Re: ASoC: TLV320AIC3x: Adding additional functionality for 3106 with [Patch] for discuss

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On 03/14/2016 04:57 PM, Timur Karaldin wrote:
> Hi, Peter!
> What's about placing capture controls in the Playback section of Alsamixer
> view?

Yeah, it is annoying. They can be fixed by:
"Line1L Switch" -> "Line1L Capture Switch" and so on. On the other hand we had
these unfortunate control names I think since the beginning and it is kind of
an ABI for existing products. I don't have issue changing them, but existing
setups relying on the control names might break...

> Does resetting of some registers during start/stop of recording
> connect with it?

Not sure what you mean.

-- 
Péter

> 
> 14.03.2016 17:47, Peter Ujfalusi пишет:
>> Hi Timur,
>>
>> On 03/14/16 13:06, Timur Karaldin wrote:
>>> Hi, colleagues!
>>>
>>> I'm newbie with ALSA devel. I need to get extra functionality from codec
>>> through ALSA mixer.
>>> Task: 1. making possibility to set-up gain for each input (XXX_2_LADCCTRL and
>>> XXX_2_RADCCTRL registers)
>> This part is a bit tricky as in the registers:
>> value 0..8 is gain from 0dB to -12dB in steps of -1.5dB and value 0xf means
>> basically mute/disconnect.
>>
>>> 2. making possibility to set-up debounce for button and headset detection (
>>> HEADSET_DETECT_CTRL_A register)
>>> 3. making possibility to enable high-power AC-coupled output
>>> (HEADSET_DETECT_CTRL_A register)
>>> 4. making possibility to enable stereo pseudo differential output
>>> (HEADSET_DETECT_CTRL_B register)
>>> 5. enable headset detection (HEADSET_DETECT_CTRL_A)
>>> 6. getting headset detect status (HEADSET_DETECT_CTRL_A)
>>>
>>> Well, my patch works in generally, but I don't like it very much, because:
>>> 1. All controls for input amplifiers' gain are shown in Alsamixer-Playback
>>> section. But not only mine, capture controls from base codec's version are
>>> shown in playback section too, for example (Right or Left) PGA Mixer Line1(L
>>> or R)/2(L or R)/MIC3(L or R), which is definitely capture settings. Should I
>>> worried about it? because, as I understand, it could not be setted correctly
>>> during changing playback/capture mode for power-saving reason.
>>> 2.  "Right PGA Mixer Line1R" control (and similar) using the same bits in
>>> LINE1R_2_RADCCTRL register as my "Line1R RADC Gain" control (and similar) and
>>> put there b"1111" when user switch disconnect it from ADC. So my control is
>>> not shown when user disconnect it from ADC.
>>> 3. When I start/stop arecord some of my options were reset (debounces, headset
>>> enable, high-power AC-coupled, pseudo differential output). When I set up it
>>> in shell and then start arecord, it's reset again. Is it the reason because of
>>> displaying controls in playback mixer section?
>>>
>>> I will be very pleased for any advise, thank you!
>>>
>>>
>>> here is my patch for tlv320aic3x.c
>>> ---
>>>   sound/soc/codecs/tlv320aic3x.c | 55
>>> +++++++++++++++++++++++++++++++++++++++---
>>>   1 file changed, 52 insertions(+), 3 deletions(-)
>>>
>>> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
>>> index d7349bc..27d031e 100644
>>> --- a/sound/soc/codecs/tlv320aic3x.c
>>> +++ b/sound/soc/codecs/tlv320aic3x.c
>>> @@ -122,11 +122,24 @@ static const struct reg_default aic3x_reg[] = {
>>>          { 108, 0x00 }, { 109, 0x00 },
>>>   };
>>>
>>> +static bool aic3106_volatile(struct device *dev, unsigned int reg)
>>> +{
>>> +    switch(reg)
>>> +    {
>>> +        case AIC3X_HEADSET_DETECT_CTRL_A:
>>> +        case AIC3X_HEADSET_DETECT_CTRL_B:
>>> +            return true;
>>> +        default:
>>> +            return false;
>>> +    }
>>> +}
>>> +
>>>   static const struct regmap_config aic3x_regmap = {
>>>          .reg_bits = 8,
>>>          .val_bits = 8,
>>>
>>>          .max_register = DAC_ICC_ADJ,
>>> +    .volatile_reg = aic3106_volatile,
>>>          .reg_defaults = aic3x_reg,
>>>          .num_reg_defaults = ARRAY_SIZE(aic3x_reg),
>>>          .cache_type = REGCACHE_RBTREE,
>>> @@ -196,7 +209,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
>>>          struct snd_kcontrol *kcontrol, int event)
>>>   {
>>>          struct snd_soc_codec *codec = w->codec;
>>> -       struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
>>> +    struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
>> You have couple of these nop changes in the patch...
>>
>>>          switch (event) {
>>>          case SND_SOC_DAPM_POST_PMU:
>>> @@ -270,6 +283,26 @@ static const struct soc_enum aic3x_agc_decay_enum[] = {
>>>          SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay),
>>>   };
>>>
>>> +static const char *aic3x_adc_gain[] = { "0dB", "-1.5dB", "-3dB", "-4.5dB",
>>> "-6dB", "-7.5dB", "-9dB", "-10.5dB", "-12dB" };
>>> +static const struct soc_enum aic3x_adc_gain_enum[] = {
>>> +       SOC_ENUM_SINGLE(MIC3LR_2_LADC_CTRL, 4, 9, aic3x_adc_gain),
>>> +       SOC_ENUM_SINGLE(MIC3LR_2_RADC_CTRL, 0, 9, aic3x_adc_gain),
>>> +       SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 3, 9, aic3x_adc_gain),
>>> +       SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 3, 9, aic3x_adc_gain),
>>> +       SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 3, 9, aic3x_adc_gain),
>>> +       SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 3, 9, aic3x_adc_gain),
>>> +};
>> So the issue is that we have the DAPM switches controlling exactly the same
>> registers. I believe if you would set the gain in a way that bit0 is 1, then
>> DAPM will think that the path is disconnected.
>> Also if you would set the gain and then mute and unmute the path you would
>> have lost the gain you wanted to have..
>>
>> The only way I can think of implementing these mixers is to have two sets of
>> custom callbacks. one set is to set/get the gain and the other is to set/get
>> the mute/disconnect on these.
>> When the path is disconnected you should not write the gain change to the
>> chip, but cache it and if the path is unmuted, you write the cached gain. When
>> you mute the path you should take the set gain first, cache it, then
>> disconnect the path.
>>
>> For these gains you should have DECLARE_TLV_DB_SCALE() and use
>> SOC_SINGLE_TLV(). Make sure that the control name matches with the
>> corresponding DAPM widget's name so ALSA can match them correctly.
>>
>>> +static const char *aic3x_headset_debounce[] = { "16ms", "32ms", "64ms",
>>> "128ms", "256ms", "512ms" };
>>> +static const struct soc_enum aic3x_headset_debounce_enum[] = {
>>> +    SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 2, 6,
>>> aic3x_headset_debounce),
>>> +};
>>> +static const char *aic3x_button_debounce[] = { "0ms", "8ms", "16ms",
>>> "32ms" };
>>> +static const struct soc_enum aic3x_button_debounce_enum[] = {
>>> +    SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 0, 4,
>>> aic3x_button_debounce),
>>> +};
>>> +
>>> +
>>> +
>>>   /*
>>>    * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
>>>    */
>>> @@ -399,6 +432,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[]
>>> = {
>>>          SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
>>>
>>>          SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
>>> +    /* Additional controls */
>>> +       SOC_ENUM("Mic3L LADC Gain", aic3x_adc_gain_enum[0]),
>>> +       SOC_ENUM("Mic3R RADC Gain", aic3x_adc_gain_enum[1]),
>>> +       SOC_ENUM("Line1L LADC Gain", aic3x_adc_gain_enum[2]),
>>> +       SOC_ENUM("Line1R RADC Gain", aic3x_adc_gain_enum[3]),
>>> +       SOC_ENUM("Line2L LADC Gain", aic3x_adc_gain_enum[4]),
>>> +       SOC_ENUM("Line2R RADC Gain", aic3x_adc_gain_enum[5]),
>> Please avoid using arrays as the driver no longer use these anymore.
>>
>>> +    SOC_ENUM("Headset Debounce", aic3x_headset_debounce_enum),
>>> +    SOC_ENUM("Button Debounce", aic3x_button_debounce_enum),
>>> +    SOC_SINGLE("Headset Detected", AIC3X_HEADSET_DETECT_CTRL_A,
>>> AIC3X_HEADSET_DETECT_SHIFT, AIC3X_HEADSET_DETECT_MASK, 0),
>>> +    SOC_SINGLE("Headset Detect Enable", AIC3X_HEADSET_DETECT_CTRL_A, 7, 1,
>>> 0),
>>> +    SOC_SINGLE("High power output Ac-coupled", AIC3X_HEADSET_DETECT_CTRL_B,
>>> 7, 1, 0),
>>> +    SOC_SINGLE("Stereo pseudodifferential output",
>>> AIC3X_HEADSET_DETECT_CTRL_B, 3, 1, 0),
>>>   };
>>>
>>>   static const struct snd_kcontrol_new aic3x_mono_controls[] = {
>>> @@ -614,7 +660,7 @@ static const struct snd_soc_dapm_widget
>>> aic3x_dapm_widgets[] = {
>>>          SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32",
>>>                           AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0),
>>>
>>> -       /* Mic Bias */
>>> +    /* Mic Bias */
>> nop change
>>
>>>          SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0,
>>>                           mic_bias_event,
>>>                           SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
>>> @@ -692,7 +738,7 @@ static const struct snd_soc_dapm_route intercon[] = {
>>>          {"Left Line2L Mux", "single-ended", "LINE2L"},
>>>          {"Left Line2L Mux", "differential", "LINE2L"},
>>>
>>> -       {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
>>> +    {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
>> nop change
>>
>>>          {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"},
>>>          {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"},
>>>          {"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
>>> @@ -814,6 +860,7 @@ static const struct snd_soc_dapm_route intercon[] = {
>>>          {"Right HPCOM Mux", "external feedback", "Right HPCOM Mixer"},
>>>          {"Right HP Com", NULL, "Right HPCOM Mux"},
>>>          {"HPRCOM", NULL, "Right HP Com"},
>>> +
>> why?
>>
>>>   };
>>>
>>>   static const struct snd_soc_dapm_route intercon_mono[] = {
>>> @@ -918,7 +965,9 @@ static int aic3x_hw_params(struct snd_pcm_substream
>>> *substream,
>>>          data = (LDAC2LCH | RDAC2RCH);
>>>          data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000;
>>>          if (params_rate(params) >= 64000)
>>> +       {
>>>                  data |= DUAL_RATE_MODE;
>>> +       }
>> why?
>>
>>>          snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data);
>>>
>>>          /* codec sample rate select */
>>> -- 
>>> and tlv320aic3x.h
>>> ---
>>>   sound/soc/codecs/tlv320aic3x.h | 6 ++++++
>>>   1 file changed, 6 insertions(+)
>>>
>>> diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
>>> index e521ac3..8ff42d2 100644
>>> --- a/sound/soc/codecs/tlv320aic3x.h
>>> +++ b/sound/soc/codecs/tlv320aic3x.h
>>> @@ -271,6 +271,12 @@ enum {
>>>          AIC3X_BUTTON_DEBOUNCE_32MS      = 3
>>>   };
>>>
>>> +typedef struct {
>>> +    struct platform_device* pdev;
>>> +    struct proc_dir_entry *proc_value;
>>> +    struct snd_soc_codec *codec;
>>> +} aic3106_detect_t;
>>> +
>>>   #define AIC3X_HEADSET_DETECT_ENABLED   0x80
>>>   #define AIC3X_HEADSET_DETECT_SHIFT     5
>>>   #define AIC3X_HEADSET_DETECT_MASK      3
>>> -- 
>>
>> When submitting please separate the patches by functionality. It is easier to
>> review the changes.
>>
> 
> 

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