diff options
Diffstat (limited to 'sound/soc/codecs')
-rw-r--r-- | sound/soc/codecs/Kconfig | 5 | ||||
-rw-r--r-- | sound/soc/codecs/Makefile | 2 | ||||
-rw-r--r-- | sound/soc/codecs/ak4613.c | 95 | ||||
-rw-r--r-- | sound/soc/codecs/ak4642.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/cs35l34.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/cs35l35.c | 94 | ||||
-rw-r--r-- | sound/soc/codecs/cs35l35.h | 6 | ||||
-rw-r--r-- | sound/soc/codecs/cs4271.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/cs53l30.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/da7213.c | 39 | ||||
-rw-r--r-- | sound/soc/codecs/da7218.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/da7219-aad.c | 31 | ||||
-rw-r--r-- | sound/soc/codecs/da7219.c | 53 | ||||
-rw-r--r-- | sound/soc/codecs/da7219.h | 5 | ||||
-rw-r--r-- | sound/soc/codecs/es8316.c | 637 | ||||
-rw-r--r-- | sound/soc/codecs/es8316.h | 129 | ||||
-rw-r--r-- | sound/soc/codecs/rt286.c | 7 | ||||
-rw-r--r-- | sound/soc/codecs/rt5645.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/rt5663.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/rt5663.h | 4 | ||||
-rw-r--r-- | sound/soc/codecs/rt5670.c | 14 | ||||
-rw-r--r-- | sound/soc/codecs/rt5677.c | 32 |
22 files changed, 1073 insertions, 99 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883ed4c8a551..f0f794186186 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 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28a63fdaf982..e878306ce46e 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 @@ -300,6 +301,7 @@ 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 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 6dd7578f0bb8..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> @@ -772,7 +774,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(50); } - } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock)); + } while ((i < DA7213_SRM_CHECK_RETRIES) && (!srm_lock)); if (!srm_lock) dev_warn(codec->dev, "SRM failed to lock\n"); @@ -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/rt286.c b/sound/soc/codecs/rt286.c index 9c365a7f758d..7899a2cdeb42 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1108,6 +1108,13 @@ static const struct dmi_system_id force_combo_jack_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform") } }, + { + .ident = "Thinkpad Helix 2nd", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad Helix 2nd") + } + }, { } }; diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 87844a45886a..206b41688d96 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3897,6 +3897,7 @@ 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/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/rt5670.c b/sound/soc/codecs/rt5670.c index e27c5a4a0a15..a5f15a104c47 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), @@ -2086,13 +2085,13 @@ 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" }, @@ -2849,6 +2848,10 @@ 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 +2886,11 @@ 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, |