[alsa-devel] [PATCH v2 2/9] ASoC: da7210: Add support for line input and mic
DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage.
Signed-off-by: Ashish Chavan ashish.chavan@kpitcummins.com Signed-off-by: David Dajun Chen dchen@diasemi.com --- Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 54 +++++++++++++++++++++++++++++++++++++++----- 1 files changed, 47 insertions(+), 7 deletions(-)
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 1ef058b..87b9ae5 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,10 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -147,6 +151,15 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7)
+/* AUX1_L bit fields */ +#define DA7210_AUX1_L_EN (1 << 7) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) + #define DA7210_VERSION "0.0.1"
/* @@ -165,6 +178,9 @@ */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0);
static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -180,6 +196,13 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) };
+static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const struct snd_kcontrol_new da7210_snd_controls[] = {
SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -193,6 +216,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3f, 0, lineout_vol_tlv), SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + + /* MIC related controls */ + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + + /* AUX related controls */ + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), };
/* Codec private data */ @@ -246,13 +284,9 @@ static int da7210_startup(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10);
} else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + /* Enable Mic,AUX1 and AUX2 */ + snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0xD); + snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0xD); }
return 0; @@ -521,6 +555,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R);
+ /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
On Thu, Oct 13, 2011 at 07:34:04PM +0530, Ashish Chavan wrote:
} else {
/* Volume 7 */
snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7);
snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7);
/* Enable Mic */
snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1);
snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1);
/* Enable Mic,AUX1 and AUX2 */
snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0xD);
snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0xD);
So, I did mention that the DAPM patch should come first when I reviewed your previous series...
On Thu, 2011-10-13 at 17:19 +0100, Mark Brown wrote:
On Thu, Oct 13, 2011 at 07:34:04PM +0530, Ashish Chavan wrote:
} else {
/* Volume 7 */
snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7);
snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7);
/* Enable Mic */
snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1);
snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1);
/* Enable Mic,AUX1 and AUX2 */
snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0xD);
snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0xD);
So, I did mention that the DAPM patch should come first when I reviewed your previous series...
Posted the series after rearranging the sequence and removing already merged patch.
Thanks for your time.
participants (2)
-
Ashish Chavan
-
Mark Brown