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authorAshish Chavan <ashish.chavan@kpitcummins.com>2011-10-15 14:50:06 +0530
committerMark Brown <broonie@opensource.wolfsonmicro.com>2011-10-17 22:43:33 +0100
commit4ced2b96f3d8b5944611e4e93b59b69ad440e10e (patch)
tree3c29132407fea5563d48324c53c579c857778537 /sound
parent0ee6e9e721fc85e093e20e7a9ca848cfa71f80a9 (diff)
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ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
This patch add controls for setting cut-off for high pass and voice filters of ADC and DAC. There are also switches to enable/disable these filters. Also removed hard coded, fixed values of these parameters used by previous version of driver. Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: David Dajun Chen <dchen@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/codecs/da7210.c57
1 files changed, 32 insertions, 25 deletions
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 7dc1259010be..fa0d5125e70b 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -167,6 +167,28 @@ static const unsigned int hp_out_tlv[] = {
static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0);
static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1);
+/* ADC and DAC high pass filter f0 value */
+static const char const *da7210_hpf_cutoff_txt[] = {
+ "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi"
+};
+
+static const struct soc_enum da7210_dac_hpf_cutoff =
+ SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt);
+
+static const struct soc_enum da7210_adc_hpf_cutoff =
+ SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt);
+
+/* ADC and DAC voice (8kHz) high pass cutoff value */
+static const char const *da7210_vf_cutoff_txt[] = {
+ "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
+};
+
+static const struct soc_enum da7210_dac_vf_cutoff =
+ SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt);
+
+static const struct soc_enum da7210_adc_vf_cutoff =
+ SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt);
+
static const struct snd_kcontrol_new da7210_snd_controls[] = {
SOC_DOUBLE_R_TLV("HeadPhone Playback Volume",
@@ -200,6 +222,16 @@ 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),
+
+ SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0),
+ SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff),
+ SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0),
+ SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff),
+
+ SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0),
+ SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff),
+ SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0),
+ SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff),
};
/* Codec private data */
@@ -275,7 +307,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
u32 dai_cfg1;
- u32 hpf_reg, hpf_mask, hpf_value;
u32 fs, bypass;
/* set DAI source to Left and Right ADC */
@@ -306,68 +337,45 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1);
- hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ?
- DA7210_DAC_HPF : DA7210_ADC_HPF;
-
switch (params_rate(params)) {
case 8000:
fs = DA7210_PLL_FS_8000;
- hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN;
- hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN;
bypass = DA7210_PLL_BYP;
break;
case 11025:
fs = DA7210_PLL_FS_11025;
- hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN;
- hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN;
bypass = 0;
break;
case 12000:
fs = DA7210_PLL_FS_12000;
- hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN;
- hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN;
bypass = DA7210_PLL_BYP;
break;
case 16000:
fs = DA7210_PLL_FS_16000;
- hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN;
- hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN;
bypass = DA7210_PLL_BYP;
break;
case 22050:
fs = DA7210_PLL_FS_22050;
- hpf_mask = DA7210_VOICE_EN;
- hpf_value = 0;
bypass = 0;
break;
case 32000:
fs = DA7210_PLL_FS_32000;
- hpf_mask = DA7210_VOICE_EN;
- hpf_value = 0;
bypass = DA7210_PLL_BYP;
break;
case 44100:
fs = DA7210_PLL_FS_44100;
- hpf_mask = DA7210_VOICE_EN;
- hpf_value = 0;
bypass = 0;
break;
case 48000:
fs = DA7210_PLL_FS_48000;
- hpf_mask = DA7210_VOICE_EN;
- hpf_value = 0;
bypass = DA7210_PLL_BYP;
break;
case 88200:
fs = DA7210_PLL_FS_88200;
- hpf_mask = DA7210_VOICE_EN;
- hpf_value = 0;
bypass = 0;
break;
case 96000:
fs = DA7210_PLL_FS_96000;
- hpf_mask = DA7210_VOICE_EN;
- hpf_value = 0;
bypass = DA7210_PLL_BYP;
break;
default:
@@ -377,7 +385,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
/* Disable active mode */
snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
- snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value);
snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs);
snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass);