| lists.openwall.net | lists / announce owl-users owl-dev john-users john-dev passwdqc-users yescrypt popa3d-users / oss-security kernel-hardening musl sabotage tlsify passwords / crypt-dev xvendor / Bugtraq Full-Disclosure linux-kernel linux-netdev linux-ext4 linux-hardening linux-cve-announce PHC | |
|
Open Source and information security mailing list archives
| ||
|
Message-Id: <E1boZ0P-00079B-1G@finisterre>
Date: Mon, 26 Sep 2016 09:43:49 -0700
From: Mark Brown <broonie@...nel.org>
To: Nikita Yushchenko <nikita.yoush@...entembedded.com>
Cc: Mark Brown <broonie@...nel.org>,
Liam Girdwood <lgirdwood@...il.com>,
Mark Brown <broonie@...nel.org>,
Jaroslav Kysela <perex@...ex.cz>,
Takashi Iwai <tiwai@...e.com>,
Bastien Nocera <hadess@...ess.net>, Jyri Sarha <jsarha@...com>,
Peter Ujfalusi <peter.ujfalusi@...com>,
alsa-devel@...a-project.org, linux-kernel@...r.kernel.org,
Chris Healy <cphealy@...il.com>, alsa-devel@...a-project.org
Subject: Applied "ASoC: tlv320aic31xx: add explicit support for tlv320dac31xx" to the asoc tree
The patch
ASoC: tlv320aic31xx: add explicit support for tlv320dac31xx
has been applied to the asoc tree at
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From ef9656b6936fb7f66e7e25d284c955f4893ac421 Mon Sep 17 00:00:00 2001
From: Nikita Yushchenko <nikita.yoush@...entembedded.com>
Date: Fri, 23 Sep 2016 14:52:52 +0300
Subject: [PATCH] ASoC: tlv320aic31xx: add explicit support for tlv320dac31xx
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@...entembedded.com>
Signed-off-by: Mark Brown <broonie@...nel.org>
---
.../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
Powered by blists - more mailing lists