub_enable_switch_controls[0]), + + SND_SOC_DAPM_OUTPUT("OUTLF"), + SND_SOC_DAPM_OUTPUT("OUTRF"), + SND_SOC_DAPM_OUTPUT("OUTLR"), + SND_SOC_DAPM_OUTPUT("OUTRR"), + SND_SOC_DAPM_OUTPUT("OUTSW"), +}; + +static const struct snd_soc_dapm_route tda7419_dapm_routes[] = { + {"Main Source Select", "SE3", "SE3L"}, + {"Main Source Select", "SE3", "SE3R"}, + {"Main Source Select", "SE2", "SE2L"}, + {"Main Source Select", "SE2", "SE2R"}, + {"Main Source Select", "SE1", "SE1L"}, + {"Main Source Select", "SE1", "SE1R"}, + {"Main Source Select", "SE", "DIFFL"}, + {"Main Source Select", "SE", "DIFFR"}, + {"Main Source Select", "QD", "DIFFL"}, + {"Main Source Select", "QD", "DIFFR"}, + + {"Second Source Select", "SE3", "SE3L"}, + {"Second Source Select", "SE3", "SE3R"}, + {"Second Source Select", "SE2", "SE2L"}, + {"Second Source Select", "SE2", "SE2R"}, + {"Second Source Select", "SE1", "SE1L"}, + {"Second Source Select", "SE1", "SE1R"}, + {"Second Source Select", "SE", "DIFFL"}, + {"Second Source Select", "SE", "DIFFR"}, + {"Second Source Select", "QD", "DIFFL"}, + {"Second Source Select", "QD", "DIFFR"}, + + {"Rear Speaker Source", "Main", "Main Source Select"}, + {"Rear Speaker Source", "Second", "Second Source Select"}, + + {"Subwoofer Enable", "Switch", "Main Source Select"}, + + {"Mix Enable", "Switch", "MIX"}, + + {"LF Output Mixer", NULL, "Main Source Select"}, + {"LF Output Mixer", "Mix to LF Speaker Switch", "Mix Enable"}, + {"RF Output Mixer", NULL, "Main Source Select"}, + {"RF Output Mixer", "Mix to RF Speaker Switch", "Mix Enable"}, + + {"OUTLF", NULL, "LF Output Mixer"}, + {"OUTRF", NULL, "RF Output Mixer"}, + {"OUTLR", NULL, "Rear Speaker Source"}, + {"OUTRR", NULL, "Rear Speaker Source"}, + {"OUTSW", NULL, "Subwoofer Enable"}, +}; + +static const struct snd_soc_component_driver tda7419_component_driver = { + .name = "tda7419", + .controls = tda7419_controls, + .num_controls = ARRAY_SIZE(tda7419_controls), + .dapm_widgets = tda7419_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tda7419_dapm_widgets), + .dapm_routes = tda7419_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tda7419_dapm_routes), +}; + +static int tda7419_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tda7419_data *tda7419; + int i, ret; + + tda7419 = devm_kzalloc(&i2c->dev, + sizeof(struct tda7419_data), + GFP_KERNEL); + if (tda7419 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, tda7419); + + tda7419->regmap = devm_regmap_init_i2c(i2c, &tda7419_regmap_config); + if (IS_ERR(tda7419->regmap)) { + ret = PTR_ERR(tda7419->regmap); + dev_err(&i2c->dev, "error initializing regmap: %d\n", + ret); + return ret; + } + + /* + * Reset registers to power-on defaults. The part does not provide a + * soft-reset function and the registers are not readable. This ensures + * that the cache matches register contents even if the registers have + * been previously initialized and not power cycled before probe. + */ + for (i = 0; i < ARRAY_SIZE(tda7419_regmap_defaults); i++) + regmap_write(tda7419->regmap, + tda7419_regmap_defaults[i].reg, + tda7419_regmap_defaults[i].def); + + ret = devm_snd_soc_register_component(&i2c->dev, + &tda7419_component_driver, NULL, 0); + if (ret < 0) { + dev_err(&i2c->dev, "error registering component: %d\n", + ret); + } + + return ret; +} + +static const struct i2c_device_id tda7419_i2c_id[] = { + { "tda7419", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tda7419_i2c_id); + +static const struct of_device_id tda7419_of_match[] = { + { .compatible = "st,tda7419" }, + { }, +}; + +static struct i2c_driver tda7419_driver = { + .driver = { + .name = "tda7419", + .of_match_table = tda7419_of_match, + }, + .probe = tda7419_probe, + .id_table = tda7419_i2c_id, +}; + +module_i2c_driver(tda7419_driver); + +MODULE_AUTHOR("Matt Porter "); +MODULE_DESCRIPTION("TDA7419 audio processor driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3