diff options
Diffstat (limited to 'sound/soc/codecs')
43 files changed, 2777 insertions, 195 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..4c7542571484 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT select SND_SOC_CS42L73 if I2C + select SND_SOC_CS4265 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CS42XX8_I2C if I2C @@ -74,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C @@ -225,11 +227,11 @@ config SND_SOC_ADAU1373 config SND_SOC_ADAU1701 tristate "Analog Devices ADAU1701 CODEC" depends on I2C - select SND_SOC_SIGMADSP + select SND_SOC_SIGMADSP_I2C config SND_SOC_ADAU17X1 tristate - select SND_SOC_SIGMADSP + select SND_SOC_SIGMADSP_REGMAP config SND_SOC_ADAU1761 tristate @@ -338,6 +340,11 @@ config SND_SOC_CS42L73 tristate "Cirrus Logic CS42L73 CODEC" depends on I2C +config SND_SOC_CS4265 + tristate "Cirrus Logic CS4265 CODEC" + depends on I2C + select REGMAP_I2C + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate "Cirrus Logic CS4270 CODEC" @@ -449,6 +456,9 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate @@ -476,6 +486,14 @@ config SND_SOC_SIGMADSP tristate select CRC32 +config SND_SOC_SIGMADSP_I2C + tristate + select SND_SOC_SIGMADSP + +config SND_SOC_SIGMADSP_REGMAP + tristate + select SND_SOC_SIGMADSP + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b8d73f..ade412e49bd0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -37,6 +37,7 @@ snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l56-objs := cs42l56.o snd-soc-cs42l73-objs := cs42l73.o +snd-soc-cs4265-objs := cs4265.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cs42xx8-objs := cs42xx8.o @@ -68,6 +69,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -77,6 +79,8 @@ snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o +snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o +snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-sn95031-objs := sn95031.o @@ -202,6 +206,7 @@ obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L56) += snd-soc-cs42l56.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o +obj-$(CONFIG_SND_SOC_CS4265) += snd-soc-cs4265.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o @@ -233,6 +238,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o @@ -240,6 +246,8 @@ obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o +obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o +obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 8d9ba4ba4bfe..e889e1b84192 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -89,8 +89,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) int ret; /* add codec as bus device for standard ac97 */ - ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL, - &ac97_bus); + ret = snd_ac97_bus(codec->component.card->snd_card, 0, soc_ac97_ops, + NULL, &ac97_bus); if (ret < 0) return ret; diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d71c59cf7bdd..370b742117ef 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -230,8 +230,10 @@ static int adau1701_reg_read(void *context, unsigned int reg, *value = 0; - for (i = 0; i < size; i++) - *value |= recv_buf[i] << (i * 8); + for (i = 0; i < size; i++) { + *value <<= 8; + *value |= recv_buf[i]; + } return 0; } diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index fd55da7cb9d4..70ab35744aba 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -968,7 +968,7 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, if (adau1977->dvdd_reg) power_off_mask = ~0; else - power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN; + power_off_mask = (unsigned int)~ADAU1977_BLOCK_POWER_SAI_LDO_EN; ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, power_off_mask, 0x00); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3ba4c0f11418..041712592e29 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -547,7 +547,7 @@ static const struct ak4642_drvdata ak4648_drvdata = { .extended_frequencies = 1, }; -static struct of_device_id ak4642_of_match[]; +static const struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -593,7 +593,7 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } -static struct of_device_id ak4642_of_match[] = { +static const struct of_device_id ak4642_of_match[] = { { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 72e953b2cb41..8107a1cac876 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -14,12 +14,18 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +static const char * const supply_names[] = { + "va", "vd" +}; + struct ak5386_priv { int reset_gpio; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { @@ -32,7 +38,42 @@ static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { { "Capture", NULL, "AINR" }, }; +static int ak5386_soc_probe(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static int ak5386_soc_remove(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +#ifdef CONFIG_PM +static int ak5386_soc_suspend(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +static int ak5386_soc_resume(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} +#else +#define ak5386_soc_suspend NULL +#define ak5386_soc_resume NULL +#endif /* CONFIG_PM */ + static struct snd_soc_codec_driver soc_codec_ak5386 = { + .probe = ak5386_soc_probe, + .remove = ak5386_soc_remove, + .suspend = ak5386_soc_suspend, + .resume = ak5386_soc_resume, .dapm_widgets = ak5386_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), .dapm_routes = ak5386_dapm_routes, @@ -122,6 +163,7 @@ static int ak5386_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct ak5386_priv *priv; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,6 +172,14 @@ static int ak5386_probe(struct platform_device *pdev) priv->reset_gpio = -EINVAL; dev_set_drvdata(dev, priv); + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) priv->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpio", 0); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f57d4c..2f2e91ac690f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +static const struct snd_soc_dapm_route arizona_mono_routes[] = { + { "OUT1R", NULL, "OUT1L" }, + { "OUT2R", NULL, "OUT2L" }, + { "OUT3R", NULL, "OUT3L" }, + { "OUT4R", NULL, "OUT4L" }, + { "OUT5R", NULL, "OUT5L" }, + { "OUT6R", NULL, "OUT6L" }, +}; + +int arizona_init_mono(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) { + if (arizona->pdata.out_mono[i]) + snd_soc_dapm_add_routes(&codec->dapm, + &arizona_mono_routes[i], 1); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_mono); + int arizona_init_gpio(struct snd_soc_codec *codec) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); @@ -1127,6 +1152,31 @@ static int arizona_startup(struct snd_pcm_substream *substream, constraint); } +static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, + unsigned int rate) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + struct reg_default dac_comp[] = { + { 0x80, 0x3 }, + { ARIZONA_DAC_COMP_1, 0 }, + { ARIZONA_DAC_COMP_2, 0 }, + { 0x80, 0x0 }, + }; + + mutex_lock(&codec->mutex); + + dac_comp[1].def = arizona->dac_comp_coeff; + if (rate >= 176400) + dac_comp[2].def = arizona->dac_comp_enabled; + + mutex_unlock(&codec->mutex); + + regmap_multi_reg_write(arizona->regmap, + dac_comp, + ARRAY_SIZE(dac_comp)); +} + static int arizona_hw_params_rate(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1153,6 +1203,15 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, switch (dai_priv->clk) { case ARIZONA_CLK_SYSCLK: + switch (priv->arizona->type) { + case WM5102: + arizona_wm5102_set_dac_comp(codec, + params_rate(params)); + break; + default: + break; + } + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, ARIZONA_SAMPLE_RATE_1_MASK, sr_val); if (base) @@ -1175,6 +1234,27 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, return 0; } +static bool arizona_aif_cfg_changed(struct snd_soc_codec *codec, + int base, int bclk, int lrclk, int frame) +{ + int val; + + val = snd_soc_read(codec, base + ARIZONA_AIF_BCLK_CTRL); + if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK)) + return true; + + val = snd_soc_read(codec, base + ARIZONA_AIF_TX_BCLK_RATE); + if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK)) + return true; + + val = snd_soc_read(codec, base + ARIZONA_AIF_FRAME_CTRL_1); + if (frame != (val & (ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK))) + return true; + + return false; +} + static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1185,26 +1265,40 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int base = dai->driver->base; const int *rates; int i, ret, val; + int channels = params_channels(params); int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int tdm_width = arizona->tdm_width[dai->id - 1]; + int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; + bool reconfig; + unsigned int aif_tx_state, aif_rx_state; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; else rates = &arizona_48k_bclk_rates[0]; - bclk_target = snd_soc_params_to_bclk(params); - if (chan_limit && chan_limit < params_channels(params)) { + if (tdm_slots) { + arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", + tdm_slots, tdm_width); + bclk_target = tdm_slots * tdm_width * params_rate(params); + channels = tdm_slots; + } else { + bclk_target = snd_soc_params_to_bclk(params); + } + + if (chan_limit && chan_limit < channels) { arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); - bclk_target /= params_channels(params); + bclk_target /= channels; bclk_target *= chan_limit; } - /* Force stereo for I2S mode */ + /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); - bclk_target *= 2; + bclk_target /= channels; + bclk_target *= channels + 1; } for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { @@ -1228,28 +1322,56 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); + + if (reconfig) { + /* Save AIF TX/RX state */ + aif_tx_state = snd_soc_read(codec, + base + ARIZONA_AIF_TX_ENABLES); + aif_rx_state = snd_soc_read(codec, + base + ARIZONA_AIF_RX_ENABLES); + /* Disable AIF TX/RX before reconfiguring it */ + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_ENABLES, 0xff, 0x0); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0); + } + ret = arizona_hw_params_rate(substream, params, dai); if (ret != 0) - return ret; + goto restore_aif; - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_TX_BCLK_RATE, - ARIZONA_AIF1TX_BCPF_MASK, lrclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_RX_BCLK_RATE, - ARIZONA_AIF1RX_BCPF_MASK, lrclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_FRAME_CTRL_1, - ARIZONA_AIF1TX_WL_MASK | - ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); - regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2, - ARIZONA_AIF1RX_WL_MASK | - ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + if (reconfig) { + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + } - return 0; +restore_aif: + if (reconfig) { + /* Restore AIF TX/RX state */ + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_ENABLES, + 0xff, aif_tx_state); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_RX_ENABLES, + 0xff, aif_rx_state); + } + return ret; } static const char *arizona_dai_clk_str(int clk_id) @@ -1324,9 +1446,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) ARIZONA_AIF1_TRI, reg); } +static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, + unsigned int base, + int channels, unsigned int mask) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int slot, i; + + for (i = 0; i < channels; ++i) { + slot = ffs(mask) - 1; + if (slot < 0) + return; + + regmap_write(arizona->regmap, base + i, slot); + + mask &= ~(1 << slot); + } + + if (mask) + arizona_aif_warn(dai, "Too many channels in TDM mask\n"); +} + +static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int base = dai->driver->base; + int rx_max_chan = dai->driver->playback.channels_max; + int tx_max_chan = dai->driver->capture.channels_max; + + /* Only support TDM for the physical AIFs */ + if (dai->id > ARIZONA_MAX_AIF) + return -ENOTSUPP; + + if (slots == 0) { + tx_mask = (1 << tx_max_chan) - 1; + rx_mask = (1 << rx_max_chan) - 1; + } + + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3, + tx_max_chan, tx_mask); + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11, + rx_max_chan, rx_mask); + + arizona->tdm_width[dai->id - 1] = slot_width; + arizona->tdm_slots[dai->id - 1] = slots; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, + .set_tdm_slot = arizona_set_tdm_slot, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, .set_tristate = arizona_set_tristate, @@ -1400,6 +1576,12 @@ static int arizona_validate_fll(struct arizona_fll *fll, { unsigned int Fvco_min; + if (fll->fout && Fout != fll->fout) { + arizona_fll_err(fll, + "Can't change output on active FLL\n"); + return -EINVAL; + } + if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) { arizona_fll_err(fll, "Can't scale %dMHz in to <=13.5MHz\n", @@ -1478,6 +1660,10 @@ static int arizona_calc_fratio(struct arizona_fll *fll, while (div <= ARIZONA_FLL_MAX_REFDIV) { for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { + if ((ARIZONA_FLL_VCO_CORNER / 2) / + (fll->vco_mult * ratio) < Fref) + break; + if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; @@ -1485,11 +1671,7 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } } - for (ratio = init_ratio - 1; ratio >= 0; ratio--) { - if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < - Fref) - break; - + for (ratio = init_ratio - 1; ratio > 0; ratio--) { if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; @@ -1616,7 +1798,7 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, ARIZONA_FLL1_CTRL_UPD | cfg->n); } -static bool arizona_is_enabled_fll(struct arizona_fll *fll) +static int arizona_is_enabled_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; unsigned int reg; @@ -1632,13 +1814,26 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll) return reg & ARIZONA_FLL1_ENA; } -static void arizona_enable_fll(struct arizona_fll *fll) +static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; int ret; bool use_sync = false; + int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; + if (already_enabled < 0) + return already_enabled; + + if (already_enabled) { + /* Facilitate smooth refclk across the transition */ + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7, + ARIZONA_FLL1_GAIN_MASK, 0); + regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, + ARIZONA_FLL1_FREERUN); + } + /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. @@ -1666,7 +1861,7 @@ static void arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_ENA, 0); } else { arizona_fll_err(fll, "No clocks provided\n"); - return; + return -EINVAL; } /* @@ -1681,25 +1876,29 @@ static void arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); - if (!arizona_is_enabled_fll(fll)) + if (!already_enabled) pm_runtime_get(arizona->dev); /* Clear any pending completions */ try_wait_for_completion(&fll->ok); regmap_update_bits_async(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, 0); - regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) regmap_update_bits_async(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); + if (already_enabled) + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + ret = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); if (ret == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; } static void arizona_disable_fll(struct arizona_fll *fll) @@ -1713,6 +1912,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); if (change) pm_runtime_put_autosuspend(arizona->dev); @@ -1721,7 +1922,7 @@ static void arizona_disable_fll(struct arizona_fll *fll) int arizona_set_fll_refclk(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - int ret; + int ret = 0; if (fll->ref_src == source && fll->ref_freq == Fref) return 0; @@ -1736,17 +1937,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_freq = Fref; if (fll->fout && Fref > 0) { - arizona_enable_fll(fll); + ret = arizona_enable_fll(fll); } - return 0; + return ret; } EXPORT_SYMBOL_GPL(arizona_set_fll_refclk); int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - int ret; + int ret = 0; if (fll->sync_src == source && fll->sync_freq == Fref && fll->fout == Fout) @@ -1768,13 +1969,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->sync_freq = Fref; fll->fout = Fout; - if (Fout) { - arizona_enable_fll(fll); - } else { + if (Fout) + ret = arizona_enable_fll(fll); + else arizona_disable_fll(fll); - } - return 0; + return ret; } EXPORT_SYMBOL_GPL(arizona_set_fll); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 05ae17f5bca3..942cfb197b6d 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); +extern int arizona_init_mono(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c new file mode 100644 index 000000000000..a20b30ca52c0 --- /dev/null +++ b/sound/soc/codecs/cs4265.c @@ -0,0 +1,682 @@ +/* + * cs4265.c -- CS4265 ALSA SoC audio driver + * + * Copyright 2014 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan@cirrus.com> + * + * 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. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/gpio/consumer.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4265.h" + +struct cs4265_private { + struct device *dev; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 format; + u32 sysclk; +}; + +static const struct reg_default cs4265_reg_defaults[] = { + { CS4265_PWRCTL, 0x0F }, + { CS4265_DAC_CTL, 0x08 }, + { CS4265_ADC_CTL, 0x00 }, + { CS4265_MCLK_FREQ, 0x00 }, + { CS4265_SIG_SEL, 0x40 }, + { CS4265_CHB_PGA_CTL, 0x00 }, + { CS4265_CHA_PGA_CTL, 0x00 }, + { CS4265_ADC_CTL2, 0x19 }, + { CS4265_DAC_CHA_VOL, 0x00 }, + { CS4265_DAC_CHB_VOL, 0x00 }, + { CS4265_DAC_CTL2, 0xC0 }, + { CS4265_SPDIF_CTL1, 0x00 }, + { CS4265_SPDIF_CTL2, 0x00 }, + { CS4265_INT_MASK, 0x00 }, + { CS4265_STATUS_MODE_MSB, 0x00 }, + { CS4265_STATUS_MODE_LSB, 0x00 }, +}; + +static bool cs4265_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4265_PWRCTL: + case CS4265_DAC_CTL: + case CS4265_ADC_CTL: + case CS4265_MCLK_FREQ: + case CS4265_SIG_SEL: + case CS4265_CHB_PGA_CTL: + case CS4265_CHA_PGA_CTL: + case CS4265_ADC_CTL2: + case CS4265_DAC_CHA_VOL: + case CS4265_DAC_CHB_VOL: + case CS4265_DAC_CTL2: + case CS4265_SPDIF_CTL1: + case CS4265_SPDIF_CTL2: + case CS4265_INT_MASK: + case CS4265_STATUS_MODE_MSB: + case CS4265_STATUS_MODE_LSB: + return true; + default: + return false; + } +} + +static bool cs4265_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4265_INT_STATUS: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(pga_tlv, -1200, 50, 0); + +static DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 0); + +static const char * const digital_input_mux_text[] = { + "SDIN1", "SDIN2" +}; + +static SOC_ENUM_SINGLE_DECL(digital_input_mux_enum, CS4265_SIG_SEL, 7, + digital_input_mux_text); + +static const struct snd_kcontrol_new digital_input_mux = + SOC_DAPM_ENUM("Digital Input Mux", digital_input_mux_enum); + +static const char * const mic_linein_text[] = { + "MIC", "LINEIN" +}; + +static SOC_ENUM_SINGLE_DECL(mic_linein_enum, CS4265_ADC_CTL2, 0, + mic_linein_text); + +static const char * const cam_mode_text[] = { + "One Byte", "Two Byte" +}; + +static SOC_ENUM_SINGLE_DECL(cam_mode_enum, CS4265_SPDIF_CTL1, 5, + cam_mode_text); + +static const char * const cam_mono_stereo_text[] = { + "Stereo", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(spdif_mono_stereo_enum, CS4265_SPDIF_CTL2, 2, + cam_mono_stereo_text); + +static const char * const mono_select_text[] = { + "Channel A", "Channel B" +}; + +static SOC_ENUM_SINGLE_DECL(spdif_mono_select_enum, CS4265_SPDIF_CTL2, 0, + mono_select_text); + +static const struct snd_kcontrol_new mic_linein_mux = + SOC_DAPM_ENUM("ADC Input Capture Mux", mic_linein_enum); + +static const struct snd_kcontrol_new loopback_ctl = + SOC_DAPM_SINGLE("Switch", CS4265_SIG_SEL, 1, 1, 0); + +static const struct snd_kcontrol_new spdif_switch = + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 0, 0); + +static const struct snd_kcontrol_new dac_switch = + SOC_DAPM_SINGLE("Switch", CS4265_PWRCTL, 1, 1, 0); + +static const struct snd_kcontrol_new cs4265_snd_controls[] = { + + SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL, + CS4265_CHB_PGA_CTL, 0, 0x28, 0x30, pga_tlv), + SOC_DOUBLE_R_TLV("DAC Volume", CS4265_DAC_CHA_VOL, + CS4265_DAC_CHB_VOL, 0, 0xFF, 1, dac_tlv), + SOC_SINGLE("De-emp 44.1kHz Switch", CS4265_DAC_CTL, 1, + 1, 0), + SOC_SINGLE("DAC INV Switch", CS4265_DAC_CTL2, 5, + 1, 0), + SOC_SINGLE("DAC Zero Cross Switch", CS4265_DAC_CTL2, 6, + 1, 0), + SOC_SINGLE("DAC Soft Ramp Switch", CS4265_DAC_CTL2, 7, + 1, 0), + SOC_SINGLE("ADC HPF Switch", CS4265_ADC_CTL, 1, + 1, 0), + SOC_SINGLE("ADC Zero Cross Switch", CS4265_ADC_CTL2, 3, + 1, 1), + SOC_SINGLE("ADC Soft Ramp Switch", CS4265_ADC_CTL2, 7, + 1, 0), + SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, + 6, 1, 0), + SOC_ENUM("C Data Access", cam_mode_enum), + SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, + 3, 1, 0), + SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), + SOC_SINGLE("MMTLR Data Switch", 0, + 1, 1, 0), + SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), + SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), +}; + +static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + SND_SOC_DAPM_AIF_OUT("DOUT", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SPDIFOUT", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADC Mux", SND_SOC_NOPM, 0, 0, &mic_linein_mux), + + SND_SOC_DAPM_ADC("ADC", NULL, CS4265_PWRCTL, 2, 1), + SND_SOC_DAPM_PGA("Pre-amp MIC", CS4265_PWRCTL, 3, + 1, NULL, 0), + + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, + 0, 0, &digital_input_mux), + + SND_SOC_DAPM_MIXER("SDIN1 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SDIN2 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SPDIF Transmitter", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0, + &loopback_ctl), + SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0, + &spdif_switch), + SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1, + &dac_switch), + + SND_SOC_DAPM_AIF_IN("DIN1", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DIN2", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("TXIN", NULL, 0, + CS4265_SPDIF_CTL2, 5, 1), + + SND_SOC_DAPM_OUTPUT("LINEOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTR"), + +}; + +static const struct snd_soc_dapm_route cs4265_audio_map[] = { + + {"DIN1", NULL, "DAI1 Playback"}, + {"DIN2", NULL, "DAI2 Playback"}, + {"SDIN1 Input Mixer", NULL, "DIN1"}, + {"SDIN2 Input Mixer", NULL, "DIN2"}, + {"Input Mux", "SDIN1", "SDIN1 Input Mixer"}, + {"Input Mux", "SDIN2", "SDIN2 Input Mixer"}, + {"DAC", "Switch", "Input Mux"}, + {"SPDIF", "Switch", "Input Mux"}, + {"LINEOUTL", NULL, "DAC"}, + {"LINEOUTR", NULL, "DAC"}, + {"SPDIFOUT", NULL, "SPDIF"}, + + {"ADC Mux", "LINEIN", "LINEINL"}, + {"ADC Mux", "LINEIN", "LINEINR"}, + {"ADC Mux", "MIC", "MICL"}, + {"ADC Mux", "MIC", "MICR"}, + {"ADC", NULL, "ADC Mux"}, + {"DOUT", NULL, "ADC"}, + {"DAI1 Capture", NULL, "DOUT"}, + {"DAI2 Capture", NULL, "DOUT"}, + + /* Loopback */ + {"Loopback", "Switch", "ADC"}, + {"DAC", NULL, "Loopback"}, +}; + +struct cs4265_clk_para { + u32 mclk; + u32 rate; + u8 fm_mode; /* values 1, 2, or 4 */ + u8 mclkdiv; +}; + +static const struct cs4265_clk_para clk_map_table[] = { + /*32k*/ + {8192000, 32000, 0, 0}, + {12288000, 32000, 0, 1}, + {16384000, 32000, 0, 2}, + {24576000, 32000, 0, 3}, + {32768000, 32000, 0, 4}, + + /*44.1k*/ + {11289600, 44100, 0, 0}, + {16934400, 44100, 0, 1}, + {22579200, 44100, 0, 2}, + {33868000, 44100, 0, 3}, + {45158400, 44100, 0, 4}, + + /*48k*/ + {12288000, 48000, 0, 0}, + {18432000, 48000, 0, 1}, + {24576000, 48000, 0, 2}, + {36864000, 48000, 0, 3}, + {49152000, 48000, 0, 4}, + + /*64k*/ + {8192000, 64000, 1, 0}, + {1228800, 64000, 1, 1}, + {1693440, 64000, 1, 2}, + {2457600, 64000, 1, 3}, + {3276800, 64000, 1, 4}, + + /* 88.2k */ + {11289600, 88200, 1, 0}, + {16934400, 88200, 1, 1}, + {22579200, 88200, 1, 2}, + {33868000, 88200, 1, 3}, + {45158400, 88200, 1, 4}, + + /* 96k */ + {12288000, 96000, 1, 0}, + {18432000, 96000, 1, 1}, + {24576000, 96000, 1, 2}, + {36864000, 96000, 1, 3}, + {49152000, 96000, 1, 4}, + + /* 128k */ + {8192000, 128000, 2, 0}, + {12288000, 128000, 2, 1}, + {16934400, 128000, 2, 2}, + {24576000, 128000, 2, 3}, + {32768000, 128000, 2, 4}, + + /* 176.4k */ + {11289600, 176400, 2, 0}, + {16934400, 176400, 2, 1}, + {22579200, 176400, 2, 2}, + {33868000, 176400, 2, 3}, + {49152000, 176400, 2, 4}, + + /* 192k */ + {12288000, 192000, 2, 0}, + {18432000, 192000, 2, 1}, + {24576000, 192000, 2, 2}, + {36864000, 192000, 2, 3}, + {49152000, 192000, 2, 4}, +}; + +static int cs4265_get_clk_index(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].rate == rate && + clk_map_table[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int cs4265_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 cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + int i; + + if (clk_id != 0) { + dev_err(codec->dev, "Invalid clk_id %d\n", clk_id); + return -EINVAL; + } + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].mclk == freq) { + cs4265->sysclk = freq; + return 0; + } + } + cs4265->sysclk = 0; + dev_err(codec->dev, "Invalid freq parameter %d\n", freq); + return -EINVAL; +} + +static int cs4265_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_MASTER, + CS4265_ADC_MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_MASTER, + 0); + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= SND_SOC_DAIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= SND_SOC_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= SND_SOC_DAIFMT_LEFT_J; + break; + default: + return -EINVAL; + } + + cs4265->format = iface; + return 0; +} + +static int cs4265_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_MUTE, + CS4265_DAC_CTL_MUTE); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_MUTE, + CS4265_SPDIF_CTL2_MUTE); + } else { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_MUTE, + 0); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_MUTE, + 0); + } + return 0; +} + +static int cs4265_pcm_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 cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + int index; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + ((cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_RIGHT_J)) + return -EINVAL; + + index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params)); + if (index >= 0) { + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_FM, clk_map_table[index].fm_mode); + snd_soc_update_bits(codec, CS4265_MCLK_FREQ, + CS4265_MCLK_FREQ_MASK, + clk_map_table[index].mclkdiv); + + } else { + dev_err(codec->dev, "can't get correct mclk\n"); + return -EINVAL; + } + + switch (cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (1 << 4)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_DIF, (1 << 4)); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_DIF, (1 << 6)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + if (params_width(params) == 16) { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (1 << 5)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 7)); + } else { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (3 << 5)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 7)); + } + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, 0); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_DIF, 0); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 6)); + + break; + default: + return -EINVAL; + } + return 0; +} + +static int cs4265_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, 0); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, + CS4265_PWRCTL_PDN); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, + CS4265_PWRCTL_PDN); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define CS4265_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + +#define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + +static const struct snd_soc_dai_ops cs4265_ops = { + .hw_params = cs4265_pcm_hw_params, + .digital_mute = cs4265_digital_mute, + .set_fmt = cs4265_set_fmt, + .set_sysclk = cs4265_set_sysclk, +}; + +static struct snd_soc_dai_driver cs4265_dai[] = { + { + .name = "cs4265-dai1", + .playback = { + .stream_name = "DAI1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .capture = { + .stream_name = "DAI1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .ops = &cs4265_ops, + }, + { + .name = "cs4265-dai2", + .playback = { + .stream_name = "DAI2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .capture = { + .stream_name = "DAI2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .ops = &cs4265_ops, + }, +}; + +static const struct snd_soc_codec_driver soc_codec_cs4265 = { + .set_bias_level = cs4265_set_bias_level, + + .dapm_widgets = cs4265_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4265_dapm_widgets), + .dapm_routes = cs4265_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs4265_audio_map), + + .controls = cs4265_snd_controls, + .num_controls = ARRAY_SIZE(cs4265_snd_controls), +}; + +static const struct regmap_config cs4265_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4265_MAX_REGISTER, + .reg_defaults = cs4265_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4265_reg_defaults), + .readable_reg = cs4265_readable_register, + .volatile_reg = cs4265_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4265_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs4265_private *cs4265; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs4265 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4265_private), + GFP_KERNEL); + if (cs4265 == NULL) + return -ENOMEM; + cs4265->dev = &i2c_client->dev; + + cs4265->regmap = devm_regmap_init_i2c(i2c_client, &cs4265_regmap); + if (IS_ERR(cs4265->regmap)) { + ret = PTR_ERR(cs4265->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, + "reset-gpios"); + if (IS_ERR(cs4265->reset_gpio)) { + ret = PTR_ERR(cs4265->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + cs4265->reset_gpio = NULL; + } else { + ret = gpiod_direction_output(cs4265->reset_gpio, 0); + if (ret) + return ret; + mdelay(1); + gpiod_set_value_cansleep(cs4265->reset_gpio, 1); + + } + + i2c_set_clientdata(i2c_client, cs4265); + + ret = regmap_read(cs4265->regmap, CS4265_CHIP_ID, ®); + devid = reg & CS4265_CHIP_ID_MASK; + if (devid != CS4265_CHIP_ID_VAL) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS4265 Device ID (%X). Expected %X\n", + devid, CS4265_CHIP_ID); + return ret; + } + dev_info(&i2c_client->dev, + "CS4265 Version %x\n", + reg & CS4265_REV_ID_MASK); + + regmap_write(cs4265->regmap, CS4265_PWRCTL, 0x0F); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_cs4265, cs4265_dai, + ARRAY_SIZE(cs4265_dai)); + return ret; +} + +static int cs4265_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct of_device_id cs4265_of_match[] = { + { .compatible = "cirrus,cs4265", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4265_of_match); + +static const struct i2c_device_id cs4265_id[] = { + { "cs4265", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs4265_id); + +static struct i2c_driver cs4265_i2c_driver = { + .driver = { + .name = "cs4265", + .owner = THIS_MODULE, + .of_match_table = cs4265_of_match, + }, + .id_table = cs4265_id, + .probe = cs4265_i2c_probe, + .remove = cs4265_i2c_remove, +}; + +module_i2c_driver(cs4265_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS4265 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan@cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4265.h b/sound/soc/codecs/cs4265.h new file mode 100644 index 000000000000..0a80a8dcec67 --- /dev/null +++ b/sound/soc/codecs/cs4265.h @@ -0,0 +1,64 @@ +/* + * cs4265.h -- CS4265 ALSA SoC audio driver + * + * Copyright 2014 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan@cirrus.com> + * + * 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. + * + */ + +#ifndef __CS4265_H__ +#define __CS4265_H__ + +#define CS4265_CHIP_ID 0x1 +#define CS4265_CHIP_ID_VAL 0xD0 +#define CS4265_CHIP_ID_MASK 0xF0 +#define CS4265_REV_ID_MASK 0x0F + +#define CS4265_PWRCTL 0x02 +#define CS4265_PWRCTL_PDN 1 + +#define CS4265_DAC_CTL 0x3 +#define CS4265_DAC_CTL_MUTE (1 << 2) +#define CS4265_DAC_CTL_DIF (3 << 4) + +#define CS4265_ADC_CTL 0x4 +#define CS4265_ADC_MASTER 1 +#define CS4265_ADC_DIF (1 << 4) +#define CS4265_ADC_FM (3 << 6) + +#define CS4265_MCLK_FREQ 0x5 +#define CS4265_MCLK_FREQ_MASK (7 << 4) + +#define CS4265_SIG_SEL 0x6 +#define CS4265_SIG_SEL_LOOP (1 << 1) + +#define CS4265_CHB_PGA_CTL 0x7 +#define CS4265_CHA_PGA_CTL 0x8 + +#define CS4265_ADC_CTL2 0x9 + +#define CS4265_DAC_CHA_VOL 0xA +#define CS4265_DAC_CHB_VOL 0xB + +#define CS4265_DAC_CTL2 0xC + +#define CS4265_INT_STATUS 0xD +#define CS4265_INT_MASK 0xE +#define CS4265_STATUS_MODE_MSB 0xF +#define CS4265_STATUS_MODE_LSB 0x10 + +#define CS4265_SPDIF_CTL1 0x11 + +#define CS4265_SPDIF_CTL2 0x12 +#define CS4265_SPDIF_CTL2_MUTE (1 << 4) +#define CS4265_SPDIF_CTL2_DIF (3 << 6) + +#define CS4265_C_DATA_BUFF 0x13 +#define CS4265_MAX_REGISTER 0x2A + +#endif diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 071fc77f2f06..969167d8b71e 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, - CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv), SOC_ENUM("Headphone Analog Gain", hp_gain_enum), SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, - CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), + CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), @@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("MIC Bias Level", mic_bias_level_enum), SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, - CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 6, 0x7f, 0x19, ipd_tlv), + 0, 0x19, 0x7F, ipd_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), @@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_MIXER_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, - CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv), SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, - 0, 0x7f, 0x19, mix_tlv), + 0, 0x19, 0x7f, mix_tlv), SOC_DOUBLE_R("PCM Mixer Switch", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index fdc4bd27b0df..c766a5a9ce80 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -318,24 +318,32 @@ static const struct soc_enum adca_swap_enum = ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); +static const struct snd_kcontrol_new adca_swap_mux = + SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 4, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); +static const struct snd_kcontrol_new pcma_swap_mux = + SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 2, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); +static const struct snd_kcontrol_new adcb_swap_mux = + SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 6, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); +static const struct snd_kcontrol_new pcmb_swap_mux = + SOC_DAPM_ENUM("Route", pcmb_swap_enum); static const struct snd_kcontrol_new hpa_switch = SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 6, 1, 1); @@ -421,15 +429,15 @@ static const struct soc_enum ng_delay_enum = static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME, - CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv), + CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv), SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME, - CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), + CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME, - CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), + CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1), SOC_SINGLE_TLV("Analog Advisory Volume", @@ -438,16 +446,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME, - CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv), + CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv), SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR, CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv), SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1), SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), @@ -467,11 +475,6 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_SINGLE("ADCA Invert", CS42L56_MISC_ADC_CTL, 2, 1, 1), SOC_SINGLE("ADCB Invert", CS42L56_MISC_ADC_CTL, 3, 1, 1), - SOC_ENUM("PCMA Swap", pcma_swap_enum), - SOC_ENUM("PCMB Swap", pcmb_swap_enum), - SOC_ENUM("ADCA Swap", adca_swap_enum), - SOC_ENUM("ADCB Swap", adcb_swap_enum), - SOC_DOUBLE("HPF Switch", CS42L56_HPF_CTL, 5, 7, 1, 1), SOC_DOUBLE("HPF Freeze Switch", CS42L56_HPF_CTL, 4, 6, 1, 1), SOC_ENUM("HPFA Corner Freq", hpfa_freq_enum), @@ -570,6 +573,16 @@ static const struct snd_soc_dapm_widget cs42l56_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCA", NULL, CS42L56_PWRCTL_1, 1, 1), SND_SOC_DAPM_ADC("ADCB", NULL, CS42L56_PWRCTL_1, 2, 1), + SND_SOC_DAPM_MUX("ADCA Swap Mux", SND_SOC_NOPM, 0, 0, + &adca_swap_mux), + SND_SOC_DAPM_MUX("ADCB Swap Mux", SND_SOC_NOPM, 0, 0, + &adcb_swap_mux), + + SND_SOC_DAPM_MUX("PCMA Swap Mux", SND_SOC_NOPM, 0, 0, + &pcma_swap_mux), + SND_SOC_DAPM_MUX("PCMB Swap Mux", SND_SOC_NOPM, 0, 0, + &pcmb_swap_mux), + SND_SOC_DAPM_DAC("DACA", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACB", NULL, SND_SOC_NOPM, 0, 0), @@ -607,8 +620,19 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = { {"Digital Output Mux", NULL, "ADCA"}, {"Digital Output Mux", NULL, "ADCB"}, - {"ADCB", NULL, "ADCB Mux"}, - {"ADCA", NULL, "ADCA Mux"}, + {"ADCB", NULL, "ADCB Swap Mux"}, + {"ADCA", NULL, "ADCA Swap Mux"}, + + {"ADCA Swap Mux", NULL, "ADCA"}, + {"ADCB Swap Mux", NULL, "ADCB"}, + + {"DACA", "Left", "ADCA Swap Mux"}, + {"DACA", "LR 2", "ADCA Swap Mux"}, + {"DACA", "Right", "ADCA Swap Mux"}, + + {"DACB", "Left", "ADCB Swap Mux"}, + {"DACB", "LR 2", "ADCB Swap Mux"}, + {"DACB", "Right", "ADCB Swap Mux"}, {"ADCA Mux", NULL, "AIN3A"}, {"ADCA Mux", NULL, "AIN2A"}, @@ -633,30 +657,32 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = { {"PGAB Input Mux", NULL, "AIN2B"}, {"PGAB Input Mux", NULL, "AIN3B"}, - {"LOB", NULL, "Lineout Right"}, - {"LOA", NULL, "Lineout Left"}, - - {"Lineout Right", "Switch", "LINEOUTB Input Mux"}, - {"Lineout Left", "Switch", "LINEOUTA Input Mux"}, + {"LOB", "Switch", "LINEOUTB Input Mux"}, + {"LOA", "Switch", "LINEOUTA Input Mux"}, {"LINEOUTA Input Mux", "PGAA", "PGAA"}, {"LINEOUTB Input Mux", "PGAB", "PGAB"}, {"LINEOUTA Input Mux", "DACA", "DACA"}, {"LINEOUTB Input Mux", "DACB", "DACB"}, - {"HPA", NULL, "Headphone Left"}, - {"HPB", NULL, "Headphone Right"}, - - {"Headphone Right", "Switch", "HPB Input Mux"}, - {"Headphone Left", "Switch", "HPA Input Mux"}, + {"HPA", "Switch", "HPB Input Mux"}, + {"HPB", "Switch", "HPA Input Mux"}, {"HPA Input Mux", "PGAA", "PGAA"}, {"HPB Input Mux", "PGAB", "PGAB"}, {"HPA Input Mux", "DACA", "DACA"}, {"HPB Input Mux", "DACB", "DACB"}, - {"DACB", NULL, "HiFi Playback"}, - {"DACA", NULL, "HiFi Playback"}, + {"DACA", NULL, "PCMA Swap Mux"}, + {"DACB", NULL, "PCMB Swap Mux"}, + + {"PCMB Swap Mux", "Left", "HiFi Playback"}, + {"PCMB Swap Mux", "LR 2", "HiFi Playback"}, + {"PCMB Swap Mux", "Right", "HiFi Playback"}, + + {"PCMA Swap Mux", "Left", "HiFi Playback"}, + {"PCMA Swap Mux", "LR 2", "HiFi Playback"}, + {"PCMA Swap Mux", "Right", "HiFi Playback"}, }; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index ae3717992d56..8658194f50bf 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0x34, + CS42L73_MICBPREPGABVOL, 0, 0x34, 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index a25bc6061a30..02b1520ae0bc 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -219,6 +219,9 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_RIGHT_J: val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; break; + case SND_SOC_DAIFMT_DSP_A: + val = CS42XX8_INTF_DAC_DIF_TDM | CS42XX8_INTF_ADC_DIF_TDM; + break; default: dev_err(codec->dev, "unsupported dai format\n"); return -EINVAL; @@ -422,7 +425,7 @@ const struct cs42xx8_driver_data cs42888_data = { }; EXPORT_SYMBOL_GPL(cs42888_data); -const struct of_device_id cs42xx8_of_match[] = { +static const struct of_device_id cs42xx8_of_match[] = { { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, { /* sentinel */ } diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h index da0b94aee419..b2c10e537ef6 100644 --- a/sound/soc/codecs/cs42xx8.h +++ b/sound/soc/codecs/cs42xx8.h @@ -128,8 +128,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap); #define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT) -#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT) -#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (5 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_TDM (6 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_SHIFT 0 #define CS42XX8_INTF_ADC_DIF_WIDTH 3 #define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT) @@ -138,8 +138,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap); #define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT) -#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT) -#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (5 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_TDM (6 << CS42XX8_INTF_ADC_DIF_SHIFT) /* ADC Control & DAC De-Emphasis (Address 05h) */ #define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d5fd00a64748..8f95b0300f1a 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -253,7 +253,7 @@ static void v253_close(struct tty_struct *tty) /* Prevent the codec driver from further accessing the modem */ codec->hw_write = NULL; cx20442->control_data = NULL; - codec->card->pop_time = 0; + codec->component.card->pop_time = 0; } /* Line discipline .hangup() */ @@ -281,7 +281,7 @@ static void v253_receive(struct tty_struct *tty, /* Set up codec driver access to modem controls */ cx20442->control_data = tty; codec->hw_write = (hw_write_t)tty->ops->write; - codec->card->pop_time = 1; + codec->component.card->pop_time = 1; } } @@ -372,7 +372,7 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, cx20442); codec->hw_write = NULL; - codec->card->pop_time = 0; + codec->component.card->pop_time = 0; return 0; } @@ -383,8 +383,8 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); if (cx20442->control_data) { - struct tty_struct *tty = cx20442->control_data; - tty_hangup(tty); + struct tty_struct *tty = cx20442->control_data; + tty_hangup(tty); } if (!IS_ERR(cx20442->por)) { diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6816578ea8f7..4a063fa88526 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2266,7 +2266,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Register for interrupts */ dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - ret = request_threaded_irq(max98090->irq, NULL, + ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 3a80ba4452df..57b0c94a710b 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -36,6 +36,7 @@ #define PCM1792A_DAC_VOL_LEFT 0x10 #define PCM1792A_DAC_VOL_RIGHT 0x11 #define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_MODE_CONTROL 0x13 #define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL #define PCM1792A_FMT_MASK 0x70 @@ -164,6 +165,8 @@ static const struct snd_kcontrol_new pcm1792a_controls[] = { SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, pcm1792a_dac_tlv), + SOC_SINGLE("DAC Invert Output Switch", PCM1792A_MODE_CONTROL, 7, 1, 0), + SOC_SINGLE("DAC Rolloff Filter Switch", PCM1792A_MODE_CONTROL, 1, 1, 0), }; static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = { diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h index 7a83d1fc102a..51d5470fee16 100644 --- a/sound/soc/codecs/pcm1792a.h +++ b/sound/soc/codecs/pcm1792a.h @@ -18,7 +18,8 @@ #define __PCM1792A_H__ #define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) #define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S16_LE) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 000000000000..218f86efd196 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1224 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao@realtek.com> + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/rt286.h> +#include <sound/hda_verbs.h> + +#include "rt286.h" + +#define RT286_VENDOR_ID 0x10ec0286 + +struct rt286_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + struct reg_default *index_cache; +}; + +static struct reg_default rt286_index_def[] = { + { 0x01, 0xaaaa }, + { 0x02, 0x8aaa }, + { 0x03, 0x0002 }, + { 0x04, 0xaf01 }, + { 0x08, 0x000d }, + { 0x09, 0xd810 }, + { 0x0a, 0x0060 }, + { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, + { 0x0f, 0x0000 }, + { 0x19, 0x0a17 }, + { 0x20, 0x0020 }, + { 0x33, 0x0208 }, + { 0x49, 0x0004 }, + { 0x4f, 0x50e9 }, + { 0x50, 0x2c00 }, + { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) + +static const struct reg_default rt286_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x0000007f }, + { 0x0023a000, 0x0000007f }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x000000c3 }, + { 0x00936000, 0x000000c3 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000097 }, + { 0x00b37200, 0x00000097 }, + { 0x00b37300, 0x00000097 }, + { 0x00c37000, 0x00000000 }, + { 0x00c37100, 0x00000080 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01439000, 0x00000080 }, + { 0x0143a000, 0x00000080 }, + { 0x01470700, 0x00000000 }, + { 0x01470500, 0x00000400 }, + { 0x01470c00, 0x00000000 }, + { 0x01470100, 0x00000000 }, + { 0x01837000, 0x00000000 }, + { 0x01870500, 0x00000400 }, + { 0x02050000, 0x00000000 }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000000 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x02040000, 0x00004002 }, + { 0x01870700, 0x00000020 }, + { 0x00830000, 0x000000c3 }, + { 0x00930000, 0x000000c3 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt286_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_PROC_COEF: + return true; + default: + return false; + } + + +} + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_SET_AUDIO_POWER: + case RT286_SET_HPO_POWER: + case RT286_SET_SPK_POWER: + case RT286_SET_DMIC1_POWER: + case RT286_SPK_MUX: + case RT286_HPO_MUX: + case RT286_ADC0_MUX: + case RT286_ADC1_MUX: + case RT286_SET_MIC1: + case RT286_SET_PIN_HPO: + case RT286_SET_PIN_SPK: + case RT286_SET_PIN_DMIC1: + case RT286_SPK_EAPD: + case RT286_SET_AMP_GAIN_HPO: + case RT286_SET_DMIC2_DEFAULT: + case RT286_DACL_GAIN: + case RT286_DACR_GAIN: + case RT286_ADCL_GAIN: + case RT286_ADCR_GAIN: + case RT286_MIC_GAIN: + case RT286_SPOL_GAIN: + case RT286_SPOR_GAIN: + case RT286_HPOL_GAIN: + case RT286_HPOR_GAIN: + case RT286_F_DAC_SWITCH: + case RT286_F_RECMIX_SWITCH: + case RT286_REC_MIC_SWITCH: + case RT286_REC_I2S_SWITCH: + case RT286_REC_LINE_SWITCH: + case RT286_REC_BEEP_SWITCH: + case RT286_DAC_FORMAT: + case RT286_ADC_FORMAT: + case RT286_COEF_INDEX: + case RT286_PROC_COEF: + case RT286_SET_AMP_GAIN_ADC_IN1: + case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_POWER(RT286_DAC_OUT1): + case RT286_SET_POWER(RT286_DAC_OUT2): + case RT286_SET_POWER(RT286_ADC_IN1): + case RT286_SET_POWER(RT286_ADC_IN2): + case RT286_SET_POWER(RT286_DMIC2): + case RT286_SET_POWER(RT286_MIC1): + return true; + default: + return false; + } +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + struct rt286_priv *rt286 = i2c_get_clientdata(client); + u8 data[4]; + int ret, i; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + if (reg == rt286->index_cache[i].reg) { + rt286->index_cache[i].def = value; + break; + } + + } + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + + if (ret == 4) + return 0; + else + pr_err("ret=%d\n", ret); + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + } + + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static void rt286_index_sync(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + } +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL2, 0x4, 0x4); + snd_soc_write(codec, RT286_SET_MIC1, 0x24); + val = snd_soc_read(codec, RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = snd_soc_read(codec, + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + snd_soc_write(codec, + RT286_SET_MIC1, 0x20); + /* power off HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x1001); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } else { + *mic = false; + } + + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE); + *mic = buf & 0x80000000; + } + + return 0; +} + +static void rt286_jack_detect_work(struct work_struct *work) +{ + struct rt286_priv *rt286 = + container_of(work, struct rt286_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, + RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN, + RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK, + RT286_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN, + RT286_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN, + RT286_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt286_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum); + +static const char * const rt286_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, + RT286_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt286_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 rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + else + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, + RT286_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x200); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(10); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + /* Clear IRQ */ + snd_soc_update_bits(rt286->codec, + RT286_IRQ_CTRL, 0x1, 0x1); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt286->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt286_suspend(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, true); + regcache_mark_dirty(rt286->regmap); + + return 0; +} + +static int rt286_resume(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, false); + rt286_index_sync(codec); + regcache_sync(rt286->regmap); + + return 0; +} +#else +#define rt286_suspend NULL +#define rt286_resume NULL +#endif + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt286-aif2", + .id = RT286_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .suspend = rt286_suspend, + .resume = rt286_resume, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x02370100, + .volatile_reg = rt286_volatile_register, + .readable_reg = rt286_readable_register, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt286_reg, + .num_reg_defaults = ARRAY_SIZE(rt286_reg), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static const struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + int i, ret; + + rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt286->regmap, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); + if (ret != RT286_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + + rt286->index_cache = rt286_index_def; + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + regmap_write(rt286->regmap, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000); + regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816); + regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + + /*Set depop parameter*/ + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); + + if (rt286->i2c->irq) { + regmap_update_bits(rt286->regmap, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt286); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +static struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 000000000000..b539b7320a79 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,198 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu@realtek.com> + * + * 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. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_SET_PIN_SFT 6 +#define RT286_SET_PIN_ENABLE 0x40 +#define RT286_SET_PIN_DISABLE 0 +#define RT286_SET_EAPD_HIGH 0x2 +#define RT286_SET_EAPD_LOW 0 + +#define RT286_MUTE_SFT 7 + +/* Verb commands */ +#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT) +#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT) +#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1) +#define RT286_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0) +#define RT286_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0) +#define RT286_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0) +#define RT286_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0) +#define RT286_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0) +#define RT286_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0) +#define RT286_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0) +#define RT286_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) +#define RT286_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000) +#define RT286_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000) +#define RT286_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000) +#define RT286_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000) +#define RT286_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000) +#define RT286_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000) +#define RT286_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000) +#define RT286_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000) +#define RT286_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100) +#define RT286_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000) +#define RT286_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100) +#define RT286_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200) +#define RT286_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300) +#define RT286_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0) +#define RT286_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0) +#define RT286_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0) +#define RT286_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_DC_GAIN 0x0d +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 +#define RT286_DEPOP_CTRL1 0x66 +#define RT286_DEPOP_CTRL2 0x67 +#define RT286_DEPOP_CTRL3 0x68 +#define RT286_DEPOP_CTRL4 0x69 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ + diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3d39f0b5b4a8..8f4c73d17c87 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1277,7 +1277,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return ret; } - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) goto err_ldo_remove; @@ -1285,13 +1285,16 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_ldo_remove; + goto err_regulator_free; /* wait for all power rails bring up */ udelay(10); return 0; +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); err_ldo_remove: if (!external_vddd) ldo_regulator_remove(codec); @@ -1361,6 +1364,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return ret; @@ -1374,6 +1379,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return 0; diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c new file mode 100644 index 000000000000..246081aae8ca --- /dev/null +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -0,0 +1,35 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/i2c.h> +#include <linux/export.h> +#include <linux/module.h> + +#include "sigmadsp.h" + +static int sigma_action_write_i2c(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return i2c_master_send(control_data, (const unsigned char *)&sa->addr, + len); +} + +int process_sigma_firmware(struct i2c_client *client, const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = client; + ssfw.write = sigma_action_write_i2c; + + return _process_sigma_firmware(&client->dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("SigmaDSP I2C firmware loader"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp-regmap.c b/sound/soc/codecs/sigmadsp-regmap.c new file mode 100644 index 000000000000..f78ed8d2cfb2 --- /dev/null +++ b/sound/soc/codecs/sigmadsp-regmap.c @@ -0,0 +1,36 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/regmap.h> +#include <linux/export.h> +#include <linux/module.h> + +#include "sigmadsp.h" + +static int sigma_action_write_regmap(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), + sa->payload, len - 2); +} + +int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, + const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = regmap; + ssfw.write = sigma_action_write_regmap; + + return _process_sigma_firmware(dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware_regmap); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("SigmaDSP regmap firmware loader"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 4068f2491232..f2de7e049bc6 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -34,23 +34,6 @@ enum { SIGMA_ACTION_END, }; -struct sigma_action { - u8 instr; - u8 len_hi; - __le16 len; - __be16 addr; - unsigned char payload[]; -} __packed; - -struct sigma_firmware { - const struct firmware *fw; - size_t pos; - - void *control_data; - int (*write)(void *control_data, const struct sigma_action *sa, - size_t len); -}; - static inline u32 sigma_action_len(struct sigma_action *sa) { return (sa->len_hi << 16) | le16_to_cpu(sa->len); @@ -138,7 +121,7 @@ process_sigma_actions(struct sigma_firmware *ssfw) return 0; } -static int _process_sigma_firmware(struct device *dev, +int _process_sigma_firmware(struct device *dev, struct sigma_firmware *ssfw, const char *name) { int ret; @@ -197,50 +180,6 @@ static int _process_sigma_firmware(struct device *dev, return ret; } - -#if IS_ENABLED(CONFIG_I2C) - -static int sigma_action_write_i2c(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return i2c_master_send(control_data, (const unsigned char *)&sa->addr, - len); -} - -int process_sigma_firmware(struct i2c_client *client, const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = client; - ssfw.write = sigma_action_write_i2c; - - return _process_sigma_firmware(&client->dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware); - -#endif - -#if IS_ENABLED(CONFIG_REGMAP) - -static int sigma_action_write_regmap(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return regmap_raw_write(control_data, be16_to_cpu(sa->addr), - sa->payload, len - 2); -} - -int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, - const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = regmap; - ssfw.write = sigma_action_write_regmap; - - return _process_sigma_firmware(dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware_regmap); - -#endif +EXPORT_SYMBOL_GPL(_process_sigma_firmware); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index e439cbd7af7d..c47cd23e9827 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -12,6 +12,26 @@ #include <linux/device.h> #include <linux/regmap.h> +struct sigma_action { + u8 instr; + u8 len_hi; + __le16 len; + __be16 addr; + unsigned char payload[]; +} __packed; + +struct sigma_firmware { + const struct firmware *fw; + size_t pos; + + void *control_data; + int (*write)(void *control_data, const struct sigma_action *sa, + size_t len); +}; + +int _process_sigma_firmware(struct device *dev, + struct sigma_firmware *ssfw, const char *name); + struct i2c_client; extern int process_sigma_firmware(struct i2c_client *client, const char *name); diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 23419109ecac..1cdae8ccc61b 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1178,7 +1178,7 @@ static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) } #endif /* CONFIG_OF */ -static void aic31xx_device_init(struct aic31xx_priv *aic31xx) +static int aic31xx_device_init(struct aic31xx_priv *aic31xx) { int ret, i; @@ -1197,7 +1197,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx) "aic31xx-reset-pin"); if (ret < 0) { dev_err(aic31xx->dev, "not able to acquire gpio\n"); - return; + return ret; } } @@ -1210,6 +1210,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx) if (ret != 0) dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + return ret; } static int aic31xx_i2c_probe(struct i2c_client *i2c, @@ -1239,7 +1240,9 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx->pdata.codec_type = id->driver_data; - aic31xx_device_init(aic31xx); + ret = aic31xx_device_init(aic31xx); + if (ret) + return ret; return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e12fafbb1e09..5360772bc1ad 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index df3a7506c023..ff006cc95520 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1404,7 +1404,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, IRQF_TRIGGER_RISING, - codec->name, codec); + codec->component.name, codec); if (ret < 0) { dev_err(codec->dev, "Could not request IRQ%d (%d)\n", dac33->irq, ret); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 69e12a311ba2..6ab157065353 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -344,17 +344,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status = -1; if (enable) { twl4030->apll_enabled++; if (twl4030->apll_enabled == 1) - status = twl4030_audio_enable_resource( + twl4030_audio_enable_resource( TWL4030_AUDIO_RES_APLL); } else { twl4030->apll_enabled--; if (!twl4030->apll_enabled) - status = twl4030_audio_disable_resource( + twl4030_audio_disable_resource( TWL4030_AUDIO_RES_APLL); } } diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index edf27acc1d77..12fc0aed7503 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -479,7 +479,7 @@ static struct snd_soc_dai_driver uda134x_dai = { static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; - struct uda134x_platform_data *pd = codec->card->dev->platform_data; + struct uda134x_platform_data *pd = codec->component.card->dev->platform_data; const struct snd_soc_dapm_widget *widgets; unsigned num_widgets; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 91a9ea2a2056..7bb0d36d4c54 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -735,8 +735,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); u16 val, expect, i; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 289b64d89abd..f60234962527 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -612,6 +612,62 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + uint16_t data; + + mutex_lock(&codec->mutex); + data = cpu_to_be16(arizona->dac_comp_coeff); + memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, + sizeof(arizona->dac_comp_coeff)); + arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = arizona->dac_comp_enabled; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + mutex_unlock(&codec->mutex); + + return 0; +} + static const char *wm5102_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -843,6 +899,12 @@ SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), SOC_ENUM("Noise Gate Hold", arizona_ng_hold), +SND_SOC_BYTES_EXT("Output Compensation Coefficient", 2, + wm5102_out_comp_coeff_get, wm5102_out_comp_coeff_put), + +SOC_SINGLE_EXT("Output Compensation Switch", 0, 0, 1, 0, + wm5102_out_comp_switch_get, wm5102_out_comp_switch_put), + WM5102_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), WM5102_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), WM5102_NG_SRC("HPOUT2L", ARIZONA_NOISE_GATE_SELECT_2L), @@ -1653,6 +1715,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-aif2", @@ -1674,6 +1737,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-aif3", @@ -1695,6 +1759,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-slim1", diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e5fcb559e90..2f2ec26d831c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1485,6 +1485,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-aif2", @@ -1506,6 +1507,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-aif3", @@ -1527,6 +1529,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-slim1", @@ -1596,6 +1599,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + arizona_init_mono(codec); ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 392285edb595..d9e634c55e81 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1341,21 +1341,18 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); struct wm8350 *wm8350 = priv->wm8350; - int irq; int ena; switch (which) { case WM8350_JDL: priv->hpl.jack = jack; priv->hpl.report = report; - irq = WM8350_IRQ_CODEC_JCK_DET_L; ena = WM8350_JDL_ENA; break; case WM8350_JDR: priv->hpr.jack = jack; priv->hpr.report = report; - irq = WM8350_IRQ_CODEC_JCK_DET_R; ena = WM8350_JDR_ENA; break; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b84940c359a1..ec3250daa93e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -281,8 +281,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; int i, val; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index a145d0431b63..e96349b04ba6 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -472,7 +472,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) * list each time to find the desired power state do so now * and save the result. */ - list_for_each_entry(w, &codec->card->widgets, list) { + list_for_each_entry(w, &codec->component.card->widgets, list) { if (w->dapm != &codec->dapm) continue; if (strcmp(w->name, "LOUT1 PGA") == 0) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b39013fba..9719d3ca8e47 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3505,6 +3505,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) return IRQ_HANDLED; } +/* Should be called with accdet_lock held */ static void wm1811_micd_stop(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -3512,14 +3513,10 @@ static void wm1811_micd_stop(struct snd_soc_codec *codec) if (!wm8994->jackdet) return; - mutex_lock(&wm8994->accdet_lock); - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK); - mutex_unlock(&wm8994->accdet_lock); - if (wm8994->wm8994->pdata.jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); @@ -3560,10 +3557,10 @@ static void wm8958_open_circuit_work(struct work_struct *work) open_circuit_work.work); struct device *dev = wm8994->wm8994->dev; - wm1811_micd_stop(wm8994->hubs.codec); - mutex_lock(&wm8994->accdet_lock); + wm1811_micd_stop(wm8994->hubs.codec); + dev_dbg(dev, "Reporting open circuit\n"); wm8994->jack_mic = false; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 69266332760e..f16ff4f56923 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -620,15 +620,12 @@ static int bg_event(struct snd_soc_dapm_widget *w, static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - int ret = 0; - switch (event) { case SND_SOC_DAPM_POST_PMU: msleep(5); break; default: WARN(1, "Invalid event %d\n", event); - ret = -EINVAL; } return 0; @@ -690,8 +687,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); u16 val, mask; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index bb9b47b956aa..ab33fe596519 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -967,6 +967,7 @@ static struct snd_soc_dai_driver wm8997_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm8997-aif2", @@ -988,6 +989,7 @@ static struct snd_soc_dai_driver wm8997_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm8997-slim1", diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 060027182dcb..f412a9911a75 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1382,7 +1382,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, int ret; int val; - dsp->card = codec->card; + dsp->card = codec->component.card; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1617,7 +1617,7 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - dsp->card = codec->card; + dsp->card = codec->component.card; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -1758,3 +1758,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_init); + +MODULE_LICENSE("GPL v2"); |