[alsa-devel] [PATCH v3 6/8] ASoC: da7210: Add support for mute and zero cross controls
This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G
It also adds digital_mute() call back.
Signed-off-by: Ashish Chavan ashish.chavan@kpitcummins.com Signed-off-by: David Dajun Chen dchen@diasemi.com --- Changes since v2: - Added digital_mute() call back and removed control for "DAC Mute Switch" - Renamed mute control switches
Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8dad77e..ef67a4d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -47,6 +47,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -58,6 +59,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -173,6 +175,9 @@ /* AUX2 bit fields */ #define DA7210_AUX2_EN (1 << 3)
+/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1"
/* @@ -221,6 +226,13 @@ static const unsigned int aux1_vol_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) };
+static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = {
SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -277,6 +289,22 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, eq_gain_tlv), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_SINGLE("ADC Left Capture Switch", DA7210_ADC, 2, 1, 0), + SOC_SINGLE("ADC Right Capture Switch", DA7210_ADC, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), };
/* @@ -674,6 +702,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; }
+static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
@@ -681,6 +721,7 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) static struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, };
static struct snd_soc_dai_driver da7210_dai = { @@ -782,6 +823,9 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN);
+ /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
participants (1)
-
Ashish Chavan