diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2017-07-06 20:56:51 +0300 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2017-07-06 20:56:51 +0300 |
commit | 920f2ecdf6c3b3526f60fbd38c68597953cad3ee (patch) | |
tree | 18188922ba38a5c53ee8d17032eb5c46dffc7fa2 /sound/soc/codecs | |
parent | 9ced560b82606b35adb33a27012a148d418a4c1f (diff) | |
parent | fc18282cdcba984ab89c74d7e844c10114ae0795 (diff) | |
download | linux-920f2ecdf6c3b3526f60fbd38c68597953cad3ee.tar.xz |
Merge tag 'sound-4.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This development cycle resulted in a fair amount of changes in both
core and driver sides. The most significant change in ALSA core is
about PCM. Also the support of of-graph card and the new DAPM widget
for DSP are noteworthy changes in ASoC core. And there're lots of
small changes splat over the tree, as you can see in diffstat.
Below are a few highlights:
ALSA core:
- Removal of set_fs() hackery from PCM core stuff, and the code
reorganization / optimization thereafter
- Improved support of PCM ack ops, and a new ABI for improved
control/status mmap handling
- Lots of constifications in various codes
ASoC core:
- The support of of-graph card, which may work as a better generic
device for a replacement of simple-card
- New widget types intended mainly for use with DSPs
ASoC drivers:
- New drivers for Allwinner V3s SoCs
- Ensonic ES8316 codec support
- More Intel SKL and KBL works
- More device support for Intel SST Atom (mostly for cheap tablets
and 2-in-1 devices)
- Support for Rockchip PDM controllers
- Support for STM32 I2S and S/PDIF controllers
- Support for ZTE AUD96P22 codecs
HD-audio:
- Support of new Realtek codecs (ALC215/ALC285/ALC289), more quirks
for HP and Dell machines
- A few more fixes for i915 component binding"
* tag 'sound-4.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (418 commits)
ALSA: hda - Fix unbalance of i915 module refcount
ASoC: Intel: Skylake: Remove driver debugfs exit
ASoC: Intel: Skylake: explicitly add the headers sst-dsp.h
ALSA: hda/realtek - Remove GPIO_MASK
ALSA: hda/realtek - Fix typo of pincfg for Dell quirk
ALSA: pcm: add a documentation for tracepoints
ALSA: atmel: ac97c: fix error return code in atmel_ac97c_probe()
ALSA: x86: fix error return code in hdmi_lpe_audio_probe()
ASoC: Intel: Skylake: Add support to read firmware registers
ASoC: Intel: Skylake: Add sram address to sst_addr structure
ASoC: Intel: Skylake: Debugfs facility to dump module config
ASoC: Intel: Skylake: Add debugfs support
ASoC: fix semicolon.cocci warnings
ASoC: rt5645: Add quirk override by module option
ASoC: rsnd: make arrays path and cmd_case static const
ASoC: audio-graph-card: add widgets and routing for external amplifier support
ASoC: audio-graph-card: update bindings for amplifier support
ASoC: rt5665: calibration should be done before jack detection
ASoC: rsnd: constify dev_pm_ops structures.
ASoC: nau8825: change crosstalk-bypass property to bool type
...
Diffstat (limited to 'sound/soc/codecs')
36 files changed, 1907 insertions, 356 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883ed4c8a551..6c78b0b49b81 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 select SND_SOC_DMIC + select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 @@ -543,6 +544,10 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC" +config SND_SOC_ES8316 + tristate "Everest Semi ES8316 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate @@ -1114,6 +1119,11 @@ config SND_SOC_WM9713 tristate select REGMAP_AC97 +config SND_SOC_ZX_AUD96P22 + tristate "ZTE ZX AUD96P22 CODEC" + depends on I2C + select REGMAP_I2C + # Amp config SND_SOC_LM4857 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28a63fdaf982..1755a54e3dc9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -224,6 +225,7 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o +snd-soc-zx-aud96p22-objs := zx_aud96p22.o # Amp snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o @@ -300,6 +302,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o @@ -455,6 +458,7 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_ADSP) += snd-soc-wm-adsp.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o +obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o # Amp obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index b2dfddead227..690edebf029e 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -94,6 +94,8 @@ struct ak4613_interface { struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; + struct snd_pcm_hw_constraint_list constraint; + unsigned int sysclk; unsigned int fmt; u8 oc; @@ -139,9 +141,7 @@ static const struct reg_default ak4613_reg[] = { #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } static const struct ak4613_interface ak4613_iface[] = { /* capture */ /* playback */ - [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, - [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, - [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + /* [0] - [2] are not supported */ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, }; @@ -254,6 +254,74 @@ static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, mutex_unlock(&priv->lock); } +static void ak4613_hw_constraints(struct ak4613_priv *priv, + struct snd_pcm_runtime *runtime) +{ + static const unsigned int ak4613_rates[] = { + 32000, + 44100, + 48000, + 64000, + 88200, + 96000, + 176400, + 192000, + }; + struct snd_pcm_hw_constraint_list *constraint = &priv->constraint; + unsigned int fs; + int i; + + constraint->list = ak4613_rates; + constraint->mask = 0; + constraint->count = 0; + + /* + * Slave Mode + * Normal: [32kHz, 48kHz] : 256fs,384fs or 512fs + * Double: [64kHz, 96kHz] : 256fs + * Quad : [128kHz,192kHz]: 128fs + * + * Master mode + * Normal: [32kHz, 48kHz] : 256fs or 512fs + * Double: [64kHz, 96kHz] : 256fs + * Quad : [128kHz,192kHz]: 128fs + */ + for (i = 0; i < ARRAY_SIZE(ak4613_rates); i++) { + /* minimum fs on each range */ + fs = (ak4613_rates[i] <= 96000) ? 256 : 128; + + if (priv->sysclk >= ak4613_rates[i] * fs) + constraint->count = i + 1; + } + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, constraint); +} + +static int ak4613_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->cnt++; + + ak4613_hw_constraints(priv, substream->runtime); + + return 0; +} + +static int ak4613_dai_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 ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->sysclk = freq; + + return 0; +} + static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; @@ -262,11 +330,9 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) fmt &= SND_SOC_DAIFMT_FORMAT_MASK; switch (fmt) { - case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_I2S: priv->fmt = fmt; - break; default: return -EINVAL; @@ -286,13 +352,8 @@ static bool ak4613_dai_fmt_matching(const struct ak4613_interface *iface, if (fmts->fmt != fmt) return false; - if (fmt == SND_SOC_DAIFMT_RIGHT_J) { - if (fmts->width != width) - return false; - } else { - if (fmts->width < width) - return false; - } + if (fmts->width != width) + return false; return true; } @@ -319,6 +380,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, case 48000: ctrl2 = DFS_NORMAL_SPEED; break; + case 64000: case 88200: case 96000: ctrl2 = DFS_DOUBLE_SPEED; @@ -345,7 +407,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, if (ak4613_dai_fmt_matching(priv->iface, is_play, fmt, width)) iface = priv->iface; } else { - for (i = ARRAY_SIZE(ak4613_iface); i >= 0; i--) { + for (i = ARRAY_SIZE(ak4613_iface) - 1; i >= 0; i--) { if (!ak4613_dai_fmt_matching(ak4613_iface + i, is_play, fmt, width)) @@ -358,7 +420,6 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, if ((priv->iface == NULL) || (priv->iface == iface)) { priv->iface = iface; - priv->cnt++; ret = 0; } mutex_unlock(&priv->lock); @@ -407,7 +468,9 @@ static int ak4613_set_bias_level(struct snd_soc_codec *codec, } static const struct snd_soc_dai_ops ak4613_dai_ops = { + .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, + .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, .hw_params = ak4613_dai_hw_params, }; @@ -420,8 +483,7 @@ static const struct snd_soc_dai_ops ak4613_dai_ops = { SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) -#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_driver ak4613_dai = { .name = "ak4613-hifi", @@ -527,6 +589,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; + priv->sysclk = 0; mutex_init(&priv->lock); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 23ab9646c351..66de8a2013a6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -433,7 +433,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int ak4642_set_mcko(struct snd_soc_codec *codec, u32 frequency) { - u32 fs_list[] = { + static const u32 fs_list[] = { [0] = 8000, [1] = 12000, [2] = 16000, @@ -447,7 +447,7 @@ static int ak4642_set_mcko(struct snd_soc_codec *codec, [14] = 29400, [15] = 44100, }; - u32 ps_list[] = { + static const u32 ps_list[] = { [0] = 256, [1] = 128, [2] = 64, diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index 7c5d1510cf2c..0a747c66cc6c 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -567,12 +567,12 @@ static int cs35l34_pcm_hw_params(struct snd_pcm_substream *substream, return ret; } -static unsigned int cs35l34_src_rates[] = { +static const unsigned int cs35l34_src_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }; -static struct snd_pcm_hw_constraint_list cs35l34_constraints = { +static const struct snd_pcm_hw_constraint_list cs35l34_constraints = { .count = ARRAY_SIZE(cs35l34_src_rates), .list = cs35l34_src_rates, }; diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index f8aef5869b03..f1ee184ecab2 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -162,6 +162,14 @@ static bool cs35l35_precious_register(struct device *dev, unsigned int reg) } } +static void cs35l35_reset(struct cs35l35_private *cs35l35) +{ + gpiod_set_value_cansleep(cs35l35->reset_gpio, 0); + usleep_range(2000, 2100); + gpiod_set_value_cansleep(cs35l35->reset_gpio, 1); + usleep_range(1000, 1100); +} + static int cs35l35_wait_for_pdn(struct cs35l35_private *cs35l35) { int ret; @@ -756,6 +764,76 @@ static int cs35l35_codec_set_sysclk(struct snd_soc_codec *codec, return ret; } +static int cs35l35_boost_inductor(struct cs35l35_private *cs35l35, + int inductor) +{ + struct regmap *regmap = cs35l35->regmap; + unsigned int bst_ipk = 0; + + /* + * Digital Boost Converter Configuration for feedback, + * ramping, switching frequency, and estimation block seeding. + */ + + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_SWFREQ_MASK, 0x00); + + regmap_read(regmap, CS35L35_BST_PEAK_I, &bst_ipk); + bst_ipk &= CS35L35_BST_IPK_MASK; + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x24); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x24); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x00); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x4E); + break; + case 1200: /* 1.2 uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x20); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x20); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x01); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x47); + break; + case 1500: /* 1.5uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x20); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x20); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x02); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x3C); + break; + case 2200: /* 2.2uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x19); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x25); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x03); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x23); + break; + default: + dev_err(cs35l35->dev, "Invalid Inductor Value %d uH\n", + inductor); + return -EINVAL; + } + return 0; +} + static int cs35l35_codec_probe(struct snd_soc_codec *codec) { struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); @@ -775,6 +853,10 @@ static int cs35l35_codec_probe(struct snd_soc_codec *codec) cs35l35->pdata.bst_ipk << CS35L35_BST_IPK_SHIFT); + ret = cs35l35_boost_inductor(cs35l35, cs35l35->pdata.boost_ind); + if (ret) + return ret; + if (cs35l35->pdata.gain_zc) regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, CS35L35_AMP_GAIN_ZC_MASK, @@ -1195,7 +1277,15 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, return -EINVAL; } - pdata->bst_ipk = (val32 - 1680) / 110; + pdata->bst_ipk = ((val32 - 1680) / 110) | CS35L35_VALID_PDATA; + } + + ret = of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val32); + if (ret >= 0) { + pdata->boost_ind = val32; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; } if (of_property_read_u32(np, "cirrus,sp-drv-strength", &val32) >= 0) @@ -1454,7 +1544,7 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, } } - gpiod_set_value_cansleep(cs35l35->reset_gpio, 1); + cs35l35_reset(cs35l35); init_completion(&cs35l35->pdn_done); diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index 5a6e43a87c4d..621bfef70d03 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -200,6 +200,12 @@ #define CS35L35_SP_I2S_DRV_MASK 0x03 #define CS35L35_SP_I2S_DRV_SHIFT 0 +/* Boost Converter Config */ +#define CS35L35_BST_CONV_COEFF_MASK 0xFF +#define CS35L35_BST_CONV_SLOPE_MASK 0xFF +#define CS35L35_BST_CONV_LBST_MASK 0x03 +#define CS35L35_BST_CONV_SWFREQ_MASK 0xF0 + /* Class H Algorithm Control */ #define CS35L35_CH_STEREO_MASK 0x40 #define CS35L35_CH_STEREO_SHIFT 6 diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e78b5f055f25..d8824773dc29 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -674,8 +674,6 @@ static int cs4271_common_probe(struct device *dev, cs4271->gpio_nreset = cs4271plat->gpio_nreset; if (gpio_is_valid(cs4271->gpio_nreset)) { - int ret; - ret = devm_gpio_request(dev, cs4271->gpio_nreset, "CS4271 Reset"); if (ret < 0) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 1e0d5973b758..06933a5d0a75 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -747,7 +747,7 @@ static unsigned int const cs53l30_src_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }; -static struct snd_pcm_hw_constraint_list src_constraints = { +static const struct snd_pcm_hw_constraint_list src_constraints = { .count = ARRAY_SIZE(cs53l30_src_rates), .list = cs53l30_src_rates, }; diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 024d83fa6a7f..c3e11897f8ae 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -13,6 +13,8 @@ */ #include <linux/acpi.h> +#include <linux/of_device.h> +#include <linux/property.h> #include <linux/clk.h> #include <linux/delay.h> #include <linux/i2c.h> @@ -1606,12 +1608,12 @@ static enum da7213_dmic_clk_rate } static struct da7213_platform_data - *da7213_of_to_pdata(struct snd_soc_codec *codec) + *da7213_fw_to_pdata(struct snd_soc_codec *codec) { - struct device_node *np = codec->dev->of_node; + struct device *dev = codec->dev; struct da7213_platform_data *pdata; - const char *of_str; - u32 of_val32; + const char *fw_str; + u32 fw_val32; pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); if (!pdata) { @@ -1619,29 +1621,29 @@ static struct da7213_platform_data return NULL; } - if (of_property_read_u32(np, "dlg,micbias1-lvl", &of_val32) >= 0) - pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, of_val32); + if (device_property_read_u32(dev, "dlg,micbias1-lvl", &fw_val32) >= 0) + pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, fw_val32); else pdata->micbias1_lvl = DA7213_MICBIAS_2_2V; - if (of_property_read_u32(np, "dlg,micbias2-lvl", &of_val32) >= 0) - pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, of_val32); + if (device_property_read_u32(dev, "dlg,micbias2-lvl", &fw_val32) >= 0) + pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, fw_val32); else pdata->micbias2_lvl = DA7213_MICBIAS_2_2V; - if (!of_property_read_string(np, "dlg,dmic-data-sel", &of_str)) - pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, of_str); + if (!device_property_read_string(dev, "dlg,dmic-data-sel", &fw_str)) + pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, fw_str); else pdata->dmic_data_sel = DA7213_DMIC_DATA_LRISE_RFALL; - if (!of_property_read_string(np, "dlg,dmic-samplephase", &of_str)) + if (!device_property_read_string(dev, "dlg,dmic-samplephase", &fw_str)) pdata->dmic_samplephase = - da7213_of_dmic_samplephase(codec, of_str); + da7213_of_dmic_samplephase(codec, fw_str); else pdata->dmic_samplephase = DA7213_DMIC_SAMPLE_ON_CLKEDGE; - if (of_property_read_u32(np, "dlg,dmic-clkrate", &of_val32) >= 0) - pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, of_val32); + if (device_property_read_u32(dev, "dlg,dmic-clkrate", &fw_val32) >= 0) + pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, fw_val32); else pdata->dmic_clk_rate = DA7213_DMIC_CLK_3_0MHZ; @@ -1713,10 +1715,9 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE); /* Handle DT/Platform data */ - if (codec->dev->of_node) - da7213->pdata = da7213_of_to_pdata(codec); - else - da7213->pdata = dev_get_platdata(codec->dev); + da7213->pdata = dev_get_platdata(codec->dev); + if (!da7213->pdata) + da7213->pdata = da7213_fw_to_pdata(codec); /* Set platform data values */ if (da7213->pdata) { diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index d256ebf9e309..6e1940eb0653 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1457,7 +1457,7 @@ static int da7218_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(DA7218_SRM_CHECK_DELAY); } - } while ((i < DA7218_SRM_CHECK_TRIES) & (!success)); + } while ((i < DA7218_SRM_CHECK_TRIES) && (!success)); if (!success) dev_warn(codec->dev, "SRM failed to lock\n"); diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 6274d79c1353..1d1d10dd92ae 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -115,19 +115,21 @@ static void da7219_aad_hptest_work(struct work_struct *work) struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); u16 tonegen_freq_hptest; - u8 pll_srm_sts, gain_ramp_ctrl, accdet_cfg8; + u8 pll_srm_sts, pll_ctrl, gain_ramp_ctrl, accdet_cfg8; int report = 0, ret = 0; - /* Lock DAPM and any Kcontrols that are affected by this test */ + /* Lock DAPM, Kcontrols affected by this test and the PLL */ snd_soc_dapm_mutex_lock(dapm); - mutex_lock(&da7219->lock); + mutex_lock(&da7219->ctrl_lock); + mutex_lock(&da7219->pll_lock); /* Ensure MCLK is available for HP test procedure */ if (da7219->mclk) { ret = clk_prepare_enable(da7219->mclk); if (ret) { dev_err(codec->dev, "Failed to enable mclk - %d\n", ret); - mutex_unlock(&da7219->lock); + mutex_unlock(&da7219->pll_lock); + mutex_unlock(&da7219->ctrl_lock); snd_soc_dapm_mutex_unlock(dapm); return; } @@ -136,12 +138,21 @@ static void da7219_aad_hptest_work(struct work_struct *work) /* * If MCLK not present, then we're using the internal oscillator and * require different frequency settings to achieve the same result. + * + * If MCLK is present, but PLL is not enabled then we enable it here to + * ensure a consistent detection procedure. */ pll_srm_sts = snd_soc_read(codec, DA7219_PLL_SRM_STS); - if (pll_srm_sts & DA7219_PLL_SRM_STS_MCLK) + if (pll_srm_sts & DA7219_PLL_SRM_STS_MCLK) { tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ); - else + + pll_ctrl = snd_soc_read(codec, DA7219_PLL_CTRL); + if ((pll_ctrl & DA7219_PLL_MODE_MASK) == DA7219_PLL_MODE_BYPASS) + da7219_set_pll(codec, DA7219_SYSCLK_PLL, + DA7219_PLL_FREQ_OUT_98304); + } else { tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ_INT_OSC); + } /* Ensure gain ramping at fastest rate */ gain_ramp_ctrl = snd_soc_read(codec, DA7219_GAIN_RAMP_CTRL); @@ -302,11 +313,17 @@ static void da7219_aad_hptest_work(struct work_struct *work) snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK, DA7219_HP_R_AMP_OE_MASK); + /* Restore PLL to previous configuration, if re-configured */ + if ((pll_srm_sts & DA7219_PLL_SRM_STS_MCLK) && + ((pll_ctrl & DA7219_PLL_MODE_MASK) == DA7219_PLL_MODE_BYPASS)) + da7219_set_pll(codec, DA7219_SYSCLK_MCLK, 0); + /* Remove MCLK, if previously enabled */ if (da7219->mclk) clk_disable_unprepare(da7219->mclk); - mutex_unlock(&da7219->lock); + mutex_unlock(&da7219->pll_lock); + mutex_unlock(&da7219->ctrl_lock); snd_soc_dapm_mutex_unlock(dapm); /* diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 99601627f83c..f71d72c22bfc 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -260,9 +260,9 @@ static int da7219_volsw_locked_get(struct snd_kcontrol *kcontrol, struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); int ret; - mutex_lock(&da7219->lock); + mutex_lock(&da7219->ctrl_lock); ret = snd_soc_get_volsw(kcontrol, ucontrol); - mutex_unlock(&da7219->lock); + mutex_unlock(&da7219->ctrl_lock); return ret; } @@ -274,9 +274,9 @@ static int da7219_volsw_locked_put(struct snd_kcontrol *kcontrol, struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); int ret; - mutex_lock(&da7219->lock); + mutex_lock(&da7219->ctrl_lock); ret = snd_soc_put_volsw(kcontrol, ucontrol); - mutex_unlock(&da7219->lock); + mutex_unlock(&da7219->ctrl_lock); return ret; } @@ -288,9 +288,9 @@ static int da7219_enum_locked_get(struct snd_kcontrol *kcontrol, struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); int ret; - mutex_lock(&da7219->lock); + mutex_lock(&da7219->ctrl_lock); ret = snd_soc_get_enum_double(kcontrol, ucontrol); - mutex_unlock(&da7219->lock); + mutex_unlock(&da7219->ctrl_lock); return ret; } @@ -302,9 +302,9 @@ static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol, struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); int ret; - mutex_lock(&da7219->lock); + mutex_lock(&da7219->ctrl_lock); ret = snd_soc_put_enum_double(kcontrol, ucontrol); - mutex_unlock(&da7219->lock); + mutex_unlock(&da7219->ctrl_lock); return ret; } @@ -424,9 +424,9 @@ static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol, u16 val; int ret; - mutex_lock(&da7219->lock); + mutex_lock(&da7219->ctrl_lock); ret = regmap_raw_read(da7219->regmap, reg, &val, sizeof(val)); - mutex_unlock(&da7219->lock); + mutex_unlock(&da7219->ctrl_lock); if (ret) return ret; @@ -458,9 +458,9 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, */ val = cpu_to_le16(ucontrol->value.integer.value[0]); - mutex_lock(&da7219->lock); + mutex_lock(&da7219->ctrl_lock); ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val)); - mutex_unlock(&da7219->lock); + mutex_unlock(&da7219->ctrl_lock); return ret; } @@ -801,7 +801,7 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(50); } - } while ((i < DA7219_SRM_CHECK_RETRIES) && (!srm_lock)); + } while ((i < DA7219_SRM_CHECK_RETRIES) & (!srm_lock)); if (!srm_lock) dev_warn(codec->dev, "SRM failed to lock\n"); @@ -1129,6 +1129,8 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } + mutex_lock(&da7219->pll_lock); + switch (clk_id) { case DA7219_CLKSRC_MCLK_SQR: snd_soc_update_bits(codec, DA7219_PLL_CTRL, @@ -1141,6 +1143,7 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, break; default: dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + mutex_unlock(&da7219->pll_lock); return -EINVAL; } @@ -1152,19 +1155,20 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (ret) { dev_err(codec_dai->dev, "Failed to set clock rate %d\n", freq); + mutex_unlock(&da7219->pll_lock); return ret; } } da7219->mclk_rate = freq; + mutex_unlock(&da7219->pll_lock); + return 0; } -static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int fref, unsigned int fout) +int da7219_set_pll(struct snd_soc_codec *codec, int source, unsigned int fout) { - struct snd_soc_codec *codec = codec_dai->codec; struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); u8 pll_ctrl, indiv_bits, indiv; @@ -1237,6 +1241,20 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->pll_lock); + ret = da7219_set_pll(codec, source, fout); + mutex_unlock(&da7219->pll_lock); + + return ret; +} + static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1741,7 +1759,8 @@ static int da7219_probe(struct snd_soc_codec *codec) unsigned int rev; int ret; - mutex_init(&da7219->lock); + mutex_init(&da7219->ctrl_lock); + mutex_init(&da7219->pll_lock); /* Regulator configuration */ ret = da7219_handle_supplies(codec); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 6baba7455fa1..8d6c3c8c8026 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -810,7 +810,8 @@ struct da7219_priv { bool wakeup_source; struct regulator_bulk_data supplies[DA7219_NUM_SUPPLIES]; struct regmap *regmap; - struct mutex lock; + struct mutex ctrl_lock; + struct mutex pll_lock; struct clk *mclk; unsigned int mclk_rate; @@ -821,4 +822,6 @@ struct da7219_priv { u8 gain_ramp_ctrl; }; +int da7219_set_pll(struct snd_soc_codec *codec, int source, unsigned int fout); + #endif /* __DA7219_H */ diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c new file mode 100644 index 000000000000..ecc02449c569 --- /dev/null +++ b/sound/soc/codecs/es8316.c @@ -0,0 +1,637 @@ +/* + * es8316.c -- es8316 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua@everest-semi.com>, + * Daniel Drake <drake@endlessm.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/acpi.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/regmap.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8316.h" + +/* In slave mode at single speed, the codec is documented as accepting 5 + * MCLK/LRCK ratios, but we also add ratio 400, which is commonly used on + * Intel Cherry Trail platforms (19.2MHz MCLK, 48kHz LRCK). + */ +#define NR_SUPPORTED_MCLK_LRCK_RATIOS 6 +static const unsigned int supported_mclk_lrck_ratios[] = { + 256, 384, 400, 512, 768, 1024 +}; + +struct es8316_priv { + unsigned int sysclk; + unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS]; + struct snd_pcm_hw_constraint_list sysclk_constraints; +}; + +/* + * ES8316 controls + */ +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9600, 50, 1); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); + +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), + 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), + 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), +); + +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-4800, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(-2400, 1200, 0), +); + +static const char * const ng_type_txt[] = + { "Constant PGA Gain", "Mute ADC Output" }; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(ES8316_ADC_ALC_NG, 6, 2, ng_type_txt); + +static const char * const adcpol_txt[] = { "Normal", "Invert" }; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(ES8316_ADC_MUTE, 1, 2, adcpol_txt); +static const char *const dacpol_txt[] = + { "Normal", "R Invert", "L Invert", "L + R Invert" }; +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8316_DAC_SET1, 0, 4, dacpol_txt); + +static const struct snd_kcontrol_new es8316_snd_controls[] = { + SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, + 4, 0, 3, 1, hpout_vol_tlv), + SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, + 0, 4, 7, 0, hpmixer_gain_tlv), + + SOC_ENUM("Playback Polarity", dacpol), + SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, + ES8316_DAC_VOLR, 0, 0xc0, 1, dac_vol_tlv), + SOC_SINGLE("DAC Soft Ramp Switch", ES8316_DAC_SET1, 4, 1, 1), + SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0), + SOC_SINGLE("DAC Notch Filter Switch", ES8316_DAC_SET2, 6, 1, 0), + SOC_SINGLE("DAC Double Fs Switch", ES8316_DAC_SET2, 7, 1, 0), + SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0), + + SOC_ENUM("Capture Polarity", adcpol), + SOC_SINGLE("Mic Boost Switch", ES8316_ADC_D2SEPGA, 0, 1, 0), + SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME, + 0, 0xc0, 1, adc_vol_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8316_ADC_PGAGAIN, + 4, 10, 0, adc_pga_gain_tlv), + SOC_SINGLE("ADC Soft Ramp Switch", ES8316_ADC_MUTE, 4, 1, 0), + SOC_SINGLE("ADC Double Fs Switch", ES8316_ADC_DMIC, 4, 1, 0), + + SOC_SINGLE("ALC Capture Switch", ES8316_ADC_ALC1, 6, 1, 0), + SOC_SINGLE_TLV("ALC Capture Max Volume", ES8316_ADC_ALC1, 0, 28, 0, + alc_max_gain_tlv), + SOC_SINGLE_TLV("ALC Capture Min Volume", ES8316_ADC_ALC2, 0, 28, 0, + alc_min_gain_tlv), + SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 10, 0, + alc_target_tlv), + SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0), + SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack Time", ES8316_ADC_ALC4, 0, 10, 0), + SOC_SINGLE("ALC Capture Noise Gate Switch", ES8316_ADC_ALC_NG, + 5, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", ES8316_ADC_ALC_NG, + 0, 31, 0), + SOC_ENUM("ALC Capture Noise Gate Type", ng_type), +}; + +/* Analog Input Mux */ +static const char * const es8316_analog_in_txt[] = { + "lin1-rin1", + "lin2-rin2", + "lin1-rin1 with 20db Boost", + "lin2-rin2 with 20db Boost" +}; +static const unsigned int es8316_analog_in_values[] = { 0, 1, 2, 3 }; +static const struct soc_enum es8316_analog_input_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_PDN_LINSEL, 4, 3, + ARRAY_SIZE(es8316_analog_in_txt), + es8316_analog_in_txt, + es8316_analog_in_values); +static const struct snd_kcontrol_new es8316_analog_in_mux_controls = + SOC_DAPM_ENUM("Route", es8316_analog_input_enum); + +static const char * const es8316_dmic_txt[] = { + "dmic disable", + "dmic data at high level", + "dmic data at low level", +}; +static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const struct soc_enum es8316_dmic_src_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, + ARRAY_SIZE(es8316_dmic_txt), + es8316_dmic_txt, + es8316_dmic_values); +static const struct snd_kcontrol_new es8316_dmic_src_controls = + SOC_DAPM_ENUM("Route", es8316_dmic_src_enum); + +/* hp mixer mux */ +static const char * const es8316_hpmux_texts[] = { + "lin1-rin1", + "lin2-rin2", + "lin-rin with Boost", + "lin-rin with Boost and PGA" +}; + +static const unsigned int es8316_hpmux_values[] = { 0, 1, 2, 3 }; + +static SOC_ENUM_SINGLE_DECL(es8316_left_hpmux_enum, ES8316_HPMIX_SEL, + 4, es8316_hpmux_texts); + +static const struct snd_kcontrol_new es8316_left_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_left_hpmux_enum); + +static SOC_ENUM_SINGLE_DECL(es8316_right_hpmux_enum, ES8316_HPMIX_SEL, + 0, es8316_hpmux_texts); + +static const struct snd_kcontrol_new es8316_right_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_right_hpmux_enum); + +/* headphone Output Mixer */ +static const struct snd_kcontrol_new es8316_out_left_mix[] = { + SOC_DAPM_SINGLE("LLIN Switch", ES8316_HPMIX_SWITCH, 6, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", ES8316_HPMIX_SWITCH, 7, 1, 0), +}; +static const struct snd_kcontrol_new es8316_out_right_mix[] = { + SOC_DAPM_SINGLE("RLIN Switch", ES8316_HPMIX_SWITCH, 2, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", ES8316_HPMIX_SWITCH, 3, 1, 0), +}; + +/* DAC data source mux */ +static const char * const es8316_dacsrc_texts[] = { + "LDATA TO LDAC, RDATA TO RDAC", + "LDATA TO LDAC, LDATA TO RDAC", + "RDATA TO LDAC, RDATA TO RDAC", + "RDATA TO LDAC, LDATA TO RDAC", +}; + +static const unsigned int es8316_dacsrc_values[] = { 0, 1, 2, 3 }; + +static SOC_ENUM_SINGLE_DECL(es8316_dacsrc_mux_enum, ES8316_DAC_SET1, + 6, es8316_dacsrc_texts); + +static const struct snd_kcontrol_new es8316_dacsrc_mux_controls = + SOC_DAPM_ENUM("Route", es8316_dacsrc_mux_enum); + +static const struct snd_soc_dapm_widget es8316_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("Bias", ES8316_SYS_PDN, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Analog power", ES8316_SYS_PDN, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", ES8316_SYS_PDN, 5, 1, NULL, 0), + + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + + /* Input Mux */ + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8316_analog_in_mux_controls), + + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8316_SYS_PDN, 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC bias", ES8316_SYS_PDN, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", ES8316_CLKMGR_CLKSW, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line input PGA", ES8316_ADC_PDN_LINSEL, + 7, 1, NULL, 0), + SND_SOC_DAPM_ADC("Mono ADC", NULL, ES8316_ADC_PDN_LINSEL, 6, 1), + SND_SOC_DAPM_MUX("Digital Mic Mux", SND_SOC_NOPM, 0, 0, + &es8316_dmic_src_controls), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 1, + ES8316_SERDATA_ADC, 6, 1), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Source Mux", SND_SOC_NOPM, 0, 0, + &es8316_dacsrc_mux_controls), + + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8316_SYS_PDN, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Clock", ES8316_CLKMGR_CLKSW, 2, 0, NULL, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8316_DAC_PDN, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8316_DAC_PDN, 4, 1), + + /* Headphone Output Side */ + SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &es8316_left_hpmux_controls), + SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &es8316_right_hpmux_controls), + SND_SOC_DAPM_MIXER("Left Headphone Mixer", ES8316_HPMIX_PDN, + 5, 1, &es8316_out_left_mix[0], + ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Headphone Mixer", ES8316_HPMIX_PDN, + 1, 1, &es8316_out_right_mix[0], + ARRAY_SIZE(es8316_out_right_mix)), + SND_SOC_DAPM_PGA("Left Headphone Mixer Out", ES8316_HPMIX_PDN, + 4, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Mixer Out", ES8316_HPMIX_PDN, + 0, 1, NULL, 0), + + SND_SOC_DAPM_OUT_DRV("Left Headphone Charge Pump", ES8316_CPHP_OUTEN, + 6, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Right Headphone Charge Pump", ES8316_CPHP_OUTEN, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8316_CPHP_PDN2, + 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump Clock", ES8316_CLKMGR_CLKSW, + 4, 0, NULL, 0), + + SND_SOC_DAPM_OUT_DRV("Left Headphone Driver", ES8316_CPHP_OUTEN, + 5, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Right Headphone Driver", ES8316_CPHP_OUTEN, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Out", ES8316_CPHP_PDN1, 2, 1, NULL, 0), + + /* pdn_Lical and pdn_Rical bits are documented as Reserved, but must + * be explicitly unset in order to enable HP output + */ + SND_SOC_DAPM_SUPPLY("Left Headphone ical", ES8316_CPHP_ICAL_VOL, + 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right Headphone ical", ES8316_CPHP_ICAL_VOL, + 3, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8316_dapm_routes[] = { + /* Recording */ + {"MIC1", NULL, "Mic Bias"}, + {"MIC2", NULL, "Mic Bias"}, + {"MIC1", NULL, "Bias"}, + {"MIC2", NULL, "Bias"}, + {"MIC1", NULL, "Analog power"}, + {"MIC2", NULL, "Analog power"}, + + {"Differential Mux", "lin1-rin1", "MIC1"}, + {"Differential Mux", "lin2-rin2", "MIC2"}, + {"Line input PGA", NULL, "Differential Mux"}, + + {"Mono ADC", NULL, "ADC Clock"}, + {"Mono ADC", NULL, "ADC Vref"}, + {"Mono ADC", NULL, "ADC bias"}, + {"Mono ADC", NULL, "Line input PGA"}, + + /* It's not clear why, but to avoid recording only silence, + * the DAC clock must be running for the ADC to work. + */ + {"Mono ADC", NULL, "DAC Clock"}, + + {"Digital Mic Mux", "dmic disable", "Mono ADC"}, + + {"I2S OUT", NULL, "Digital Mic Mux"}, + + /* Playback */ + {"DAC Source Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + + {"Left DAC", NULL, "DAC Clock"}, + {"Right DAC", NULL, "DAC Clock"}, + + {"Left DAC", NULL, "DAC Vref"}, + {"Right DAC", NULL, "DAC Vref"}, + + {"Left DAC", NULL, "DAC Source Mux"}, + {"Right DAC", NULL, "DAC Source Mux"}, + + {"Left Headphone Mux", "lin-rin with Boost and PGA", "Line input PGA"}, + {"Right Headphone Mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Left Headphone Mixer", "LLIN Switch", "Left Headphone Mux"}, + {"Left Headphone Mixer", "Left DAC Switch", "Left DAC"}, + + {"Right Headphone Mixer", "RLIN Switch", "Right Headphone Mux"}, + {"Right Headphone Mixer", "Right DAC Switch", "Right DAC"}, + + {"Left Headphone Mixer Out", NULL, "Left Headphone Mixer"}, + {"Right Headphone Mixer Out", NULL, "Right Headphone Mixer"}, + + {"Left Headphone Charge Pump", NULL, "Left Headphone Mixer Out"}, + {"Right Headphone Charge Pump", NULL, "Right Headphone Mixer Out"}, + + {"Left Headphone Charge Pump", NULL, "Headphone Charge Pump"}, + {"Right Headphone Charge Pump", NULL, "Headphone Charge Pump"}, + + {"Left Headphone Charge Pump", NULL, "Headphone Charge Pump Clock"}, + {"Right Headphone Charge Pump", NULL, "Headphone Charge Pump Clock"}, + + {"Left Headphone Driver", NULL, "Left Headphone Charge Pump"}, + {"Right Headphone Driver", NULL, "Right Headphone Charge Pump"}, + + {"HPOL", NULL, "Left Headphone Driver"}, + {"HPOR", NULL, "Right Headphone Driver"}, + + {"HPOL", NULL, "Left Headphone ical"}, + {"HPOR", NULL, "Right Headphone ical"}, + + {"Headphone Out", NULL, "Bias"}, + {"Headphone Out", NULL, "Analog power"}, + {"HPOL", NULL, "Headphone Out"}, + {"HPOR", NULL, "Headphone Out"}, +}; + +static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + int i; + int count = 0; + + es8316->sysclk = freq; + + if (freq == 0) + return 0; + + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (freq % ratio == 0) + es8316->allowed_rates[count++] = freq / ratio; + } + + es8316->sysclk_constraints.list = es8316->allowed_rates; + es8316->sysclk_constraints.count = count; + + return 0; +} + +static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 serdata1 = 0; + u8 serdata2 = 0; + u8 clksw; + u8 mask; + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Codec driver only supports slave mode\n"); + return -EINVAL; + } + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) { + dev_err(codec->dev, "Codec driver only supports I2S format\n"); + return -EINVAL; + } + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + serdata1 |= ES8316_SERDATA1_BCLK_INV; + serdata2 |= ES8316_SERDATA2_ADCLRP; + break; + case SND_SOC_DAIFMT_IB_NF: + serdata1 |= ES8316_SERDATA1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + serdata2 |= ES8316_SERDATA2_ADCLRP; + break; + default: + return -EINVAL; + } + + mask = ES8316_SERDATA1_MASTER | ES8316_SERDATA1_BCLK_INV; + snd_soc_update_bits(codec, ES8316_SERDATA1, mask, serdata1); + + mask = ES8316_SERDATA2_FMT_MASK | ES8316_SERDATA2_ADCLRP; + snd_soc_update_bits(codec, ES8316_SERDATA_ADC, mask, serdata2); + snd_soc_update_bits(codec, ES8316_SERDATA_DAC, mask, serdata2); + + /* Enable BCLK and MCLK inputs in slave mode */ + clksw = ES8316_CLKMGR_CLKSW_MCLK_ON | ES8316_CLKMGR_CLKSW_BCLK_ON; + snd_soc_update_bits(codec, ES8316_CLKMGR_CLKSW, clksw, clksw); + + return 0; +} + +static int es8316_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + + if (es8316->sysclk == 0) { + dev_err(codec->dev, "No sysclk provided\n"); + return -EINVAL; + } + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC. + */ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &es8316->sysclk_constraints); + + return 0; +} + +static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + u8 wordlen = 0; + + if (!es8316->sysclk) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wordlen = ES8316_SERDATA2_LEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wordlen = ES8316_SERDATA2_LEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wordlen = ES8316_SERDATA2_LEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wordlen = ES8316_SERDATA2_LEN_32; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ES8316_SERDATA_DAC, + ES8316_SERDATA2_LEN_MASK, wordlen); + snd_soc_update_bits(codec, ES8316_SERDATA_ADC, + ES8316_SERDATA2_LEN_MASK, wordlen); + return 0; +} + +static int es8316_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ES8316_DAC_SET1, 0x20, + mute ? 0x20 : 0); + return 0; +} + +#define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops es8316_ops = { + .startup = es8316_pcm_startup, + .hw_params = es8316_pcm_hw_params, + .set_fmt = es8316_set_dai_fmt, + .set_sysclk = es8316_set_dai_sysclk, + .digital_mute = es8316_mute, +}; + +static struct snd_soc_dai_driver es8316_dai = { + .name = "ES8316 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ES8316_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ES8316_FORMATS, + }, + .ops = &es8316_ops, + .symmetric_rates = 1, +}; + +static int es8316_probe(struct snd_soc_codec *codec) +{ + /* Reset codec and enable current state machine */ + snd_soc_write(codec, ES8316_RESET, 0x3f); + usleep_range(5000, 5500); + snd_soc_write(codec, ES8316_RESET, ES8316_RESET_CSM_ON); + msleep(30); + + /* + * Documentation is unclear, but this value from the vendor driver is + * needed otherwise audio output is silent. + */ + snd_soc_write(codec, ES8316_SYS_VMIDSEL, 0xff); + + /* + * Documentation for this register is unclear and incomplete, + * but here is a vendor-provided value that improves volume + * and quality for Intel CHT platforms. + */ + snd_soc_write(codec, ES8316_CLKMGR_ADCOSR, 0x32); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_es8316 = { + .probe = es8316_probe, + .idle_bias_off = true, + + .component_driver = { + .controls = es8316_snd_controls, + .num_controls = ARRAY_SIZE(es8316_snd_controls), + .dapm_widgets = es8316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8316_dapm_widgets), + .dapm_routes = es8316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes), + }, +}; + +static const struct regmap_config es8316_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x53, + .cache_type = REGCACHE_RBTREE, +}; + +static int es8316_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct es8316_priv *es8316; + struct regmap *regmap; + + es8316 = devm_kzalloc(&i2c_client->dev, sizeof(struct es8316_priv), + GFP_KERNEL); + if (es8316 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, es8316); + + regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_es8316, + &es8316_dai, 1); +} + +static int es8316_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id es8316_i2c_id[] = { + {"es8316", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8316_i2c_id); + +static const struct of_device_id es8316_of_match[] = { + { .compatible = "everest,es8316", }, + {}, +}; +MODULE_DEVICE_TABLE(of, es8316_of_match); + +static const struct acpi_device_id es8316_acpi_match[] = { + {"ESSX8316", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8316_acpi_match); + +static struct i2c_driver es8316_i2c_driver = { + .driver = { + .name = "es8316", + .acpi_match_table = ACPI_PTR(es8316_acpi_match), + .of_match_table = of_match_ptr(es8316_of_match), + }, + .probe = es8316_i2c_probe, + .remove = es8316_i2c_remove, + .id_table = es8316_i2c_id, +}; +module_i2c_driver(es8316_i2c_driver); + +MODULE_DESCRIPTION("Everest Semi ES8316 ALSA SoC Codec Driver"); +MODULE_AUTHOR("David Yang <yangxiaohua@everest-semi.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h new file mode 100644 index 000000000000..6bcdd63ea459 --- /dev/null +++ b/sound/soc/codecs/es8316.h @@ -0,0 +1,129 @@ +/* + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang <yangxiaohua@everest-semi.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 _ES8316_H +#define _ES8316_H + +/* + * ES8316 register space + */ + +/* Reset Control */ +#define ES8316_RESET 0x00 + +/* Clock Management */ +#define ES8316_CLKMGR_CLKSW 0x01 +#define ES8316_CLKMGR_CLKSEL 0x02 +#define ES8316_CLKMGR_ADCOSR 0x03 +#define ES8316_CLKMGR_ADCDIV1 0x04 +#define ES8316_CLKMGR_ADCDIV2 0x05 +#define ES8316_CLKMGR_DACDIV1 0x06 +#define ES8316_CLKMGR_DACDIV2 0x07 +#define ES8316_CLKMGR_CPDIV 0x08 + +/* Serial Data Port Control */ +#define ES8316_SERDATA1 0x09 +#define ES8316_SERDATA_ADC 0x0a +#define ES8316_SERDATA_DAC 0x0b + +/* System Control */ +#define ES8316_SYS_VMIDSEL 0x0c +#define ES8316_SYS_PDN 0x0d +#define ES8316_SYS_LP1 0x0e +#define ES8316_SYS_LP2 0x0f +#define ES8316_SYS_VMIDLOW 0x10 +#define ES8316_SYS_VSEL 0x11 +#define ES8316_SYS_REF 0x12 + +/* Headphone Mixer */ +#define ES8316_HPMIX_SEL 0x13 +#define ES8316_HPMIX_SWITCH 0x14 +#define ES8316_HPMIX_PDN 0x15 +#define ES8316_HPMIX_VOL 0x16 + +/* Charge Pump Headphone driver */ +#define ES8316_CPHP_OUTEN 0x17 +#define ES8316_CPHP_ICAL_VOL 0x18 +#define ES8316_CPHP_PDN1 0x19 +#define ES8316_CPHP_PDN2 0x1a +#define ES8316_CPHP_LDOCTL 0x1b + +/* Calibration */ +#define ES8316_CAL_TYPE 0x1c +#define ES8316_CAL_SET 0x1d +#define ES8316_CAL_HPLIV 0x1e +#define ES8316_CAL_HPRIV 0x1f +#define ES8316_CAL_HPLMV 0x20 +#define ES8316_CAL_HPRMV 0x21 + +/* ADC Control */ +#define ES8316_ADC_PDN_LINSEL 0x22 +#define ES8316_ADC_PGAGAIN 0x23 +#define ES8316_ADC_D2SEPGA 0x24 +#define ES8316_ADC_DMIC 0x25 +#define ES8316_ADC_MUTE 0x26 +#define ES8316_ADC_VOLUME 0x27 +#define ES8316_ADC_ALC1 0x29 +#define ES8316_ADC_ALC2 0x2a +#define ES8316_ADC_ALC3 0x2b +#define ES8316_ADC_ALC4 0x2c +#define ES8316_ADC_ALC5 0x2d +#define ES8316_ADC_ALC_NG 0x2e + +/* DAC Control */ +#define ES8316_DAC_PDN 0x2f +#define ES8316_DAC_SET1 0x30 +#define ES8316_DAC_SET2 0x31 +#define ES8316_DAC_SET3 0x32 +#define ES8316_DAC_VOLL 0x33 +#define ES8316_DAC_VOLR 0x34 + +/* GPIO */ +#define ES8316_GPIO_SEL 0x4d +#define ES8316_GPIO_DEBOUNCE 0x4e +#define ES8316_GPIO_FLAG 0x4f + +/* Test mode */ +#define ES8316_TESTMODE 0x50 +#define ES8316_TEST1 0x51 +#define ES8316_TEST2 0x52 +#define ES8316_TEST3 0x53 + +/* + * Field definitions + */ + +/* ES8316_RESET */ +#define ES8316_RESET_CSM_ON 0x80 + +/* ES8316_CLKMGR_CLKSW */ +#define ES8316_CLKMGR_CLKSW_MCLK_ON 0x40 +#define ES8316_CLKMGR_CLKSW_BCLK_ON 0x20 + +/* ES8316_SERDATA1 */ +#define ES8316_SERDATA1_MASTER 0x80 +#define ES8316_SERDATA1_BCLK_INV 0x20 + +/* ES8316_SERDATA_ADC and _DAC */ +#define ES8316_SERDATA2_FMT_MASK 0x3 +#define ES8316_SERDATA2_FMT_I2S 0x00 +#define ES8316_SERDATA2_FMT_LEFTJ 0x01 +#define ES8316_SERDATA2_FMT_RIGHTJ 0x02 +#define ES8316_SERDATA2_FMT_PCM 0x03 +#define ES8316_SERDATA2_ADCLRP 0x20 +#define ES8316_SERDATA2_LEN_MASK 0x1c +#define ES8316_SERDATA2_LEN_24 0x00 +#define ES8316_SERDATA2_LEN_20 0x04 +#define ES8316_SERDATA2_LEN_18 0x08 +#define ES8316_SERDATA2_LEN_16 0x0c +#define ES8316_SERDATA2_LEN_32 0x10 + +#endif diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8c5ae1fc23a9..22ed0dc88f0a 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -25,17 +25,6 @@ #include <drm/drm_crtc.h> /* This is only to get MAX_ELD_BYTES */ -struct hdmi_device { - struct device *dev; - struct list_head list; - int cnt; -}; -#define pos_to_hdmi_device(pos) container_of((pos), struct hdmi_device, list) -LIST_HEAD(hdmi_device_list); -static DEFINE_MUTEX(hdmi_mutex); - -#define DAI_NAME_SIZE 16 - #define HDMI_CODEC_CHMAP_IDX_UNKNOWN -1 struct hdmi_codec_channel_map_table { @@ -293,7 +282,6 @@ struct hdmi_codec_priv { struct hdmi_codec_daifmt daifmt[2]; struct mutex current_stream_lock; struct snd_pcm_substream *current_stream; - struct snd_pcm_hw_constraint_list ratec; uint8_t eld[MAX_ELD_BYTES]; struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; @@ -702,6 +690,7 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, } static struct snd_soc_dai_driver hdmi_i2s_dai = { + .name = "i2s-hifi", .id = DAI_ID_I2S, .playback = { .stream_name = "Playback", @@ -716,6 +705,7 @@ static struct snd_soc_dai_driver hdmi_i2s_dai = { }; static const struct snd_soc_dai_driver hdmi_spdif_dai = { + .name = "spdif-hifi", .id = DAI_ID_SPDIF, .playback = { .stream_name = "Playback", @@ -728,30 +718,16 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { .pcm_new = hdmi_codec_pcm_new, }; -static char hdmi_dai_name[][DAI_NAME_SIZE] = { - "hdmi-hifi.0", - "hdmi-hifi.1", - "hdmi-hifi.2", - "hdmi-hifi.3", -}; - -static int hdmi_of_xlate_dai_name(struct snd_soc_component *component, - struct of_phandle_args *args, - const char **dai_name) +static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) { - int id; + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + int ret = -ENOTSUPP; /* see snd_soc_get_dai_id() */ - if (args->args_count) - id = args->args[0]; - else - id = 0; + if (hcp->hcd.ops->get_dai_id) + ret = hcp->hcd.ops->get_dai_id(component, endpoint); - if (id < ARRAY_SIZE(hdmi_dai_name)) { - *dai_name = hdmi_dai_name[id]; - return 0; - } - - return -EAGAIN; + return ret; } static struct snd_soc_codec_driver hdmi_codec = { @@ -762,7 +738,7 @@ static struct snd_soc_codec_driver hdmi_codec = { .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, .num_dapm_routes = ARRAY_SIZE(hdmi_routes), - .of_xlate_dai_name = hdmi_of_xlate_dai_name, + .of_xlate_dai_id = hdmi_of_xlate_dai_id, }, }; @@ -771,8 +747,6 @@ static int hdmi_codec_probe(struct platform_device *pdev) struct hdmi_codec_pdata *hcd = pdev->dev.platform_data; struct device *dev = &pdev->dev; struct hdmi_codec_priv *hcp; - struct hdmi_device *hd; - struct list_head *pos; int dai_count, i = 0; int ret; @@ -794,35 +768,6 @@ static int hdmi_codec_probe(struct platform_device *pdev) if (!hcp) return -ENOMEM; - hd = NULL; - mutex_lock(&hdmi_mutex); - list_for_each(pos, &hdmi_device_list) { - struct hdmi_device *tmp = pos_to_hdmi_device(pos); - - if (tmp->dev == dev->parent) { - hd = tmp; - break; - } - } - - if (!hd) { - hd = devm_kzalloc(dev, sizeof(*hd), GFP_KERNEL); - if (!hd) { - mutex_unlock(&hdmi_mutex); - return -ENOMEM; - } - - hd->dev = dev->parent; - - list_add_tail(&hd->list, &hdmi_device_list); - } - mutex_unlock(&hdmi_mutex); - - if (hd->cnt >= ARRAY_SIZE(hdmi_dai_name)) { - dev_err(dev, "too many hdmi codec are deteced\n"); - return -EINVAL; - } - hcp->hcd = *hcd; mutex_init(&hcp->current_stream_lock); @@ -835,14 +780,11 @@ static int hdmi_codec_probe(struct platform_device *pdev) hcp->daidrv[i] = hdmi_i2s_dai; hcp->daidrv[i].playback.channels_max = hcd->max_i2s_channels; - hcp->daidrv[i].name = hdmi_dai_name[hd->cnt++]; i++; } - if (hcd->spdif) { + if (hcd->spdif) hcp->daidrv[i] = hdmi_spdif_dai; - hcp->daidrv[i].name = hdmi_dai_name[hd->cnt++]; - } ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv, dai_count); @@ -859,20 +801,8 @@ static int hdmi_codec_probe(struct platform_device *pdev) static int hdmi_codec_remove(struct platform_device *pdev) { struct device *dev = &pdev->dev; - struct list_head *pos; struct hdmi_codec_priv *hcp; - mutex_lock(&hdmi_mutex); - list_for_each(pos, &hdmi_device_list) { - struct hdmi_device *tmp = pos_to_hdmi_device(pos); - - if (tmp->dev == dev->parent) { - list_del(pos); - break; - } - } - mutex_unlock(&hdmi_mutex); - hcp = dev_get_drvdata(dev); kfree(hcp->chmap_info); snd_soc_unregister_codec(dev); diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 0247edc9c84e..2a40a69a7513 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -132,7 +132,7 @@ enum rates { pcm_rate_48, max_pcm_rate, }; -struct ni_div_rates { +static const struct ni_div_rates { u32 mclk; u16 ni[max_pcm_rate]; } ni_div[] = { diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index d8e8590746af..a78802920c3c 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -223,8 +223,8 @@ struct pm8916_wcd_analog_priv { u16 codec_version; struct clk *mclk; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; - bool micbias1_cap_mode; - bool micbias2_cap_mode; + unsigned int micbias1_cap_mode; + unsigned int micbias2_cap_mode; }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; @@ -285,7 +285,7 @@ static void pm8916_wcd_analog_micbias_enable(struct snd_soc_codec *codec) static int pm8916_wcd_analog_enable_micbias_ext(struct snd_soc_codec *codec, int event, - int reg, u32 cap_mode) + int reg, unsigned int cap_mode) { switch (event) { case SND_SOC_DAPM_POST_PMU: diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cca974d26136..3a309b18035e 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1125,6 +1125,57 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /** + * nau8824_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @tx_mask: Bitmask representing active TX slots. Ex. + * 0xf for normal 4 channel TDM. + * 0xf0 for shifted 4 channel TDM + * @rx_mask: Bitmask [0:1] representing active DACR RX slots. + * Bitmask [2:3] representing active DACL RX slots. + * 00=CH0,01=CH1,10=CH2,11=CH3. Ex. + * 0xf for DACL/R selecting TDM CH3. + * 0xf0 for DACL/R selecting shifted TDM CH3. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * Configures a DAI for TDM operation. Only support 4 slots TDM. + */ +static int nau8824_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 nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + unsigned int tslot_l = 0, ctrl_val = 0; + + if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf0) && (rx_mask & 0xf)) || + ((rx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf) && (tx_mask & 0xf0))) + return -EINVAL; + + ctrl_val |= (NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN); + if (tx_mask & 0xf0) { + tslot_l = 4 * slot_width; + ctrl_val |= (tx_mask >> 4); + } else { + ctrl_val |= tx_mask; + } + if (rx_mask & 0xf0) + ctrl_val |= ((rx_mask >> 4) << NAU8824_TDM_DACR_RX_SFT); + else + ctrl_val |= (rx_mask << NAU8824_TDM_DACR_RX_SFT); + + regmap_update_bits(nau8824->regmap, NAU8824_REG_TDM_CTRL, + NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN | + NAU8824_TDM_DACL_RX_MASK | NAU8824_TDM_DACR_RX_MASK | + NAU8824_TDM_TX_MASK, ctrl_val); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_LEFT_TIME_SLOT, + NAU8824_TSLOT_L_MASK, tslot_l); + + return 0; +} + +/** * nau8824_calc_fll_param - Calculate FLL parameters. * @fll_in: external clock provided to codec. * @fs: sampling rate. @@ -1440,6 +1491,7 @@ static struct snd_soc_codec_driver nau8824_codec_driver = { static const struct snd_soc_dai_ops nau8824_dai_ops = { .hw_params = nau8824_hw_params, .set_fmt = nau8824_set_fmt, + .set_tdm_slot = nau8824_set_tdm_slot, }; #define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h index 87ac9a382aed..21eae2431c83 100644 --- a/sound/soc/codecs/nau8824.h +++ b/sound/soc/codecs/nau8824.h @@ -258,6 +258,18 @@ #define NAU8824_I2S_MS_SLAVE (0 << NAU8824_I2S_MS_SFT) #define NAU8824_I2S_BLK_DIV_MASK 0x7 +/* PORT0_LEFT_TIME_SLOT (0x1E) */ +#define NAU8824_TSLOT_L_MASK 0x3ff + +/* TDM_CTRL (0x20) */ +#define NAU8824_TDM_MODE (0x1 << 15) +#define NAU8824_TDM_OFFSET_EN (0x1 << 14) +#define NAU8824_TDM_DACL_RX_SFT 6 +#define NAU8824_TDM_DACL_RX_MASK (0x3 << NAU8824_TDM_DACL_RX_SFT) +#define NAU8824_TDM_DACR_RX_SFT 4 +#define NAU8824_TDM_DACR_RX_MASK (0x3 << NAU8824_TDM_DACR_RX_SFT) +#define NAU8824_TDM_TX_MASK 0xf + /* ADC_FILTER_CTRL (0x24) */ #define NAU8824_ADC_SYNC_DOWN_MASK 0x3 #define NAU8824_ADC_SYNC_DOWN_32 0 diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 97fbeba9498f..46a30eaa7ace 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1612,7 +1612,6 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_sync(dapm); break; case 2: - case 3: dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); type = SND_JACK_HEADSET; @@ -1632,6 +1631,11 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); break; + case 3: + /* detect error case */ + dev_err(nau8825->dev, "detection error; disable mic function\n"); + type = SND_JACK_HEADPHONE; + break; } /* Leaving HPOL/R grounded after jack insert by default. They will be @@ -1682,7 +1686,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); - if (!nau8825->high_imped) { + if (!nau8825->xtalk_bypass && !nau8825->high_imped) { /* Apply the cross talk suppression in the * headset without high impedance. */ @@ -2328,6 +2332,13 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + /* Reset the configuration of jack type for detection */ + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_HSD_CTRL, 0xf, 0xf); /* Cancel and reset cross talk detection funciton */ nau8825_xtalk_cancel(nau8825); /* Turn off all interruptions before system shutdown. Keep the @@ -2351,6 +2362,10 @@ static int __maybe_unused nau8825_suspend(struct snd_soc_codec *codec) disable_irq(nau8825->irq); snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + /* Power down codec power; don't suppoet button wakeup */ + snd_soc_dapm_disable_pin(nau8825->dapm, "SAR"); + snd_soc_dapm_disable_pin(nau8825->dapm, "MICBIAS"); + snd_soc_dapm_sync(nau8825->dapm); regcache_cache_only(nau8825->regmap, true); regcache_mark_dirty(nau8825->regmap); @@ -2425,10 +2440,13 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_insert_debounce); dev_dbg(dev, "jack-eject-debounce: %d\n", nau8825->jack_eject_debounce); + dev_dbg(dev, "crosstalk-bypass: %d\n", + nau8825->xtalk_bypass); } static int nau8825_read_device_properties(struct device *dev, struct nau8825 *nau8825) { + int ret; nau8825->jkdet_enable = device_property_read_bool(dev, "nuvoton,jkdet-enable"); @@ -2436,30 +2454,60 @@ static int nau8825_read_device_properties(struct device *dev, "nuvoton,jkdet-pull-enable"); nau8825->jkdet_pull_up = device_property_read_bool(dev, "nuvoton,jkdet-pull-up"); - device_property_read_u32(dev, "nuvoton,jkdet-polarity", + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", &nau8825->jkdet_polarity); - device_property_read_u32(dev, "nuvoton,micbias-voltage", + if (ret) + nau8825->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", &nau8825->micbias_voltage); - device_property_read_u32(dev, "nuvoton,vref-impedance", + if (ret) + nau8825->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", &nau8825->vref_impedance); - device_property_read_u32(dev, "nuvoton,sar-threshold-num", + if (ret) + nau8825->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,sar-threshold-num", &nau8825->sar_threshold_num); - device_property_read_u32_array(dev, "nuvoton,sar-threshold", + if (ret) + nau8825->sar_threshold_num = 4; + ret = device_property_read_u32_array(dev, "nuvoton,sar-threshold", nau8825->sar_threshold, nau8825->sar_threshold_num); - device_property_read_u32(dev, "nuvoton,sar-hysteresis", + if (ret) { + nau8825->sar_threshold[0] = 0x08; + nau8825->sar_threshold[1] = 0x12; + nau8825->sar_threshold[2] = 0x26; + nau8825->sar_threshold[3] = 0x73; + } + ret = device_property_read_u32(dev, "nuvoton,sar-hysteresis", &nau8825->sar_hysteresis); - device_property_read_u32(dev, "nuvoton,sar-voltage", + if (ret) + nau8825->sar_hysteresis = 0; + ret = device_property_read_u32(dev, "nuvoton,sar-voltage", &nau8825->sar_voltage); - device_property_read_u32(dev, "nuvoton,sar-compare-time", + if (ret) + nau8825->sar_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,sar-compare-time", &nau8825->sar_compare_time); - device_property_read_u32(dev, "nuvoton,sar-sampling-time", + if (ret) + nau8825->sar_compare_time = 1; + ret = device_property_read_u32(dev, "nuvoton,sar-sampling-time", &nau8825->sar_sampling_time); - device_property_read_u32(dev, "nuvoton,short-key-debounce", + if (ret) + nau8825->sar_sampling_time = 1; + ret = device_property_read_u32(dev, "nuvoton,short-key-debounce", &nau8825->key_debounce); - device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + if (ret) + nau8825->key_debounce = 3; + ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce", &nau8825->jack_insert_debounce); - device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + if (ret) + nau8825->jack_insert_debounce = 7; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", &nau8825->jack_eject_debounce); + if (ret) + nau8825->jack_eject_debounce = 0; + nau8825->xtalk_bypass = device_property_read_bool(dev, + "nuvoton,crosstalk-bypass"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 514fd13c2f46..8aee5c8647ae 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -476,6 +476,7 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; + int xtalk_bypass; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b1ddf0..1b6796c4c471 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -9,6 +9,7 @@ * published by the Free Software Foundation. */ +#include <linux/acpi.h> #include <linux/fs.h> #include <linux/module.h> #include <linux/moduleparam.h> @@ -395,14 +396,14 @@ static const char * const rt5514_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo1_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD0_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); static const struct snd_kcontrol_new rt5514_sto1_dmic_mux = SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5514_stereo1_dmic_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo2_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD1_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); @@ -906,9 +907,23 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; - if (slots == 4) + switch (slots) { + case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; + break; + + case 6: + val |= RT5514_TDMSLOT_SEL_RX_6CH | RT5514_TDMSLOT_SEL_TX_6CH; + break; + case 8: + val |= RT5514_TDMSLOT_SEL_RX_8CH | RT5514_TDMSLOT_SEL_TX_8CH; + break; + + case 2: + default: + break; + } switch (slot_width) { case 20: @@ -919,6 +934,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, val |= RT5514_CH_LEN_RX_24 | RT5514_CH_LEN_TX_24; break; + case 25: + val |= RT5514_TDM_MODE2; + break; + case 32: val |= RT5514_CH_LEN_RX_32 | RT5514_CH_LEN_TX_32; break; @@ -930,7 +949,8 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_TDM_MODE | RT5514_TDMSLOT_SEL_RX_MASK | RT5514_TDMSLOT_SEL_TX_MASK | - RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK, val); + RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | + RT5514_TDM_MODE2, val); return 0; } @@ -1076,6 +1096,14 @@ static const struct of_device_id rt5514_of_match[] = { MODULE_DEVICE_TABLE(of, rt5514_of_match); #endif +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5514_acpi_match[] = { + { "10EC5514", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5514_acpi_match); +#endif + static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) { device_property_read_u32(dev, "realtek,dmic-init-delay-ms", @@ -1179,6 +1207,7 @@ static const struct dev_pm_ops rt5514_i2_pm_ops = { static struct i2c_driver rt5514_i2c_driver = { .driver = { .name = "rt5514", + .acpi_match_table = ACPI_PTR(rt5514_acpi_match), .of_match_table = of_match_ptr(rt5514_of_match), .pm = &rt5514_i2_pm_ops, }, diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 5d343fb6d125..02bc212a86d9 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -117,6 +117,8 @@ #define RT5514_POW_ADCFEDL_BIT 0 /* RT5514_I2S_CTRL1 (0x2010) */ +#define RT5514_TDM_MODE2 (0x1 << 30) +#define RT5514_TDM_MODE2_SFT 30 #define RT5514_TDM_MODE (0x1 << 28) #define RT5514_TDM_MODE_SFT 28 #define RT5514_I2S_LR_MASK (0x1 << 26) @@ -136,6 +138,8 @@ #define RT5514_TDMSLOT_SEL_RX_MASK (0x3 << 10) #define RT5514_TDMSLOT_SEL_RX_SFT 10 #define RT5514_TDMSLOT_SEL_RX_4CH (0x1 << 10) +#define RT5514_TDMSLOT_SEL_RX_6CH (0x2 << 10) +#define RT5514_TDMSLOT_SEL_RX_8CH (0x3 << 10) #define RT5514_CH_LEN_RX_MASK (0x3 << 8) #define RT5514_CH_LEN_RX_SFT 8 #define RT5514_CH_LEN_RX_16 (0x0 << 8) @@ -145,6 +149,8 @@ #define RT5514_TDMSLOT_SEL_TX_MASK (0x3 << 6) #define RT5514_TDMSLOT_SEL_TX_SFT 6 #define RT5514_TDMSLOT_SEL_TX_4CH (0x1 << 6) +#define RT5514_TDMSLOT_SEL_TX_6CH (0x2 << 6) +#define RT5514_TDMSLOT_SEL_TX_8CH (0x3 << 6) #define RT5514_CH_LEN_TX_MASK (0x3 << 4) #define RT5514_CH_LEN_TX_SFT 4 #define RT5514_CH_LEN_TX_16 (0x0 << 4) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 87844a45886a..9ec58166f7c4 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -34,6 +34,17 @@ #include "rl6231.h" #include "rt5645.h" +#define QUIRK_INV_JD1_1(q) ((q) & 1) +#define QUIRK_LEVEL_IRQ(q) (((q) >> 1) & 1) +#define QUIRK_IN2_DIFF(q) (((q) >> 2) & 1) +#define QUIRK_JD_MODE(q) (((q) >> 4) & 7) +#define QUIRK_DMIC1_DATA_PIN(q) (((q) >> 8) & 3) +#define QUIRK_DMIC2_DATA_PIN(q) (((q) >> 12) & 3) + +static unsigned int quirk = -1; +module_param(quirk, uint, 0444); +MODULE_PARM_DESC(quirk, "RT5645 pdata quirk override"); + #define RT5645_DEVICE_ID 0x6308 #define RT5650_DEVICE_ID 0x6419 @@ -59,7 +70,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, - {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x1c, 0xfd70}, {RT5645_PR_BASE + 0x20, 0x611f}, {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, @@ -3151,7 +3162,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } - if (rt5645->pdata.jd_invert) + if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); } else { /* jack out */ @@ -3172,7 +3183,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_disable_pin(dapm, "LDO2"); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); - if (rt5645->pdata.jd_invert) + if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } @@ -3238,24 +3249,16 @@ static void rt5645_jack_detect_work(struct work_struct *work) snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); return; - case 1: /* 2 port */ - val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0070; - break; - default: /* 1 port */ - val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0020; + default: /* read rt5645 jd1_1 status */ + val = snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x1000; break; } - switch (val) { - /* jack in */ - case 0x30: /* 2 port */ - case 0x0: /* 1 port or 2 port */ - if (rt5645->jack_type == 0) { - report = rt5645_jack_detect(rt5645->codec, 1); - /* for push button and jack out */ - break; - } + if (!val && (rt5645->jack_type == 0)) { /* jack in */ + report = rt5645_jack_detect(rt5645->codec, 1); + } else if (!val && rt5645->jack_type != 0) { + /* for push button and jack out */ btn_type = 0; if (snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x4) { /* button pressed */ @@ -3302,19 +3305,12 @@ static void rt5645_jack_detect_work(struct work_struct *work) mod_timer(&rt5645->btn_check_timer, msecs_to_jiffies(100)); } - - break; - /* jack out */ - case 0x70: /* 2 port */ - case 0x10: /* 2 port */ - case 0x20: /* 1 port */ + } else { + /* jack out */ report = 0; snd_soc_update_bits(rt5645->codec, RT5645_INT_IRQ_ST, 0x1, 0x0); rt5645_jack_detect(rt5645->codec, 0); - break; - default: - break; } snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); @@ -3601,7 +3597,7 @@ static struct rt5645_platform_data buddy_platform_data = { .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, - .jd_invert = true, + .level_trigger_irq = true, }; static struct dmi_system_id dmi_platform_intel_broadwell[] = { @@ -3614,6 +3610,33 @@ static struct dmi_system_id dmi_platform_intel_broadwell[] = { { } }; +static struct rt5645_platform_data gpd_win_platform_data = { + .jd_mode = 3, + .inv_jd1_1 = true, +}; + +static const struct dmi_system_id dmi_platform_gpd_win[] = { + { + /* + * Match for the GPDwin which unfortunately uses somewhat + * generic dmi strings, which is why we test for 4 strings. + * Comparing against 23 other byt/cht boards, board_vendor + * and board_name are unique to the GPDwin, where as only one + * other board has the same board_serial and 3 others have + * the same default product_name. Also the GPDwin is the + * only device to have both board_ and product_name not set. + */ + .ident = "GPD Win", + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Default string"), + DMI_MATCH(DMI_BOARD_SERIAL, "Default string"), + DMI_MATCH(DMI_PRODUCT_NAME, "Default string"), + }, + }, + {} +}; + static bool rt5645_check_dp(struct device *dev) { if (device_property_present(dev, "realtek,in2-differential") || @@ -3664,6 +3687,17 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, rt5645_parse_dt(rt5645, &i2c->dev); else if (dmi_check_system(dmi_platform_intel_braswell)) rt5645->pdata = general_platform_data; + else if (dmi_check_system(dmi_platform_gpd_win)) + rt5645->pdata = gpd_win_platform_data; + + if (quirk != -1) { + rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); + rt5645->pdata.level_trigger_irq = QUIRK_LEVEL_IRQ(quirk); + rt5645->pdata.inv_jd1_1 = QUIRK_INV_JD1_1(quirk); + rt5645->pdata.jd_mode = QUIRK_JD_MODE(quirk); + rt5645->pdata.dmic1_data_pin = QUIRK_DMIC1_DATA_PIN(quirk); + rt5645->pdata.dmic2_data_pin = QUIRK_DMIC2_DATA_PIN(quirk); + } rt5645->gpiod_hp_det = devm_gpiod_get_optional(&i2c->dev, "hp-detect", GPIOD_IN); @@ -3745,6 +3779,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret); } + regmap_update_bits(rt5645->regmap, RT5645_CLSD_OUT_CTRL, 0xc0, 0xc0); + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); @@ -3848,12 +3884,16 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, default: break; } + if (rt5645->pdata.inv_jd1_1) { + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); + } } regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2); - if (rt5645->pdata.jd_invert) { + if (rt5645->pdata.level_trigger_irq) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } @@ -3897,6 +3937,7 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); + del_timer_sync(&rt5645->btn_check_timer); snd_soc_unregister_codec(&i2c->dev); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index f5d34153e21f..db05b60d5002 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -586,44 +586,6 @@ static const struct snd_kcontrol_new hpo_r_mute_control = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL, RT5651_R_MUTE_SFT, 1, 1); -/* INL/R source */ -static const char * const rt5651_inl_src[] = {"IN2P", "HPOVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl_enum, RT5651_INL1_INR1_VOL, - RT5651_INL_SEL_SFT, rt5651_inl_src); - -static const struct snd_kcontrol_new rt5651_inl1_mux = - SOC_DAPM_ENUM("INL1 source", rt5651_inl_enum); - -static const char * const rt5651_inr1_src[] = {"IN2N", "HPOVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr1_enum, RT5651_INL1_INR1_VOL, - RT5651_INR_SEL_SFT, rt5651_inr1_src); - -static const struct snd_kcontrol_new rt5651_inr1_mux = - SOC_DAPM_ENUM("INR1 source", rt5651_inr1_enum); - -static const char * const rt5651_inl2_src[] = {"IN3P", "OUTVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl2_enum, RT5651_INL2_INR2_VOL, - RT5651_INL_SEL_SFT, rt5651_inl2_src); - -static const struct snd_kcontrol_new rt5651_inl2_mux = - SOC_DAPM_ENUM("INL2 source", rt5651_inl2_enum); - -static const char * const rt5651_inr2_src[] = {"IN3N", "OUTVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr2_enum, RT5651_INL2_INR2_VOL, - RT5651_INR_SEL_SFT, rt5651_inr2_src); - -static const struct snd_kcontrol_new rt5651_inr2_mux = - SOC_DAPM_ENUM("INR2 source", rt5651_inr2_enum); - - /* Stereo ADC source */ static const char * const rt5651_stereo1_adc1_src[] = {"DD MIX", "ADC"}; @@ -955,11 +917,7 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { RT5651_PWR_IN2_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, RT5651_PWR_IN2_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), - SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), - SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), - SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), + /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index a32508d7dcfd..a33202affeb1 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2847,6 +2847,8 @@ static int rt5663_resume(struct snd_soc_codec *codec) regcache_cache_only(rt5663->regmap, false); regcache_sync(rt5663->regmap); + rt5663_irq(0, rt5663); + return 0; } #else @@ -3141,7 +3143,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_DIG_MISC, RT5663_DIG_GATE_CTRL_MASK, RT5663_DIG_GATE_CTRL_EN); regmap_update_bits(rt5663->regmap, RT5663_AUTO_1MRC_CLK, - RT5663_IRQ_POW_SAV_MASK, RT5663_IRQ_POW_SAV_EN); + RT5663_IRQ_MANUAL_MASK, RT5663_IRQ_MANUAL_EN); regmap_update_bits(rt5663->regmap, RT5663_IRQ_1, RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN); regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h index d77fae619f2f..4621812c94d8 100644 --- a/sound/soc/codecs/rt5663.h +++ b/sound/soc/codecs/rt5663.h @@ -590,6 +590,10 @@ #define RT5663_IRQ_POW_SAV_JD1_SHIFT 14 #define RT5663_IRQ_POW_SAV_JD1_DIS (0x0 << 14) #define RT5663_IRQ_POW_SAV_JD1_EN (0x1 << 14) +#define RT5663_IRQ_MANUAL_MASK (0x1 << 8) +#define RT5663_IRQ_MANUAL_SHIFT 8 +#define RT5663_IRQ_MANUAL_DIS (0x0 << 8) +#define RT5663_IRQ_MANUAL_EN (0x1 << 8) /* IRQ Control 1 (0x00b6) */ #define RT5663_EN_CB_JD_MASK (0x1 << 3) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 8cd22307f5b6..370ed54d1e15 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -70,6 +70,7 @@ struct rt5665_priv { int jack_type; int irq_work_delay_time; unsigned int sar_adc_value; + bool calibration_done; }; static const struct reg_default rt5665_reg[] = { @@ -912,46 +913,46 @@ static const char * const rt5665_data_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_01_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_1_01_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_1_DS_ADC_SLOT01_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_23_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_1_23_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_1_DS_ADC_SLOT23_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_45_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_1_45_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_1_DS_ADC_SLOT45_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_1_67_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_1_67_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_1_DS_ADC_SLOT67_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_2_01_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_2_01_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_2_DS_ADC_SLOT01_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_2_23_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_2_23_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_2_DS_ADC_SLOT23_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_2_45_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_2_45_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_2_DS_ADC_SLOT45_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if1_2_67_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if1_2_67_adc_enum, RT5665_TDM_CTRL_2, RT5665_I2S1_2_DS_ADC_SLOT67_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if2_1_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if2_1_dac_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_1_DAC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if2_1_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if2_1_adc_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_1_ADC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if2_2_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if2_2_dac_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_2_DAC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if2_2_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if2_2_adc_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_2_ADC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if3_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if3_dac_enum, RT5665_DIG_INF3_DATA, RT5665_IF3_DAC_SEL_SFT, rt5665_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5665_if3_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5665_if3_adc_enum, RT5665_DIG_INF3_DATA, RT5665_IF3_ADC_SEL_SFT, rt5665_data_select); static const struct snd_kcontrol_new rt5665_if1_1_01_adc_swap_mux = @@ -1305,6 +1306,11 @@ static void rt5665_jack_detect_handler(struct work_struct *work) usleep_range(10000, 15000); } + while (!rt5665->calibration_done) { + pr_debug("%s calibration not ready\n", __func__); + usleep_range(10000, 15000); + } + mutex_lock(&rt5665->calibrate_mutex); val = snd_soc_read(rt5665->codec, RT5665_AJD1_CTRL) & 0x0010; @@ -1819,14 +1825,14 @@ static const char * const rt5665_dac2_src[] = { "IF1 DAC2", "IF2_1 DAC", "IF2_2 DAC", "IF3 DAC", "Mono ADC MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_l2_enum, RT5665_DAC2_CTRL, RT5665_DAC_L2_SEL_SFT, rt5665_dac2_src); static const struct snd_kcontrol_new rt5665_dac_l2_mux = SOC_DAPM_ENUM("Digital DAC L2 Source", rt5665_dac_l2_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_r2_enum, RT5665_DAC2_CTRL, RT5665_DAC_R2_SEL_SFT, rt5665_dac2_src); @@ -1839,14 +1845,14 @@ static const char * const rt5665_dac3_src[] = { "IF1 DAC2", "IF2_1 DAC", "IF2_2 DAC", "IF3 DAC", "STO2 ADC MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_l3_enum, RT5665_DAC3_CTRL, RT5665_DAC_L3_SEL_SFT, rt5665_dac3_src); static const struct snd_kcontrol_new rt5665_dac_l3_mux = SOC_DAPM_ENUM("Digital DAC L3 Source", rt5665_dac_l3_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_r3_enum, RT5665_DAC3_CTRL, RT5665_DAC_R3_SEL_SFT, rt5665_dac3_src); @@ -1859,14 +1865,14 @@ static const char * const rt5665_sto1_adc1_src[] = { "DD Mux", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adc1l_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADC1L_SRC_SFT, rt5665_sto1_adc1_src); static const struct snd_kcontrol_new rt5665_sto1_adc1l_mux = SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5665_sto1_adc1l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adc1r_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADC1R_SRC_SFT, rt5665_sto1_adc1_src); @@ -1879,14 +1885,14 @@ static const char * const rt5665_sto1_adc_src[] = { "ADC1 L", "ADC1 R", "ADC2 L", "ADC2 R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adcl_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADCL_SRC_SFT, rt5665_sto1_adc_src); static const struct snd_kcontrol_new rt5665_sto1_adcl_mux = SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5665_sto1_adcl_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adcr_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADCR_SRC_SFT, rt5665_sto1_adc_src); @@ -1899,14 +1905,14 @@ static const char * const rt5665_sto1_adc2_src[] = { "DAC MIX", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adc2l_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADC2L_SRC_SFT, rt5665_sto1_adc2_src); static const struct snd_kcontrol_new rt5665_sto1_adc2l_mux = SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5665_sto1_adc2l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_adc2r_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_ADC2R_SRC_SFT, rt5665_sto1_adc2_src); @@ -1919,7 +1925,7 @@ static const char * const rt5665_sto1_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_dmic_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_DMIC_SRC_SFT, rt5665_sto1_dmic_src); @@ -1931,7 +1937,7 @@ static const char * const rt5665_sto1_dd_l_src[] = { "STO2 DAC", "MONO DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_dd_l_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_DD_L_SRC_SFT, rt5665_sto1_dd_l_src); @@ -1943,7 +1949,7 @@ static const char * const rt5665_sto1_dd_r_src[] = { "STO2 DAC", "MONO DAC", "AEC REF" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto1_dd_r_enum, RT5665_STO1_ADC_MIXER, RT5665_STO1_DD_R_SRC_SFT, rt5665_sto1_dd_r_src); @@ -1956,7 +1962,7 @@ static const char * const rt5665_mono_adc_l2_src[] = { "DAC MIXL", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_l2_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_L2_SRC_SFT, rt5665_mono_adc_l2_src); @@ -1970,7 +1976,7 @@ static const char * const rt5665_mono_adc_l1_src[] = { "DD Mux", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_l1_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_L1_SRC_SFT, rt5665_mono_adc_l1_src); @@ -1982,14 +1988,14 @@ static const char * const rt5665_mono_dd_src[] = { "STO2 DAC", "MONO DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_dd_l_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_DD_L_SRC_SFT, rt5665_mono_dd_src); static const struct snd_kcontrol_new rt5665_mono_dd_l_mux = SOC_DAPM_ENUM("Mono DD L Source", rt5665_mono_dd_l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_dd_r_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_DD_R_SRC_SFT, rt5665_mono_dd_src); @@ -2002,14 +2008,14 @@ static const char * const rt5665_mono_adc_src[] = { "ADC1 L", "ADC1 R", "ADC2 L", "ADC2 R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_l_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_L_SRC_SFT, rt5665_mono_adc_src); static const struct snd_kcontrol_new rt5665_mono_adc_l_mux = SOC_DAPM_ENUM("Mono ADC L Source", rt5665_mono_adc_l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adcr_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_R_SRC_SFT, rt5665_mono_adc_src); @@ -2022,7 +2028,7 @@ static const char * const rt5665_mono_dmic_l_src[] = { "DMIC1 L", "DMIC2 L" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_dmic_l_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_DMIC_L_SRC_SFT, rt5665_mono_dmic_l_src); @@ -2035,7 +2041,7 @@ static const char * const rt5665_mono_adc_r2_src[] = { "DAC MIXR", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_r2_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_R2_SRC_SFT, rt5665_mono_adc_r2_src); @@ -2048,7 +2054,7 @@ static const char * const rt5665_mono_adc_r1_src[] = { "DD Mux", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_adc_r1_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_ADC_R1_SRC_SFT, rt5665_mono_adc_r1_src); @@ -2061,7 +2067,7 @@ static const char * const rt5665_mono_dmic_r_src[] = { "DMIC1 R", "DMIC2 R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_mono_dmic_r_enum, RT5665_MONO_ADC_MIXER, RT5665_MONO_DMIC_R_SRC_SFT, rt5665_mono_dmic_r_src); @@ -2075,14 +2081,14 @@ static const char * const rt5665_sto2_adc1_src[] = { "DD Mux", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adc1l_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADC1L_SRC_SFT, rt5665_sto2_adc1_src); static const struct snd_kcontrol_new rt5665_sto2_adc1l_mux = SOC_DAPM_ENUM("Stereo2 ADC1L Source", rt5665_sto2_adc1l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adc1r_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADC1R_SRC_SFT, rt5665_sto2_adc1_src); @@ -2095,14 +2101,14 @@ static const char * const rt5665_sto2_adc_src[] = { "ADC1 L", "ADC1 R", "ADC2 L" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adcl_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADCL_SRC_SFT, rt5665_sto2_adc_src); static const struct snd_kcontrol_new rt5665_sto2_adcl_mux = SOC_DAPM_ENUM("Stereo2 ADCL Source", rt5665_sto2_adcl_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adcr_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADCR_SRC_SFT, rt5665_sto2_adc_src); @@ -2115,14 +2121,14 @@ static const char * const rt5665_sto2_adc2_src[] = { "DAC MIX", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adc2l_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADC2L_SRC_SFT, rt5665_sto2_adc2_src); static const struct snd_kcontrol_new rt5665_sto2_adc2l_mux = SOC_DAPM_ENUM("Stereo2 ADC2L Source", rt5665_sto2_adc2l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_adc2r_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_ADC2R_SRC_SFT, rt5665_sto2_adc2_src); @@ -2135,7 +2141,7 @@ static const char * const rt5665_sto2_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_dmic_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_DMIC_SRC_SFT, rt5665_sto2_dmic_src); @@ -2147,7 +2153,7 @@ static const char * const rt5665_sto2_dd_l_src[] = { "STO2 DAC", "MONO DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_dd_l_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_DD_L_SRC_SFT, rt5665_sto2_dd_l_src); @@ -2159,7 +2165,7 @@ static const char * const rt5665_sto2_dd_r_src[] = { "STO2 DAC", "MONO DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_sto2_dd_r_enum, RT5665_STO2_ADC_MIXER, RT5665_STO2_DD_R_SRC_SFT, rt5665_sto2_dd_r_src); @@ -2172,14 +2178,14 @@ static const char * const rt5665_dac1_src[] = { "IF1 DAC1", "IF2_1 DAC", "IF2_2 DAC", "IF3 DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_r1_enum, RT5665_AD_DA_MIXER, RT5665_DAC1_R_SEL_SFT, rt5665_dac1_src); static const struct snd_kcontrol_new rt5665_dac_r1_mux = SOC_DAPM_ENUM("DAC R1 Source", rt5665_dac_r1_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dac_l1_enum, RT5665_AD_DA_MIXER, RT5665_DAC1_L_SEL_SFT, rt5665_dac1_src); @@ -2192,14 +2198,14 @@ static const char * const rt5665_dig_dac_mix_src[] = { "Stereo1 DAC Mixer", "Stereo2 DAC Mixer", "Mono DAC Mixer" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dig_dac_mixl_enum, RT5665_A_DAC1_MUX, RT5665_DAC_MIX_L_SFT, rt5665_dig_dac_mix_src); static const struct snd_kcontrol_new rt5665_dig_dac_mixl_mux = SOC_DAPM_ENUM("DAC Digital Mixer L Source", rt5665_dig_dac_mixl_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_dig_dac_mixr_enum, RT5665_A_DAC1_MUX, RT5665_DAC_MIX_R_SFT, rt5665_dig_dac_mix_src); @@ -2212,14 +2218,14 @@ static const char * const rt5665_alg_dac1_src[] = { "Stereo1 DAC Mixer", "DAC1", "DMIC1" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_alg_dac_l1_enum, RT5665_A_DAC1_MUX, RT5665_A_DACL1_SFT, rt5665_alg_dac1_src); static const struct snd_kcontrol_new rt5665_alg_dac_l1_mux = SOC_DAPM_ENUM("Analog DAC L1 Source", rt5665_alg_dac_l1_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_alg_dac_r1_enum, RT5665_A_DAC1_MUX, RT5665_A_DACR1_SFT, rt5665_alg_dac1_src); @@ -2232,14 +2238,14 @@ static const char * const rt5665_alg_dac2_src[] = { "Mono DAC Mixer", "DAC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_alg_dac_l2_enum, RT5665_A_DAC2_MUX, RT5665_A_DACL2_SFT, rt5665_alg_dac2_src); static const struct snd_kcontrol_new rt5665_alg_dac_l2_mux = SOC_DAPM_ENUM("Analog DAC L2 Source", rt5665_alg_dac_l2_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_alg_dac_r2_enum, RT5665_A_DAC2_MUX, RT5665_A_DACR2_SFT, rt5665_alg_dac2_src); @@ -2253,7 +2259,7 @@ static const char * const rt5665_if2_1_adc_in_src[] = { "IF1 DAC2", "IF2_2 DAC", "IF3 DAC", "DAC1 MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if2_1_adc_in_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_1_ADC_IN_SFT, rt5665_if2_1_adc_in_src); @@ -2266,7 +2272,7 @@ static const char * const rt5665_if2_2_adc_in_src[] = { "IF1 DAC2", "IF2_1 DAC", "IF3 DAC", "DAC1 MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if2_2_adc_in_enum, RT5665_DIG_INF2_DATA, RT5665_IF2_2_ADC_IN_SFT, rt5665_if2_2_adc_in_src); @@ -2280,7 +2286,7 @@ static const char * const rt5665_if3_adc_in_src[] = { "IF1 DAC2", "IF2_1 DAC", "IF2_2 DAC", "DAC1 MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if3_adc_in_enum, RT5665_DIG_INF3_DATA, RT5665_IF3_ADC_IN_SFT, rt5665_if3_adc_in_src); @@ -2293,14 +2299,14 @@ static const char * const rt5665_pdm_src[] = { "Stereo1 DAC", "Stereo2 DAC", "Mono DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_pdm_l_enum, RT5665_PDM_OUT_CTRL, RT5665_PDM1_L_SFT, rt5665_pdm_src); static const struct snd_kcontrol_new rt5665_pdm_l_mux = SOC_DAPM_ENUM("PDM L Source", rt5665_pdm_l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_pdm_r_enum, RT5665_PDM_OUT_CTRL, RT5665_PDM1_R_SFT, rt5665_pdm_src); @@ -2314,7 +2320,7 @@ static const char * const rt5665_if1_1_adc1_data_src[] = { "STO1 ADC", "IF2_1 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_1_adc1_data_enum, RT5665_TDM_CTRL_3, RT5665_IF1_ADC1_SEL_SFT, rt5665_if1_1_adc1_data_src); @@ -2326,7 +2332,7 @@ static const char * const rt5665_if1_1_adc2_data_src[] = { "STO2 ADC", "IF2_2 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_1_adc2_data_enum, RT5665_TDM_CTRL_3, RT5665_IF1_ADC2_SEL_SFT, rt5665_if1_1_adc2_data_src); @@ -2338,7 +2344,7 @@ static const char * const rt5665_if1_1_adc3_data_src[] = { "MONO ADC", "IF3 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_1_adc3_data_enum, RT5665_TDM_CTRL_3, RT5665_IF1_ADC3_SEL_SFT, rt5665_if1_1_adc3_data_src); @@ -2350,7 +2356,7 @@ static const char * const rt5665_if1_2_adc1_data_src[] = { "STO1 ADC", "IF1 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_2_adc1_data_enum, RT5665_TDM_CTRL_4, RT5665_IF1_ADC1_SEL_SFT, rt5665_if1_2_adc1_data_src); @@ -2362,7 +2368,7 @@ static const char * const rt5665_if1_2_adc2_data_src[] = { "STO2 ADC", "IF2_1 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_2_adc2_data_enum, RT5665_TDM_CTRL_4, RT5665_IF1_ADC2_SEL_SFT, rt5665_if1_2_adc2_data_src); @@ -2374,7 +2380,7 @@ static const char * const rt5665_if1_2_adc3_data_src[] = { "MONO ADC", "IF2_2 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_2_adc3_data_enum, RT5665_TDM_CTRL_4, RT5665_IF1_ADC3_SEL_SFT, rt5665_if1_2_adc3_data_src); @@ -2386,7 +2392,7 @@ static const char * const rt5665_if1_2_adc4_data_src[] = { "DAC1", "IF3 DAC", }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_if1_2_adc4_data_enum, RT5665_TDM_CTRL_4, RT5665_IF1_ADC4_SEL_SFT, rt5665_if1_2_adc4_data_src); @@ -2401,14 +2407,14 @@ static const char * const rt5665_tdm_adc_data_src[] = { "4123", "4132", "4213", "4231", "4312", "4321" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_tdm1_adc_data_enum, RT5665_TDM_CTRL_3, RT5665_TDM_ADC_SEL_SFT, rt5665_tdm_adc_data_src); static const struct snd_kcontrol_new rt5665_tdm1_adc_mux = SOC_DAPM_ENUM("TDM1 ADC Mux", rt5665_tdm1_adc_data_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5665_tdm2_adc_data_enum, RT5665_TDM_CTRL_4, RT5665_TDM_ADC_SEL_SFT, rt5665_tdm_adc_data_src); @@ -2607,7 +2613,7 @@ static int rt5665_i2s_pin_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int val1, val2, mask1, mask2 = 0; + unsigned int val1, val2, mask1 = 0, mask2 = 0; switch (w->shift) { case RT5665_PWR_I2S2_1_BIT: @@ -2635,13 +2641,17 @@ static int rt5665_i2s_pin_event(struct snd_soc_dapm_widget *w, } switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, mask1, val1); + if (mask1) + snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, + mask1, val1); if (mask2) snd_soc_update_bits(codec, RT5665_GPIO_CTRL_2, mask2, val2); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, mask1, 0); + if (mask1) + snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, + mask1, 0); if (mask2) snd_soc_update_bits(codec, RT5665_GPIO_CTRL_2, mask2, 0); @@ -2684,6 +2694,8 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = { RT5665_DAC_MONO_R_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5665_ASRC_1, RT5665_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5665_ASRC_1, + RT5665_ADC_STO2_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono L ASRC", 1, RT5665_ASRC_1, RT5665_ADC_MONO_L_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono R ASRC", 1, RT5665_ASRC_1, @@ -3227,6 +3239,7 @@ static const struct snd_soc_dapm_route rt5665_dapm_routes[] = { /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc}, {"ADC Mono Left Filter", NULL, "ADC Mono L ASRC", is_using_asrc}, {"ADC Mono Right Filter", NULL, "ADC Mono R ASRC", is_using_asrc}, {"DAC Mono Left Filter", NULL, "DAC Mono L ASRC", is_using_asrc}, @@ -4688,6 +4701,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) regmap_write(rt5665->regmap, RT5665_ASRC_8, 0x0120); out_unlock: + rt5665->calibration_done = true; mutex_unlock(&rt5665->calibrate_mutex); } @@ -4922,7 +4936,7 @@ static struct acpi_device_id rt5665_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5665_acpi_match); #endif -struct i2c_driver rt5665_i2c_driver = { +static struct i2c_driver rt5665_i2c_driver = { .driver = { .name = "rt5665", .of_match_table = of_match_ptr(rt5665_of_match), diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index e27c5a4a0a15..0ec7985ed306 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1717,7 +1717,6 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), /* DSP */ SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -2023,7 +2022,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, { "Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter" }, - { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, { "Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter" }, @@ -2062,7 +2060,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, { "Stereo2 ADC MIXL", NULL, "ADC Stereo2 Filter" }, - { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, { "Stereo2 ADC MIXR", NULL, "ADC Stereo2 Filter" }, @@ -2086,13 +2083,13 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, - { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "TxDP_ADC" }, { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, - { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "TxDP_ADC" }, { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, @@ -2445,10 +2442,9 @@ static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) +static int rt5670_set_codec_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) { - struct snd_soc_codec *codec = dai->codec; struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); unsigned int reg_val = 0; @@ -2472,7 +2468,7 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, if (clk_id != RT5670_SCLK_S_RCCLK) rt5670->sysclk_src = clk_id; - dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + dev_dbg(codec->dev, "Sysclk : %dHz clock id : %d\n", freq, clk_id); return 0; } @@ -2724,7 +2720,6 @@ static int rt5670_resume(struct snd_soc_codec *codec) static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .hw_params = rt5670_hw_params, .set_fmt = rt5670_set_dai_fmt, - .set_sysclk = rt5670_set_dai_sysclk, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, }; @@ -2777,6 +2772,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { .resume = rt5670_resume, .set_bias_level = rt5670_set_bias_level, .idle_bias_off = true, + .set_sysclk = rt5670_set_codec_sysclk, .component_driver = { .controls = rt5670_snd_controls, .num_controls = ARRAY_SIZE(rt5670_snd_controls), @@ -2849,6 +2845,10 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad Tablet B"), }, }, + {} +}; + +static const struct dmi_system_id dmi_platform_intel_bytcht_jdmode2[] = { { .ident = "Lenovo Thinkpad Tablet 10", .matches = { @@ -2883,6 +2883,11 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; rt5670->pdata.dev_gpio = true; rt5670->pdata.jd_mode = 1; + } else if (dmi_check_system(dmi_platform_intel_bytcht_jdmode2)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.dev_gpio = true; + rt5670->pdata.jd_mode = 2; } rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 65ac4518ad06..36e530a36c82 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -41,15 +41,6 @@ #define RT5677_PR_BASE (RT5677_PR_RANGE_BASE + (0 * RT5677_PR_SPACING)) -/* GPIO indexes defined by ACPI */ -enum { - RT5677_GPIO_PLUG_DET = 0, - RT5677_GPIO_MIC_PRESENT_L = 1, - RT5677_GPIO_HOTWORD_DET_L = 2, - RT5677_GPIO_DSP_INT = 3, - RT5677_GPIO_HP_AMP_SHDN_L = 4, -}; - static const struct regmap_range_cfg rt5677_ranges[] = { { .name = "PR", @@ -5030,7 +5021,6 @@ static const struct regmap_config rt5677_regmap = { static const struct i2c_device_id rt5677_i2c_id[] = { { "rt5677", RT5677 }, { "rt5676", RT5676 }, - { "RT5677CE:00", RT5677 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -5041,28 +5031,19 @@ static const struct of_device_id rt5677_of_match[] = { }; MODULE_DEVICE_TABLE(of, rt5677_of_match); -static const struct acpi_gpio_params plug_det_gpio = { RT5677_GPIO_PLUG_DET, 0, false }; -static const struct acpi_gpio_params mic_present_gpio = { RT5677_GPIO_MIC_PRESENT_L, 0, false }; -static const struct acpi_gpio_params headphone_enable_gpio = { RT5677_GPIO_HP_AMP_SHDN_L, 0, false }; - -static const struct acpi_gpio_mapping bdw_rt5677_gpios[] = { - { "plug-det-gpios", &plug_det_gpio, 1 }, - { "mic-present-gpios", &mic_present_gpio, 1 }, - { "headphone-enable-gpios", &headphone_enable_gpio, 1 }, - { NULL }, +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5677_acpi_match[] = { + { "RT5677CE", RT5677 }, + { } }; +MODULE_DEVICE_TABLE(acpi, rt5677_acpi_match); +#endif static void rt5677_read_acpi_properties(struct rt5677_priv *rt5677, struct device *dev) { - int ret; u32 val; - ret = acpi_dev_add_driver_gpios(ACPI_COMPANION(dev), - bdw_rt5677_gpios); - if (ret) - dev_warn(dev, "Failed to add driver gpios\n"); - if (!device_property_read_u32(dev, "DCLK", &val)) rt5677->pdata.dmic2_clk_pin = val; @@ -5301,6 +5282,7 @@ static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", .of_match_table = rt5677_of_match, + .acpi_match_table = ACPI_PTR(rt5677_acpi_match), }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a2702edeb77..8f6814c1eb6b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; +/* AVC: Threshold dB -> register: pre-calculated values */ +static const u16 avc_thr_db2reg[97] = { + 0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068, + 0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F, + 0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414, + 0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172, + 0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083, + 0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E, + 0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010, + 0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005, + 0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002, + 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000}; + /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -382,6 +396,65 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, return 0; } +/* + * custom function to get AVC threshold + * + * The threshold dB is calculated by rearranging the calculation from the + * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> + * dB = ( fls(register_value) - 14.347 ) * 6.02 + * + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_get_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db, i; + u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); + + /* register value 0 => -96dB */ + if (!reg) { + ucontrol->value.integer.value[0] = 96; + ucontrol->value.integer.value[1] = 96; + return 0; + } + + /* get dB from register value (rounded down) */ + for (i = 0; avc_thr_db2reg[i] > reg; i++) + ; + db = i; + + ucontrol->value.integer.value[0] = db; + ucontrol->value.integer.value[1] = db; + + return 0; +} + +/* + * custom function to put AVC threshold + * + * The register value is calculated by following formula: + * register_value = 10^(dB/20) * 0.636 * 2^15 + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_put_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db; + u16 reg; + + db = (int)ucontrol->value.integer.value[0]; + if (db < 0 || db > 96) + return -EINVAL; + reg = avc_thr_db2reg[db]; + snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg); + + return 0; +} + static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ @@ -396,6 +469,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); /* tlv for lineout volume, 31 steps of .5db each */ static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0); +/* tlv for dap avc max gain, 0db, 6db, 12db */ +static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0); + +/* tlv for dap avc threshold, */ +static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600); + static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { /* SOC_DOUBLE_S8_TLV with invert */ { @@ -434,6 +513,16 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 0x1f, 1, lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + + /* Automatic Volume Control (DAP AVC) */ + SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), + SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), + SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0, + avc_max_gain), + SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0), + SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, + 0, 96, 0, avc_get_threshold, avc_put_threshold, + avc_threshold), }; /* mute the codec used by alsa core */ diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8a90ba8cd71..d7d03c92cb8a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1210,7 +1210,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { static struct snd_soc_dai_driver dac31xx_dai_driver[] = { { - .name = "tlv32dac31xx-hifi", + .name = "tlv320dac31xx-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 20695b691aff..65c059b5ffd7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -482,8 +482,6 @@ struct wm_coeff_ctl_ops { struct snd_ctl_elem_value *ucontrol); int (*xput)(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - int (*xinfo)(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); }; struct wm_coeff_ctl { @@ -1890,7 +1888,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, } if (be32_to_cpu(val) != 0xbedead) - adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbeadead\n", + adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n", pos + len, be32_to_cpu(val)); alg = kzalloc(len * 2, GFP_KERNEL | GFP_DMA); @@ -2654,7 +2652,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", mc->shift); + snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); dsp->preloaded = ucontrol->value.integer.value[0]; diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c new file mode 100644 index 000000000000..032fb7cf6cbd --- /dev/null +++ b/sound/soc/codecs/zx_aud96p22.c @@ -0,0 +1,403 @@ +/* + * Copyright (C) 2017 Sanechips Technology Co., Ltd. + * Copyright 2017 Linaro Ltd. + * + * Author: Baoyou Xie <baoyou.xie@linaro.org> + * + * 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/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/tlv.h> + +#define AUD96P22_RESET 0x00 +#define RST_DAC_DPZ BIT(0) +#define RST_ADC_DPZ BIT(1) +#define AUD96P22_I2S1_CONFIG_0 0x03 +#define I2S1_MS_MODE BIT(3) +#define I2S1_MODE_MASK 0x7 +#define I2S1_MODE_RIGHT_J 0x0 +#define I2S1_MODE_I2S 0x1 +#define I2S1_MODE_LEFT_J 0x2 +#define AUD96P22_PD_0 0x15 +#define AUD96P22_PD_1 0x16 +#define AUD96P22_PD_3 0x18 +#define AUD96P22_PD_4 0x19 +#define AUD96P22_MUTE_0 0x1d +#define AUD96P22_MUTE_2 0x1f +#define AUD96P22_MUTE_4 0x21 +#define AUD96P22_RECVOL_0 0x24 +#define AUD96P22_RECVOL_1 0x25 +#define AUD96P22_PGA1VOL_0 0x26 +#define AUD96P22_PGA1VOL_1 0x27 +#define AUD96P22_LMVOL_0 0x34 +#define AUD96P22_LMVOL_1 0x35 +#define AUD96P22_HS1VOL_0 0x38 +#define AUD96P22_HS1VOL_1 0x39 +#define AUD96P22_PGA1SEL_0 0x47 +#define AUD96P22_PGA1SEL_1 0x48 +#define AUD96P22_LDR1SEL_0 0x59 +#define AUD96P22_LDR1SEL_1 0x60 +#define AUD96P22_LDR2SEL_0 0x5d +#define AUD96P22_REG_MAX 0xfb + +struct aud96p22_priv { + struct regmap *regmap; +}; + +static int aud96p22_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct aud96p22_priv *priv = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = priv->regmap; + + if (event != SND_SOC_DAPM_POST_PMU) + return -EINVAL; + + /* Assert/de-assert the bit to reset ADC data path */ + regmap_update_bits(regmap, AUD96P22_RESET, RST_ADC_DPZ, 0); + regmap_update_bits(regmap, AUD96P22_RESET, RST_ADC_DPZ, RST_ADC_DPZ); + + return 0; +} + +static int aud96p22_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct aud96p22_priv *priv = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = priv->regmap; + + if (event != SND_SOC_DAPM_POST_PMU) + return -EINVAL; + + /* Assert/de-assert the bit to reset DAC data path */ + regmap_update_bits(regmap, AUD96P22_RESET, RST_DAC_DPZ, 0); + regmap_update_bits(regmap, AUD96P22_RESET, RST_DAC_DPZ, RST_DAC_DPZ); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(lm_tlv, -11550, 50, 0); +static const DECLARE_TLV_DB_SCALE(hs_tlv, -3900, 300, 0); +static const DECLARE_TLV_DB_SCALE(rec_tlv, -9550, 50, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1800, 100, 0); + +static const struct snd_kcontrol_new aud96p22_snd_controls[] = { + /* Volume control */ + SOC_DOUBLE_R_TLV("Master Playback Volume", AUD96P22_LMVOL_0, + AUD96P22_LMVOL_1, 0, 0xff, 0, lm_tlv), + SOC_DOUBLE_R_TLV("Headphone Volume", AUD96P22_HS1VOL_0, + AUD96P22_HS1VOL_1, 0, 0xf, 0, hs_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", AUD96P22_RECVOL_0, + AUD96P22_RECVOL_1, 0, 0xff, 0, rec_tlv), + SOC_DOUBLE_R_TLV("Analogue Capture Volume", AUD96P22_PGA1VOL_0, + AUD96P22_PGA1VOL_1, 0, 0x37, 0, pga_tlv), + + /* Mute control */ + SOC_DOUBLE("Master Playback Switch", AUD96P22_MUTE_2, 0, 1, 1, 1), + SOC_DOUBLE("Headphone Switch", AUD96P22_MUTE_2, 4, 5, 1, 1), + SOC_DOUBLE("Line Out Switch", AUD96P22_MUTE_4, 0, 1, 1, 1), + SOC_DOUBLE("Speaker Switch", AUD96P22_MUTE_4, 2, 3, 1, 1), + SOC_DOUBLE("Master Capture Switch", AUD96P22_MUTE_0, 0, 1, 1, 1), + SOC_DOUBLE("Analogue Capture Switch", AUD96P22_MUTE_0, 2, 3, 1, 1), +}; + +/* Input mux kcontrols */ +static const unsigned int ain_mux_values[] = { + 0, 1, 3, 4, 5, +}; + +static const char * const ainl_mux_texts[] = { + "AINL1 differential", + "AINL1 single-ended", + "AINL3 single-ended", + "AINL2 differential", + "AINL2 single-ended", +}; + +static const char * const ainr_mux_texts[] = { + "AINR1 differential", + "AINR1 single-ended", + "AINR3 single-ended", + "AINR2 differential", + "AINR2 single-ended", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(ainl_mux_enum, AUD96P22_PGA1SEL_0, + 0, 0x7, ainl_mux_texts, ain_mux_values); +static SOC_VALUE_ENUM_SINGLE_DECL(ainr_mux_enum, AUD96P22_PGA1SEL_1, + 0, 0x7, ainr_mux_texts, ain_mux_values); + +static const struct snd_kcontrol_new ainl_mux_kcontrol = + SOC_DAPM_ENUM("AINL Mux", ainl_mux_enum); +static const struct snd_kcontrol_new ainr_mux_kcontrol = + SOC_DAPM_ENUM("AINR Mux", ainr_mux_enum); + +/* Output mixer kcontrols */ +static const struct snd_kcontrol_new ld1_left_kcontrols[] = { + SOC_DAPM_SINGLE("DACL LD1L Switch", AUD96P22_LDR1SEL_0, 0, 1, 0), + SOC_DAPM_SINGLE("AINL LD1L Switch", AUD96P22_LDR1SEL_0, 1, 1, 0), + SOC_DAPM_SINGLE("AINR LD1L Switch", AUD96P22_LDR1SEL_0, 2, 1, 0), +}; + +static const struct snd_kcontrol_new ld1_right_kcontrols[] = { + SOC_DAPM_SINGLE("DACR LD1R Switch", AUD96P22_LDR1SEL_1, 8, 1, 0), + SOC_DAPM_SINGLE("AINR LD1R Switch", AUD96P22_LDR1SEL_1, 9, 1, 0), + SOC_DAPM_SINGLE("AINL LD1R Switch", AUD96P22_LDR1SEL_1, 10, 1, 0), +}; + +static const struct snd_kcontrol_new ld2_kcontrols[] = { + SOC_DAPM_SINGLE("DACL LD2 Switch", AUD96P22_LDR2SEL_0, 0, 1, 0), + SOC_DAPM_SINGLE("AINL LD2 Switch", AUD96P22_LDR2SEL_0, 1, 1, 0), + SOC_DAPM_SINGLE("DACR LD2 Switch", AUD96P22_LDR2SEL_0, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget aud96p22_dapm_widgets[] = { + /* Overall power bit */ + SND_SOC_DAPM_SUPPLY("POWER", AUD96P22_PD_0, 0, 0, NULL, 0), + + /* Input pins */ + SND_SOC_DAPM_INPUT("AINL1P"), + SND_SOC_DAPM_INPUT("AINL2P"), + SND_SOC_DAPM_INPUT("AINL3"), + SND_SOC_DAPM_INPUT("AINL1N"), + SND_SOC_DAPM_INPUT("AINL2N"), + SND_SOC_DAPM_INPUT("AINR2N"), + SND_SOC_DAPM_INPUT("AINR1N"), + SND_SOC_DAPM_INPUT("AINR3"), + SND_SOC_DAPM_INPUT("AINR2P"), + SND_SOC_DAPM_INPUT("AINR1P"), + + /* Input muxes */ + SND_SOC_DAPM_MUX("AINLMUX", AUD96P22_PD_1, 2, 0, &ainl_mux_kcontrol), + SND_SOC_DAPM_MUX("AINRMUX", AUD96P22_PD_1, 3, 0, &ainr_mux_kcontrol), + + /* ADCs */ + SND_SOC_DAPM_ADC_E("ADCL", "Capture Left", AUD96P22_PD_1, 0, 0, + aud96p22_adc_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADCR", "Capture Right", AUD96P22_PD_1, 1, 0, + aud96p22_adc_event, SND_SOC_DAPM_POST_PMU), + + /* DACs */ + SND_SOC_DAPM_DAC_E("DACL", "Playback Left", AUD96P22_PD_3, 0, 0, + aud96p22_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_DAC_E("DACR", "Playback Right", AUD96P22_PD_3, 1, 0, + aud96p22_dac_event, SND_SOC_DAPM_POST_PMU), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("LD1L", AUD96P22_PD_3, 6, 0, ld1_left_kcontrols, + ARRAY_SIZE(ld1_left_kcontrols)), + SND_SOC_DAPM_MIXER("LD1R", AUD96P22_PD_3, 7, 0, ld1_right_kcontrols, + ARRAY_SIZE(ld1_right_kcontrols)), + SND_SOC_DAPM_MIXER("LD2", AUD96P22_PD_4, 2, 0, ld2_kcontrols, + ARRAY_SIZE(ld2_kcontrols)), + + /* Headset power switch */ + SND_SOC_DAPM_SUPPLY("HS1L", AUD96P22_PD_3, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HS1R", AUD96P22_PD_3, 5, 0, NULL, 0), + + /* Output pins */ + SND_SOC_DAPM_OUTPUT("HSOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTMP"), + SND_SOC_DAPM_OUTPUT("LINEOUTMN"), + SND_SOC_DAPM_OUTPUT("LINEOUTR"), + SND_SOC_DAPM_OUTPUT("HSOUTR"), +}; + +static const struct snd_soc_dapm_route aud96p22_dapm_routes[] = { + { "AINLMUX", "AINL1 differential", "AINL1N" }, + { "AINLMUX", "AINL1 single-ended", "AINL1P" }, + { "AINLMUX", "AINL3 single-ended", "AINL3" }, + { "AINLMUX", "AINL2 differential", "AINL2N" }, + { "AINLMUX", "AINL2 single-ended", "AINL2P" }, + + { "AINRMUX", "AINR1 differential", "AINR1N" }, + { "AINRMUX", "AINR1 single-ended", "AINR1P" }, + { "AINRMUX", "AINR3 single-ended", "AINR3" }, + { "AINRMUX", "AINR2 differential", "AINR2N" }, + { "AINRMUX", "AINR2 single-ended", "AINR2P" }, + + { "ADCL", NULL, "AINLMUX" }, + { "ADCR", NULL, "AINRMUX" }, + + { "ADCL", NULL, "POWER" }, + { "ADCR", NULL, "POWER" }, + { "DACL", NULL, "POWER" }, + { "DACR", NULL, "POWER" }, + + { "LD1L", "DACL LD1L Switch", "DACL" }, + { "LD1L", "AINL LD1L Switch", "AINLMUX" }, + { "LD1L", "AINR LD1L Switch", "AINRMUX" }, + + { "LD1R", "DACR LD1R Switch", "DACR" }, + { "LD1R", "AINR LD1R Switch", "AINRMUX" }, + { "LD1R", "AINL LD1R Switch", "AINLMUX" }, + + { "LD2", "DACL LD2 Switch", "DACL" }, + { "LD2", "AINL LD2 Switch", "AINLMUX" }, + { "LD2", "DACR LD2 Switch", "DACR" }, + + { "HSOUTL", NULL, "LD1L" }, + { "HSOUTR", NULL, "LD1R" }, + { "HSOUTL", NULL, "HS1L" }, + { "HSOUTR", NULL, "HS1R" }, + + { "LINEOUTL", NULL, "LD1L" }, + { "LINEOUTR", NULL, "LD1R" }, + + { "LINEOUTMP", NULL, "LD2" }, + { "LINEOUTMN", NULL, "LD2" }, +}; + +static struct snd_soc_codec_driver aud96p22_driver = { + .component_driver = { + .controls = aud96p22_snd_controls, + .num_controls = ARRAY_SIZE(aud96p22_snd_controls), + .dapm_widgets = aud96p22_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aud96p22_dapm_widgets), + .dapm_routes = aud96p22_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aud96p22_dapm_routes), + }, +}; + +static int aud96p22_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct aud96p22_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + struct regmap *regmap = priv->regmap; + unsigned int val; + + /* Master/slave mode */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val = I2S1_MS_MODE; + break; + default: + return -EINVAL; + } + + regmap_update_bits(regmap, AUD96P22_I2S1_CONFIG_0, I2S1_MS_MODE, val); + + /* Audio format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = I2S1_MODE_RIGHT_J; + break; + case SND_SOC_DAIFMT_I2S: + val = I2S1_MODE_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = I2S1_MODE_LEFT_J; + break; + default: + return -EINVAL; + } + + regmap_update_bits(regmap, AUD96P22_I2S1_CONFIG_0, I2S1_MODE_MASK, val); + + return 0; +} + +static struct snd_soc_dai_ops aud96p22_dai_ops = { + .set_fmt = aud96p22_set_fmt, +}; + +#define AUD96P22_RATES SNDRV_PCM_RATE_8000_192000 +#define AUD96P22_FORMATS (\ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver aud96p22_dai = { + .name = "aud96p22-dai", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AUD96P22_RATES, + .formats = AUD96P22_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AUD96P22_RATES, + .formats = AUD96P22_FORMATS, + }, + .ops = &aud96p22_dai_ops, +}; + +static const struct regmap_config aud96p22_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = AUD96P22_REG_MAX, + .cache_type = REGCACHE_RBTREE, +}; + +static int aud96p22_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct aud96p22_priv *priv; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &aud96p22_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(dev, "failed to init i2c regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + ret = snd_soc_register_codec(dev, &aud96p22_driver, &aud96p22_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec: %d\n", ret); + return ret; + } + + return 0; +} + +static int aud96p22_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +const struct of_device_id aud96p22_dt_ids[] = { + { .compatible = "zte,zx-aud96p22", }, + { } +}; +MODULE_DEVICE_TABLE(of, aud96p22_dt_ids); + +static struct i2c_driver aud96p22_i2c_driver = { + .driver = { + .name = "zx_aud96p22", + .of_match_table = aud96p22_dt_ids, + }, + .probe = aud96p22_i2c_probe, + .remove = aud96p22_i2c_remove, +}; +module_i2c_driver(aud96p22_i2c_driver); + +MODULE_DESCRIPTION("ZTE ASoC AUD96P22 CODEC driver"); +MODULE_AUTHOR("Baoyou Xie <baoyou.xie@linaro.org>"); +MODULE_LICENSE("GPL v2"); |