[PATCH] ASoC: tlv320aic31xx: add explicit support for tlv320dac31xx
From: Nikita Yushchenko
Date: Fri Sep 23 2016 - 07:52:52 EST
tlv320dac31xx is a subset of tlv320aic31xx:
- it does not have MIC inputs and ADC, thus capture is not supported,
- it has analog inputs AIN1/AIN2 that can be mixed into output.
Although tlv320dac31xx does work with tlv320aic31xx driver, this setup
does register non-existent widgets and non-existent capture stream.
Thus userspace lists non-existent objects in user interfaces, an can
access these, causing operations with device registers that are
declared as "reserved" in tlv320dac31xx datasheet.
This patch fixes this situation by separating controls/widgets/routes
into common, aic31xx-specific, and dac31xx-specific parts. Only parts
that match actual hardware (as declared in "compatible" device tree
property) are registered.
Changes from v1:
- update device tree binding documentation,
- rebased on top of "ASoC: codec duplicated callback function goes to
component on tlv320aic31xx" commit.
Signed-off-by: Nikita Yushchenko <nikita.yoush@xxxxxxxxxxxxxxxxxx>
Signed-off-by: Mark Brown <broonie@xxxxxxxxxx>
---
.../devicetree/bindings/sound/tlv320aic31xx.txt | 9 +-
sound/soc/codecs/tlv320aic31xx.c | 212 +++++++++++++++------
sound/soc/codecs/tlv320aic31xx.h | 2 +
3 files changed, 164 insertions(+), 59 deletions(-)
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
index eff12be5e789..9340d2ddcc54 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
@@ -11,6 +11,7 @@ Required properties:
"ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
"ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
"ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
+ "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP)
- reg - <int> - I2C slave address
- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
@@ -37,9 +38,11 @@ CODEC output pins:
* MICBIAS
CODEC input pins:
- * MIC1LP
- * MIC1RP
- * MIC1LM
+ * MIC1LP, devices with ADC
+ * MIC1RP, devices with ADC
+ * MIC1LM, devices with ADC
+ * AIN1, devices without ADC
+ * AIN2, devices without ADC
The pins can be used in referring sound node's audio-routing property.
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index e46fb472e48d..725173b12725 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -273,10 +273,20 @@ static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0);
/*
* controls to be exported to the user space
*/
-static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
+static const struct snd_kcontrol_new common31xx_snd_controls[] = {
SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL,
AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv),
+ SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
+ AIC31XX_HPRGAIN, 2, 1, 0),
+ SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
+ AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),
+
+ SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
+ AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1,
adc_fgain_tlv),
@@ -286,14 +296,6 @@ static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0,
119, 0, mic_pga_tlv),
-
- SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
- AIC31XX_HPRGAIN, 2, 1, 0),
- SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
- AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),
-
- SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
- AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
};
static const struct snd_kcontrol_new aic311x_snd_controls[] = {
@@ -397,17 +399,28 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const struct snd_kcontrol_new left_output_switches[] = {
+static const struct snd_kcontrol_new aic31xx_left_output_switches[] = {
SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0),
SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0),
};
-static const struct snd_kcontrol_new right_output_switches[] = {
+static const struct snd_kcontrol_new aic31xx_right_output_switches[] = {
SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0),
};
+static const struct snd_kcontrol_new dac31xx_left_output_switches[] = {
+ SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
+ SOC_DAPM_SINGLE("From AIN1", AIC31XX_DACMIXERROUTE, 5, 1, 0),
+ SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new dac31xx_right_output_switches[] = {
+ SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
+ SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 1, 1, 0),
+};
+
static const struct snd_kcontrol_new p_term_mic1lp =
SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum);
@@ -457,7 +470,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget common31xx_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("DAC Left Input",
@@ -473,14 +486,7 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
- /* Output Mixers */
- SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
- left_output_switches,
- ARRAY_SIZE(left_output_switches)),
- SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
- right_output_switches,
- ARRAY_SIZE(right_output_switches)),
-
+ /* HP */
SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0,
&aic31xx_dapm_hpl_switch),
SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0,
@@ -494,10 +500,34 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
NULL, 0, aic31xx_dapm_power_event,
SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
- /* ADC */
- SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
- aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_POST_PMD),
+ /* Mic Bias */
+ SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+static const struct snd_soc_dapm_widget dac31xx_dapm_widgets[] = {
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("AIN1"),
+ SND_SOC_DAPM_INPUT("AIN2"),
+
+ /* Output Mixers */
+ SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
+ dac31xx_left_output_switches,
+ ARRAY_SIZE(dac31xx_left_output_switches)),
+ SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
+ dac31xx_right_output_switches,
+ ARRAY_SIZE(dac31xx_right_output_switches)),
+};
+
+static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("MIC1LP"),
+ SND_SOC_DAPM_INPUT("MIC1RP"),
+ SND_SOC_DAPM_INPUT("MIC1LM"),
/* Input Selection to MIC_PGA */
SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0,
@@ -507,24 +537,25 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0,
&p_term_mic1lm),
+ /* ADC */
+ SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
+ aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0,
&m_term_mic1lm),
+
/* Enabling & Disabling MIC Gain Ctl */
SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA,
7, 1, NULL, 0),
- /* Mic Bias */
- SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-
- /* Outputs */
- SND_SOC_DAPM_OUTPUT("HPL"),
- SND_SOC_DAPM_OUTPUT("HPR"),
-
- /* Inputs */
- SND_SOC_DAPM_INPUT("MIC1LP"),
- SND_SOC_DAPM_INPUT("MIC1RP"),
- SND_SOC_DAPM_INPUT("MIC1LM"),
+ /* Output Mixers */
+ SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
+ aic31xx_left_output_switches,
+ ARRAY_SIZE(aic31xx_left_output_switches)),
+ SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
+ aic31xx_right_output_switches,
+ ARRAY_SIZE(aic31xx_right_output_switches)),
};
static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = {
@@ -554,7 +585,7 @@ static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route
-aic31xx_audio_map[] = {
+common31xx_audio_map[] = {
/* DAC Input Routing */
{"DAC Left Input", "Left Data", "DAC IN"},
{"DAC Left Input", "Right Data", "DAC IN"},
@@ -565,6 +596,31 @@ aic31xx_audio_map[] = {
{"DAC Left", NULL, "DAC Left Input"},
{"DAC Right", NULL, "DAC Right Input"},
+ /* HPL path */
+ {"HP Left", "Switch", "Output Left"},
+ {"HPL Driver", NULL, "HP Left"},
+ {"HPL", NULL, "HPL Driver"},
+
+ /* HPR path */
+ {"HP Right", "Switch", "Output Right"},
+ {"HPR Driver", NULL, "HP Right"},
+ {"HPR", NULL, "HPR Driver"},
+};
+
+static const struct snd_soc_dapm_route
+dac31xx_audio_map[] = {
+ /* Left Output */
+ {"Output Left", "From Left DAC", "DAC Left"},
+ {"Output Left", "From AIN1", "AIN1"},
+ {"Output Left", "From AIN2", "AIN2"},
+
+ /* Right Output */
+ {"Output Right", "From Right DAC", "DAC Right"},
+ {"Output Right", "From AIN2", "AIN2"},
+};
+
+static const struct snd_soc_dapm_route
+aic31xx_audio_map[] = {
/* Mic input */
{"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"},
{"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"},
@@ -595,16 +651,6 @@ aic31xx_audio_map[] = {
/* Right Output */
{"Output Right", "From Right DAC", "DAC Right"},
{"Output Right", "From MIC1RP", "MIC1RP"},
-
- /* HPL path */
- {"HP Left", "Switch", "Output Left"},
- {"HPL Driver", NULL, "HP Left"},
- {"HPL", NULL, "HPL Driver"},
-
- /* HPR path */
- {"HP Right", "Switch", "Output Right"},
- {"HPR Driver", NULL, "HP Right"},
- {"HPR", NULL, "HPR Driver"},
};
static const struct snd_soc_dapm_route
@@ -633,6 +679,13 @@ static int aic31xx_add_controls(struct snd_soc_codec *codec)
int ret = 0;
struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ if (!(aic31xx->pdata.codec_type & DAC31XX_BIT))
+ ret = snd_soc_add_codec_controls(
+ codec, aic31xx_snd_controls,
+ ARRAY_SIZE(aic31xx_snd_controls));
+ if (ret)
+ return ret;
+
if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT)
ret = snd_soc_add_codec_controls(
codec, aic311x_snd_controls,
@@ -651,6 +704,30 @@ static int aic31xx_add_widgets(struct snd_soc_codec *codec)
struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
int ret = 0;
+ if (aic31xx->pdata.codec_type & DAC31XX_BIT) {
+ ret = snd_soc_dapm_new_controls(
+ dapm, dac31xx_dapm_widgets,
+ ARRAY_SIZE(dac31xx_dapm_widgets));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, dac31xx_audio_map,
+ ARRAY_SIZE(dac31xx_audio_map));
+ if (ret)
+ return ret;
+ } else {
+ ret = snd_soc_dapm_new_controls(
+ dapm, aic31xx_dapm_widgets,
+ ARRAY_SIZE(aic31xx_dapm_widgets));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, aic31xx_audio_map,
+ ARRAY_SIZE(aic31xx_audio_map));
+ if (ret)
+ return ret;
+ }
+
if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) {
ret = snd_soc_dapm_new_controls(
dapm, aic311x_dapm_widgets,
@@ -1115,12 +1192,12 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
.suspend_bias_off = true,
.component_driver = {
- .controls = aic31xx_snd_controls,
- .num_controls = ARRAY_SIZE(aic31xx_snd_controls),
- .dapm_widgets = aic31xx_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets),
- .dapm_routes = aic31xx_audio_map,
- .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map),
+ .controls = common31xx_snd_controls,
+ .num_controls = ARRAY_SIZE(common31xx_snd_controls),
+ .dapm_widgets = common31xx_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(common31xx_dapm_widgets),
+ .dapm_routes = common31xx_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(common31xx_audio_map),
},
};
@@ -1131,6 +1208,21 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = {
.digital_mute = aic31xx_dac_mute,
};
+static struct snd_soc_dai_driver dac31xx_dai_driver[] = {
+ {
+ .name = "tlv32dac31xx-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AIC31XX_RATES,
+ .formats = AIC31XX_FORMATS,
+ },
+ .ops = &aic31xx_dai_ops,
+ .symmetric_rates = 1,
+ }
+};
+
static struct snd_soc_dai_driver aic31xx_dai_driver[] = {
{
.name = "tlv320aic31xx-hifi",
@@ -1261,9 +1353,16 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
if (ret)
return ret;
- return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
- aic31xx_dai_driver,
- ARRAY_SIZE(aic31xx_dai_driver));
+ if (aic31xx->pdata.codec_type & DAC31XX_BIT)
+ return snd_soc_register_codec(&i2c->dev,
+ &soc_codec_driver_aic31xx,
+ dac31xx_dai_driver,
+ ARRAY_SIZE(dac31xx_dai_driver));
+ else
+ return snd_soc_register_codec(&i2c->dev,
+ &soc_codec_driver_aic31xx,
+ aic31xx_dai_driver,
+ ARRAY_SIZE(aic31xx_dai_driver));
}
static int aic31xx_i2c_remove(struct i2c_client *i2c)
@@ -1279,6 +1378,7 @@ static const struct i2c_device_id aic31xx_i2c_id[] = {
{ "tlv320aic3110", AIC3110 },
{ "tlv320aic3120", AIC3120 },
{ "tlv320aic3111", AIC3111 },
+ { "tlv320dac3100", DAC3100 },
{ }
};
MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index ac9b146526eb..5acd5b69fb83 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -24,12 +24,14 @@
#define AIC31XX_STEREO_CLASS_D_BIT 0x1
#define AIC31XX_MINIDSP_BIT 0x2
+#define DAC31XX_BIT 0x4
enum aic31xx_type {
AIC3100 = 0,
AIC3110 = AIC31XX_STEREO_CLASS_D_BIT,
AIC3120 = AIC31XX_MINIDSP_BIT,
AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT),
+ DAC3100 = DAC31XX_BIT,
};
struct aic31xx_pdata {
--
2.9.3