[alsa-devel] [PATCH v7 3/3] ASoC: da7210: Add support for line input and mic

[Ashish Chavan]

On Thu, 2011-10-20 at 18:01 +0100, Girdwood, Liam wrote:
> On 20 October 2011 15:42, Ashish Chavan <ashish.chavan at kpitcummins.com> wrote:
> > 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 at kpitcummins.com>
> > Signed-off-by: David Dajun Chen <dchen at diasemi.com>
> > ---
> > Changes since v2:
> > - Removed static enable of mic and aux, as now DAPM will take care of
> > that
> >
> > Changes since v1:
> > - Removed explicit setting of default gains
> > - Removed control to set mic bias voltage
> > ---
> >  sound/soc/codecs/da7210.c |   32 ++++++++++++++++++++++++++++++++
> >  1 files changed, 32 insertions(+), 0 deletions(-)
> >
> > diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
> > index eaec60a..2f38b39 100644
> > --- a/sound/soc/codecs/da7210.c
> > +++ b/sound/soc/codecs/da7210.c
> > @@ -181,9 +181,14 @@
> >
> >  /* AUX1_L bit fields */
> >  #define DA7210_AUX1_L_VOL              (0x3F << 0)
> > +#define DA7210_AUX1_L_EN               (1 << 7)
> >
> >  /* AUX1_R bit fields */
> >  #define DA7210_AUX1_R_VOL              (0x3F << 0)
> > +#define DA7210_AUX1_R_EN               (1 << 7)
> > +
> > +/* AUX2 bit fields */
> > +#define DA7210_AUX2_EN                 (1 << 3)
> >
> >  /* Minimum INPGA and AUX1 volume to enable noise suppression */
> >  #define DA7210_INPGA_MIN_VOL_NS                0x0A  /* 10.5dB */
> > @@ -234,9 +239,19 @@ 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 DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0);
> >  static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 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);
> >
> >  /* ADC and DAC high pass filter f0 value */
> >  static const char const *da7210_hpf_cutoff_txt[] = {
> > @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = {
> >        SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0,
> >                       mono_vol_tlv),
> >
> > +       SOC_DOUBLE_R_TLV("Mic Capture Volume",
> > +                        DA7210_MIC_L, DA7210_MIC_R,
> > +                        0, 0x5, 0, mic_vol_tlv),
> > +       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),
> > +
> >        /* DAC Equalizer  controls */
> >        SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0),
> >        SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1,
> > @@ -928,6 +954,12 @@ 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);
> > +
> 
> Your comment states this is now done by DAPM ?
> 

Yes, but that comment is for other part of code, not this one. Actually,
first version of patch series has this patch before DAPM patch. At that
time, the setting of in-mixer and  out-mixer was taken care by startup
function. I added enables for mic and aux in to startup function at that
time. After Mark's comments, I rearranged the patch series and made DAPM
patch before this one. Now as, DAPM patch removed entire startup
function and hence the code portion related to that comment doesn't
exists. I know it is bit confusing, but that's how the patch has
evolved! Do you think we should remove confusing comment from the
version history?

Anyways, as explained in the comment in DAPM patch, this enables are
essential and doesn't cost power as DAPM takes care of STANDBY bits.