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

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)

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(-)

sound/soc/codecs/tlv320aic3x.c +13 -0

index d7349bc..27d031e 100644
@@ -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,

sound/soc/codecs/tlv320aic3x.c +0 -0

        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),
 };
 

sound/soc/codecs/tlv320aic3x.c +11 -0

+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 */

sound/soc/codecs/tlv320aic3x.c +0 -0

@@ -399,6 +432,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[]
 = {

sound/soc/codecs/tlv320aic3x.c +6 -0

+    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[] = {

sound/soc/codecs/tlv320aic3x.c +0 -0

        snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data);
 

        /* codec sample rate select */

Signature · 4 lines

and tlv320aic3x.h
---
 sound/soc/codecs/tlv320aic3x.h | 6 ++++++
 1 file changed, 6 insertions(+)

sound/soc/codecs/tlv320aic3x.h +0 -0

index e521ac3..8ff42d2 100644
@@ -271,6 +271,12 @@ enum {
        AIC3X_BUTTON_DEBOUNCE_32MS      = 3
 };

Hi, Peter!
What's about placing capture controls in the Playback section of Alsamixer view?

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

14.03.2016 17:47, Peter Ujfalusi пишет:

Quoted 237 lines

Hi Timur,

Quoted 6 lines · Timur Karaldin

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.

Quoted 64 lines

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(-)

sound/soc/codecs/tlv320aic3x.c +13 -0

index d7349bc..27d031e 100644
@@ -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,

sound/soc/codecs/tlv320aic3x.c static int mic_bias_event(struct snd_soc_dapm_widget *w, · line 212

Quoted 5 lines

sound/soc/codecs/tlv320aic3x.c +1 -1

@@ -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...

Quoted 6 lines

sound/soc/codecs/tlv320aic3x.c +0 -0

         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),
  };
 

sound/soc/codecs/tlv320aic3x.c static const struct soc_enum aic3x_agc_decay_enum[] = { · line 295

Quoted 10 lines

sound/soc/codecs/tlv320aic3x.c +9 -0

+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.

Quoted 20 lines

sound/soc/codecs/tlv320aic3x.c +5 -0

+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" };

sound/soc/codecs/tlv320aic3x.c +2 -0

+static const struct soc_enum aic3x_button_debounce_enum[] = {
+    SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 0, 4,

aic3x_button_debounce),

sound/soc/codecs/tlv320aic3x.c +4 -0

+};
+
+
+
  /*
   * 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[]
 = {

sound/soc/codecs/tlv320aic3x.c static const struct snd_kcontrol_new aic3x_snd_controls[] · line 442

Quoted 10 lines

sound/soc/codecs/tlv320aic3x.c +7 -0

         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.

Quoted 14 lines

sound/soc/codecs/tlv320aic3x.c +5 -0

+    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[] = {

sound/soc/codecs/tlv320aic3x.c static const struct snd_soc_dapm_widget · line 664

Quoted 6 lines

sound/soc/codecs/tlv320aic3x.c +1 -1

@@ -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

sound/soc/codecs/tlv320aic3x.c static const struct snd_soc_dapm_route intercon[] = { · line 741

Quoted 9 lines

sound/soc/codecs/tlv320aic3x.c +0 -0

         SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0,
                          mic_bias_event,

SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),

sound/soc/codecs/tlv320aic3x.c +1 -1

@@ -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

sound/soc/codecs/tlv320aic3x.c static const struct snd_soc_dapm_route intercon[] = { · line 863

Quoted 8 lines

sound/soc/codecs/tlv320aic3x.c +1 -0

         {"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?

sound/soc/codecs/tlv320aic3x.c static int aic3x_hw_params(struct snd_pcm_substream · line 971

Quoted 11 lines

sound/soc/codecs/tlv320aic3x.c +2 -0

  };
 
  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?

Quoted 17 lines

sound/soc/codecs/tlv320aic3x.c +0 -0

         snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data);
 

         /* codec sample rate select */

Signature · 4 lines

and tlv320aic3x.h
---
  sound/soc/codecs/tlv320aic3x.h | 6 ++++++
  1 file changed, 6 insertions(+)

sound/soc/codecs/tlv320aic3x.h +0 -0

index e521ac3..8ff42d2 100644
@@ -271,6 +271,12 @@ enum {
         AIC3X_BUTTON_DEBOUNCE_32MS      = 3
  };

sound/soc/codecs/tlv320aic3x.h enum { · line 282

Quoted 10 lines

sound/soc/codecs/tlv320aic3x.h +6 -0

 
+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.

Quoted 3 lines

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...

Quoted 2 lines

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

Not sure what you mean.

14.03.2016 19:38, Peter Ujfalusi пишет:

Quoted 10 lines · Timur Karaldin

Quoted 3 lines

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...

Quoted 4 lines

Quoted 2 lines

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

Not sure what you mean.

I mean current situation:

#i2cdump -f -y -r 13-13 1 0x1b
Result: 0x00
#arecord test.wav &
#i2cdump -f -y -r 13-13 1 0x1b
Result: 0x00

# amixer sset 'Headset Detect Enable' *on*
#i2cdump -f -y -r 13-13 1 0x1b
Result:*0x80*
#killall arecord
#i2cdump -f -y -r 13-13 1 0x1b
Result: 0x00
#arecord test.wav &
#i2cdump -f -y -r 13-13 1 0x1b
Result:*0x00
*#killall arecord*
*amixer sset 'Headset Detect Enable'*on**
*#i2cdump -f -y -r 13-13 1 0x1b
Result:**0x00
**#amixer sset 'Headset Detect Enable' off
*amixer: Invalid command!*

So state of 'Headset Detect Enable' and others of my controls were not stored during arecord/aplay restart (open and close device).

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.

Hi Peter!

14.03.2016 17:47, Peter Ujfalusi пишет:

Quoted 5 lines

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

Quoted 8 lines

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.

That's not clear for me. As I understand, I need to create SOC_SINGLE_TLV() for each gain I would like to change, with the same name as SOC_DAPM_SINGLE_AIC3x (for example "Line1L Switch" for Line1) and make callbacks with cached gain values. Am I right?

May be the better sollution is to change SOC_DAPM_SINGLE_AIC3x on/off switch to other type with switch and gain control together? So it could have some connected states with different gains and disconnected state (Something like volume/mute control).

Quoted 16 lines

Hi Peter!

14.03.2016 17:47, Peter Ujfalusi пишет:

Quoted 5 lines

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

Quoted 8 lines

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.

Quoted 4 lines

That's not clear for me. As I understand, I need to create SOC_SINGLE_TLV() for each gain I would like to change, with the same name as SOC_DAPM_SINGLE_AIC3x (for example "Line1L Switch" for Line1) and make callbacks with cached gain values. Am I right?

Mostly yes, but you need to name the controls:
existing switch + new gain control
"Line1L Switch" + "Line1L Volume"
"Line1R Switch" + "Line1R Volume"

In this way alsamixer can combine them.

Quoted 4 lines

May be the better sollution is to change SOC_DAPM_SINGLE_AIC3x on/off switch to other type with switch and gain control together? So it could have some connected states with different gains and disconnected state (Something like volume/mute control).

Something based on SOC_DAPM_SINGLE_TLV should work. But that means again change in Kcontrol names.

Ok, I choose the way #1 without changing kcontrol names.
Another thing I don't understand is the way for caching gain volume if I have switches muted. Is there any proper way for storing cached values between gain and switch handlers includes soft reset or I need manually check kcontrol name in handler and cache values in extended private structure? The same question for switch handler, because I need to share cached values between these handlers.

15.03.2016 15:40, Peter Ujfalusi пишет:

Quoted 35 lines · Timur Karaldin

Quoted 16 lines

Hi Peter!

14.03.2016 17:47, Peter Ujfalusi пишет:

Quoted 5 lines

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

Quoted 8 lines

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.

Quoted 4 lines

That's not clear for me. As I understand, I need to create SOC_SINGLE_TLV() for each gain I would like to change, with the same name as SOC_DAPM_SINGLE_AIC3x (for example "Line1L Switch" for Line1) and make callbacks with cached gain values. Am I right?

Mostly yes, but you need to name the controls:
existing switch + new gain control
"Line1L Switch" + "Line1L Volume"
"Line1R Switch" + "Line1R Volume"

In this way alsamixer can combine them.

Quoted 4 lines

May be the better sollution is to change SOC_DAPM_SINGLE_AIC3x on/off switch to other type with switch and gain control together? So it could have some connected states with different gains and disconnected state (Something like volume/mute control).

Something based on SOC_DAPM_SINGLE_TLV should work. But that means again change in Kcontrol names.

Hi Timur,

please avoid top posting.

Quoted 6 lines · Timur Karaldin

Ok, I choose the way #1 without changing kcontrol names.
Another thing I don't understand is the way for caching gain volume if I have switches muted. Is there any proper way for storing cached values between gain and switch handlers includes soft reset or I need manually check kcontrol name in handler and cache values in extended private structure? The same question for switch handler, because I need to share cached values between these handlers.

Since the problematic registers are: 17, 18, .. 24. I would have an array inside struct aic3x_priv to store the configured gain. You could store the mute as well, but since regmap is caching the content I would simply read back the gain value and if it is 0xf, it is muted, so update only the local shadow, if it is not muted, write the gain.
In the mute/unmute controls I would then save the gain to shadow when muting the path and write the cached gain to the register when unmuting it. You should also return correct 0/1 for the unmute state and not the actual register value.

Hi Peter!

16.03.2016 11:57, Peter Ujfalusi пишет:

Quoted 21 lines

Hi Timur,

please avoid top posting.

Quoted 7 lines · Timur Karaldin

Ok, I choose the way #1 without changing kcontrol names.
Another thing I don't understand is the way for caching gain volume if I have switches muted. Is there any proper way for storing cached values between gain and switch handlers includes soft reset or I need manually check kcontrol name in handler and cache values in extended private structure? The same question for switch handler, because I need to share cached values between these handlers.

Since the problematic registers are: 17, 18, .. 24. I would have an array inside struct aic3x_priv to store the configured gain. You could store the mute as well, but since regmap is caching the content I would simply read back the gain value and if it is 0xf, it is muted, so update only the local shadow, if it is not muted, write the gain.
In the mute/unmute controls I would then save the gain to shadow when muting the path and write the cached gain to the register when unmuting it. You should also return correct 0/1 for the unmute state and not the actual register value.

Actually I try to avoid complex handler, with switch-case which depends on reg number or kcontrol name inside of it.
I have found similar discussion with you about one year ago here
http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087430.html. There is a proposal to use TLV_DB_RANGE_HEAD() and TLV_DB_SCALE_ITEM() because of invert with SOC_DAPM_SINGLE_TLV.

As I understand we could make tlv with using
0 as mute, skip values 1-6, use values 7-15 (7 = -12dB and 15=0dB) then declare SOC_DAPM_SINGLE_TLV as "Line1L Volume" for example. Is it still actual?

Few things which I didn't understand:
- should I use "Line1L Switch" together with "Line1L Volume" in this case. I think no, because DAPM_SINGLE_TLV should replace switch.

- i don't understand how declare tlv with skipping values 1-6

Cheers, Tim

Quoted 24 lines

Hi Peter!

16.03.2016 11:57, Peter Ujfalusi пишет:

Quoted 21 lines

Hi Timur,

please avoid top posting.

Quoted 7 lines · Timur Karaldin

Ok, I choose the way #1 without changing kcontrol names.
Another thing I don't understand is the way for caching gain volume if I have switches muted. Is there any proper way for storing cached values between gain and switch handlers includes soft reset or I need manually check kcontrol name in handler and cache values in extended private structure? The same question for switch handler, because I need to share cached values between these handlers.

Since the problematic registers are: 17, 18, .. 24. I would have an array inside struct aic3x_priv to store the configured gain. You could store the mute as well, but since regmap is caching the content I would simply read back the gain value and if it is 0xf, it is muted, so update only the local shadow, if it is not muted, write the gain.
In the mute/unmute controls I would then save the gain to shadow when muting the path and write the cached gain to the register when unmuting it. You should also return correct 0/1 for the unmute state and not the actual register value.

your mail client does not seem to wrap the lines correctly, can you check that.

Quoted 7 lines

Actually I try to avoid complex handler, with switch-case which depends on reg number or kcontrol name inside of it.
I have found similar discussion with you about one year ago here
http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087430.html. There is a proposal to use TLV_DB_RANGE_HEAD() and TLV_DB_SCALE_ITEM() because of invert with SOC_DAPM_SINGLE_TLV.

As I recall there were some issues with that back then, but can not recall. At least I don't see followup patches regarding to that.

Quoted 4 lines

As I understand we could make tlv with using
0 as mute, skip values 1-6, use values 7-15 (7 = -12dB and 15=0dB) then declare SOC_DAPM_SINGLE_TLV as "Line1L Volume" for example. Is it still actual?

Worth a try ;)
The naming needs to be different, take a look at twl4030 codec's twl4030_dapm_dbypass_tlv and how it is used.

Quoted 3 lines

Few things which I didn't understand:
- should I use "Line1L Switch" together with "Line1L Volume" in this case. I think no, because DAPM_SINGLE_TLV should replace switch.

Since we will have the gain and mute with volume control we will have only one control.

Quoted 1 line

- i don't understand how declare tlv with skipping values 1-6

Yeah, I'm not sure how leaving 'hole' between the TLV_DB_SCALE_ITEMs will be handled if it works.

Quoted 1 line

Cheers, Tim

Quoted 1 line

your mail client does not seem to wrap the lines correctly, can you check that.

I have no idea how these lines should looks, so it's very hard for me to see what's wrong. Could you point me how it looks in original?

Ok, now it's much more clear for me.
Another question is register behaviour during soft reset. There is "aic3x_set_power" handle. In this handle kernel makes SOFT_RESET, markes cache as dirty, then power down the codec for handle power down request. But as I could see main volumes like "HP DAC" and "PCM" stores values between close and open in mixer and I could not see any code to handle it. On the other hand my controls do not save states, as you mentioned because of SOFT_RESET, could you explain such different behaviour?

Quoted 8 lines

Ok, now it's much more clear for me.
Another question is register behaviour during soft reset. There is "aic3x_set_power" handle. In this handle kernel makes SOFT_RESET, markes cache as dirty, then power down the codec for handle power down request. But as I could see main volumes like "HP DAC" and "PCM" stores values between close and open in mixer and I could not see any code to handle it. On the other hand my controls do not save states, as you mentioned because of SOFT_RESET, could you explain such different behaviour?

All cached registers are going to be restored after power on with exception of volatile registers. You need to restore the bits in a volatile registers in the driver.

Quoted 3 lines

Quoted 1 line

your mail client does not seem to wrap the lines correctly, can you check that.

I have no idea how these lines should looks, so it's very hard for me to see what's wrong. Could you point me how it looks in original?

They are looong.

see: https://wiki.openstack.org/wiki/MailingListEtiquette

Quoted 8 lines

Ok, now it's much more clear for me.
Another question is register behaviour during soft reset. There is "aic3x_set_power" handle. In this handle kernel makes SOFT_RESET, markes cache as dirty, then power down the codec for handle power down request. But as I could see main volumes like "HP DAC" and "PCM" stores values between close and open in mixer and I could not see any code to handle it. On the other hand my controls do not save states, as you mentioned because of SOFT_RESET, could you explain such different behaviour?

All cached registers are going to be restored after power on with exception of volatile registers. You need to restore the bits in a volatile registers in the driver.

Hi Peter!
Here is my patch v2. It works only for aic3106, but I don't have any others. Also patch contains some debug output, I would clean it. I like behaviour of this improvements and I didn't know any issue right now for these add-ons except debug output and that it's correct only for aic3106 (didn't check for others even on datasheet level). Thank you for all your help!

sound/soc/codecs/tlv320aic3x.c +17 -1

 };
 
 static const struct reg_default aic3x_reg[] = {

@@ -134,16 +135,175 @@ static const struct regmap_config aic3x_regmap = {
 
 #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
        SOC_SINGLE_EXT(xname, reg, shift, mask, invert, \
-               snd_soc_dapm_get_volsw, snd_soc_dapm_put_volsw_aic3x)
+               snd_soc_dapm_get_volsw_aic3x, snd_soc_dapm_put_volsw_aic3x)
+
+#define SOC_DOUBLE_R_AIC3X_TLV(xname, reg, rreg, shift, mask, invert) \
+       SOC_DOUBLE_R_EXT_TLV(xname, reg, rreg, shift, mask, invert, \
+       snd_soc_get_gain_aic3x, snd_soc_put_gain_aic3x, gain_stage_tlv)
+
+#define SOC_SINGLE_AIC3X_TLV(xname, reg, shift, mask, invert) \
+       SOC_SINGLE_EXT_TLV(xname, reg, shift, mask, invert, \
+       snd_soc_get_gain_aic3x, snd_soc_put_gain_aic3x, gain_stage_tlv)
+
+#define        SOC_SINGLE_EXT_VOL(xname, reg, shift, mask, invert) \
+       SOC_SINGLE_EXT(xname, reg, shift, mask, invert, \
+               snd_soc_get_volsw_uncached, snd_soc_put_volsw)
+
+/*
+ * Headset detect flag, button press detect flag and headset type flag is
 read only register,
+ * but we could not declare these regs as volatile because other bits are RW,

so we use
+ * unchached get hanler for these regs.

sound/soc/codecs/tlv320aic3x.c +16 -0

+*/
+static int snd_soc_get_volsw_uncached( struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value * ucontrol)
+{
+       int ret;
+       struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+       struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+
+       regcache_cache_bypass(aic3x->regmap, true);
+       ret = snd_soc_get_volsw(kcontrol, ucontrol);
+       regcache_cache_bypass(aic3x->regmap, false);
+       return ret;
+}
+
+/*
+ * All input lines have additional gain controls which can be controled by

volume control.
+ * 0x0 - means 0dB and 0x8 - means -12 dB. It's a litlle bit tricky because value 0xF means mute.
+ * value 0x9-0xE reserverd, so in case of switch muted we should store value in cache and should
+ * not set up it to register

sound/soc/codecs/tlv320aic3x.c +4 -0

+{
+       struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+       struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+       struct soc_mixer_control *mc = (struct soc_mixer_control

*)kcontrol->private_value;

sound/soc/codecs/tlv320aic3x.c +10 -0

+       unsigned int reg;
+       unsigned int shift = mc->shift;
+       int max = mc->max;
+       unsigned int mask = (1<<fls(max))-1;
+       unsigned int invert = mc->invert;
+
+       u8 regdata, newval, regval;
+        u8 loop = 0;
+
+       //First of all we need to cache value. Then we need to test each value

on mute in register

sound/soc/codecs/tlv320aic3x.c +4 -0

+       //If not muted, we should setup new value
+       reg = mc->reg;
+        newval = ( ucontrol->value.integer.value[0] & mask );
+       printk(KERN_INFO "set gain control reg:%d val:%0X mask:%0X shift:%0X

invert %d\n", reg,newval, mask, shift,invert);

sound/soc/codecs/tlv320aic3x.c +12 -0

+               if(invert)
+               {
+                       newval = ( 8 - newval ) & mask;
+                       printk(KERN_INFO "inverted val %0X\n", newval);
+               }
+               //read, test and update if first reg if needs
+               regdata = snd_soc_read(codec, reg);
+               regval = ( regdata >> shift) & mask;
+
+               if( reg >= MIC3LR_2_LADC_CTRL && reg <= LINE1L_2_RADC_CTRL )
+               {
+                       printk(KERN_INFO "cached_gain before

%0X\n",aic3x->cached_gain[reg-MIC3LR_2_LADC_CTRL]);
+ aic3x->cached_gain[reg-MIC3LR_2_LADC_CTRL]&=~(mask<<shift);
+ aic3x->cached_gain[reg-MIC3LR_2_LADC_CTRL]|=(newval<<shift);
+ printk(KERN_INFO "cached_gain after
%0X\n",aic3x->cached_gain[reg-MIC3LR_2_LADC_CTRL]);

sound/soc/codecs/tlv320aic3x.c +4 -0

+               }
+
+               if ( regval!=0xf && regval!=newval )
+                       snd_soc_update_bits( codec, reg, mask<<shift,

newval<<shift );

sound/soc/codecs/tlv320aic3x.c +4 -0

+
+                if(snd_soc_volsw_is_stereo(mc))
+                {
+                        loop=!loop;     //switch loop, so for first time loop

will be true after and for second time it will be false, so we will finish loop

sound/soc/codecs/tlv320aic3x.c +4 -0

+                        if(loop) //if first time
+                        {
+                                reg=mc->rreg;
+                                newval = (ucontrol->value.integer.value[1] &

mask);
+ printk(KERN_INFO "control is stereo second
reg:%d val:%0X\n", reg,newval);

sound/soc/codecs/tlv320aic3x.c +10 -0

+                                continue;
+                        }
+                }
+
+        }while(loop);
+       return 0;
+}
+
+/*
+ * All input lines have additional gain controls which can be controled by

volume control.
+ * 0x0 - means 0dB and 0x8 - means -12 dB. It's a litlle bit tricky because value 0xF means mute.
+ * value 0x9-0xE reserverd, so in case of switch muted we should read value from cache in aic3x_priv
+ * and should not read it from regcache or register

sound/soc/codecs/tlv320aic3x.c +7 -0

+*/
+static int snd_soc_get_gain_aic3x( struct snd_kcontrol *kcontrol,
+                                  struct snd_ctl_elem_value *ucontrol)
+{
+       struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+       struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+       struct soc_mixer_control *mc = (struct soc_mixer_control

*)kcontrol->private_value;

sound/soc/codecs/tlv320aic3x.c +10 -0

+       unsigned int reg;
+       unsigned int shift = mc->shift;
+       int max = mc->max;
+       unsigned int mask = (1<<fls(max))-1;
+       unsigned int invert = mc->invert;
+
+       u8 regdata, regval;
+        u8 loop = 0;
+
+       //First of all we need to cache value. Then we need to test each value

on mute in register

sound/soc/codecs/tlv320aic3x.c +4 -0

+       //If not muted, we should set up new value
+       reg = mc->reg;
+
+       printk(KERN_INFO "get gain control reg:%d mask:%0X shift:%0X invert

%d\n", reg, mask, shift,invert);

sound/soc/codecs/tlv320aic3x.c +8 -0

+               regdata = snd_soc_read(codec, reg);
+               regval = ( regdata >> shift) & mask;
+               printk(KERN_INFO "regval:%0X\n", regval);
+
+               //check if register is muted then return cached value
+               if( regval == 0xf )
+               {
+                       if( reg >=MIC3LR_2_LADC_CTRL && reg <=

LINE1L_2_RADC_CTRL )
+ regval =
(aic3x->cached_gain[reg-MIC3LR_2_LADC_CTRL]>>shift) & mask;

sound/soc/codecs/tlv320aic3x.c +13 -0

+                       printk(KERN_INFO "regval from cache:%0X\n", regval);
+
+               }
+               if( invert )
+               {
+                       regval = (8 - regval) & mask;
+                       printk(KERN_INFO "regval after invert:%0X\n", regval);
+               }
+               ucontrol->value.integer.value[loop] = regval;
+
+                if(snd_soc_volsw_is_stereo(mc))
+                {
+                        loop=loop?0:1; //switch loop, so for first time loop

will be 1 after and for second time it will be 0, so we will exit from while-loop

sound/soc/codecs/tlv320aic3x.c +4 -0

+                        if(loop) //if first time
+                        {
+                                reg=mc->rreg;
+                               printk(KERN_INFO "control is stereo second

reg:%d\n", reg);

sound/soc/codecs/tlv320aic3x.c +17 -3

+                                continue;
+                        }
+                }
+
+        }while(loop);
+       return 0;
+}
 
 /*
  * All input lines are connected when !0xf and disconnected with 0xf bit field,
- * so we have to use specific dapm_put call for input mixer
+ * so we have to use specific dapm_put call for input mixer. In case of unmute
+ * we should set up register value from cached values from aic3x_priv struct.
  */
 static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
                                        struct snd_ctl_elem_value *ucontrol)
 {
        struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+       struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        unsigned int reg = mc->reg;

@@ -151,20 +311,32 @@ static int snd_soc_dapm_put_volsw_aic3x(struct
 snd_kcontrol *kcontrol,
        int max = mc->max;
        unsigned int mask = (1 << fls(max)) - 1;
        unsigned int invert = mc->invert;
-       unsigned short val;
+       unsigned int val;
        struct snd_soc_dapm_update update;
        int connect, change;
 
        val = (ucontrol->value.integer.value[0] & mask);
+       printk(KERN_INFO "dapm put volsw val:%X reg:%X mask:%X shift:%X
 invert:%X\n", val, reg, mask, shift, invert);
 
        mask = 0xf;
        if (val)
+       {
                val = mask;
-
+               printk(KERN_INFO "new val:%X\n", val);
+       }
        connect = !!val;
 
        if (invert)
+       {

        mask <<= shift;
        val <<= shift;

sound/soc/codecs/tlv320aic3x.c +16 -0

@@ -183,6 +355,36 @@ static int snd_soc_dapm_put_volsw_aic3x(struct
 snd_kcontrol *kcontrol,
        return change;
 }
 
+/*
+ * Based on standart handler snd_soc_dapm_get_volsw, but changing mask to 0xF
+ *
+ */
+static int snd_soc_dapm_get_volsw_aic3x(struct snd_kcontrol *kcontrol,
+                                       struct snd_ctl_elem_value *ucontrol)
+{
+       struct soc_mixer_control *mc =
+               (struct soc_mixer_control *)kcontrol->private_value;
+       unsigned int reg = mc->reg;
+       unsigned int shift = mc->shift;
+       int max = mc->max;
+       unsigned int mask = (1 << fls(max)) - 1;
+       int ret;
+
+       //register contains 4 bits, so we change max temoprally to read

register by original handler, then return it back

sound/soc/codecs/tlv320aic3x.c +3 -0

+       mc->max = 15;
+       ret = snd_soc_dapm_get_volsw( kcontrol, ucontrol);
+       printk(KERN_INFO "soc_dapm_get_volsw_aic3x:%ld reg:%X shift:%X max:%X

newmax:%X\n", ucontrol->value.integer.value[0], reg, shift, max, mc->max );

sound/soc/codecs/tlv320aic3x.c +3 -0

+       mc->max = max;
+
+       //all gains except mute (0xf) after invertion is not equl to 0, so we

need to set 1(on)

sound/soc/codecs/tlv320aic3x.c +16 -0

+       //for all values except 0. 0 is mute, so we do not need to change it
+       if(ucontrol->value.integer.value[0]!=0)
+               ucontrol->value.integer.value[0]=1;
+       printk(KERN_INFO "new value:%ld\n", ucontrol->value.integer.value[0]);
+
+       return ret;
+}
+
 /*
  * mic bias power on/off share the same register bits with
  * output voltage of mic bias. when power on mic bias, we

@@ -270,6 +472,14 @@ static const struct soc_enum aic3x_agc_decay_enum[] = {
        SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay),
 };
 
+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

*/

sound/soc/codecs/tlv320aic3x.c +7 -0

@@ -287,6 +497,9 @@ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0);
  */
 static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1);
 
+static DECLARE_TLV_DB_SCALE(gain_stage_tlv, -1200, 150, 0);
+
+
 static const struct snd_kcontrol_new aic3x_snd_controls[] = {
        /* Output */
        SOC_DOUBLE_R_TLV("PCM Playback Volume",

@@ -399,6 +612,24 @@ 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_DOUBLE_R_AIC3X_TLV("Mic3L Volume", MIC3LR_2_LADC_CTRL,
 MIC3LR_2_RADC_CTRL, 4, 8, 1),
+       SOC_DOUBLE_R_AIC3X_TLV("Mic3R Volume", MIC3LR_2_LADC_CTRL,
 MIC3LR_2_RADC_CTRL, 0, 8, 1),
+       SOC_DOUBLE_R_AIC3X_TLV("Line1L Volume", LINE1L_2_LADC_CTRL,

LINE1L_2_RADC_CTRL, 3, 8, 1),
+ SOC_DOUBLE_R_AIC3X_TLV("Line1R Volume", LINE1R_2_LADC_CTRL,
LINE1R_2_RADC_CTRL, 3, 8, 1),

sound/soc/codecs/tlv320aic3x.c +5 -0

+       SOC_SINGLE_AIC3X_TLV("Line2L Volume", LINE2L_2_LADC_CTRL, 3, 8, 1),
+       SOC_SINGLE_AIC3X_TLV("Line2R Volume", LINE2R_2_RADC_CTRL, 3, 8, 1),
+       SOC_ENUM("Headset Jack Debounce", aic3x_headset_debounce_enum ),
+       SOC_ENUM("Button Press Debounce", aic3x_button_debounce_enum ),
+       SOC_SINGLE("Headset Detect Enable", AIC3X_HEADSET_DETECT_CTRL_A, 7, 1,

0),

sound/soc/codecs/tlv320aic3x.c +7 -0

+       SOC_SINGLE_EXT_VOL("Headset Detect Type", AIC3X_HEADSET_DETECT_CTRL_A,
+               AIC3X_HEADSET_DETECT_A_SHIFT, AIC3X_HEADSET_DETECT_A_MASK, 0),
+       SOC_SINGLE_EXT_VOL("Button Detect Flag", AIC3X_HEADSET_DETECT_CTRL_B,
+               AIC3X_BUTTON_DETECT_SHIFT, AIC3X_BUTTON_DETECT_MASK, 0),
+       SOC_SINGLE_EXT_VOL("Headset Detect Flag", AIC3X_HEADSET_DETECT_CTRL_B,
+               AIC3X_HEADSET_DETECT_B_SHIFT, AIC3X_HEADSET_DETECT_B_MASK, 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[] = {
--------------------------------------

sound/soc/codecs/tlv320aic3x.h +7 -2

index e521ac3..bcac87f 100644
@@ -271,9 +271,20 @@ 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
+#define AIC3X_HEADSET_DETECT_A_SHIFT   5