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-rw-r--r--sound/soc/codecs/ak4642.c31
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/imx/imx-ssi.c2
-rw-r--r--sound/soc/omap/ams-delta.c34
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c4
-rw-r--r--sound/soc/soc-dapm.c12
6 files changed, 30 insertions, 55 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 16bd1e7d238..f8e10ced244 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
-
- SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
};
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
- SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+ SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
- SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+ &ak4642_headphone_control),
- SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
&ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
static const struct snd_soc_dapm_route ak4642_intercon[] = {
/* Outputs */
- {"HPOUTL", NULL, "HPOUTL Mixer"},
- {"HPOUTR", NULL, "HPOUTR Mixer"},
+ {"HPOUTL", NULL, "HPL Out"},
+ {"HPOUTR", NULL, "HPR Out"},
{"LINEOUT", NULL, "LINEOUT Mixer"},
- {"HPOUTL Mixer", "DACH", "DAC"},
- {"HPOUTR Mixer", "DACH", "DAC"},
+ {"HPL Out", NULL, "Headphone Enable"},
+ {"HPR Out", NULL, "Headphone Enable"},
+
+ {"Headphone Enable", "Switch", "DACH"},
+
+ {"DACH", NULL, "DAC"},
+
{"LINEOUT Mixer", "DACL", "DAC"},
};
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 5bcb350bacc..15d467ff91b 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1988,7 +1988,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
static const struct soc_enum str_enum =
SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 9203cdd0a15..4f81ed45632 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+ strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
break;
}
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 78563bbbbf0..41586b26ce9 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -426,29 +426,6 @@ static struct snd_soc_ops ams_delta_ops = {
};
-/* Board specific codec bias level control */
-static int ams_delta_set_bias_level(struct snd_soc_card *card,
- struct snd_soc_dapm_context *dapm,
- enum snd_soc_bias_level level)
-{
- switch (level) {
- case SND_SOC_BIAS_ON:
- case SND_SOC_BIAS_PREPARE:
- case SND_SOC_BIAS_STANDBY:
- if (card->dapm.bias_level == SND_SOC_BIAS_OFF)
- ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
- AMS_DELTA_LATCH2_MODEM_NRESET);
- break;
- case SND_SOC_BIAS_OFF:
- if (card->dapm.bias_level != SND_SOC_BIAS_OFF)
- ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
- 0);
- }
- card->dapm.bias_level = level;
-
- return 0;
-}
-
/* Digital mute implemented using modem/CPU multiplexer.
* Shares hardware with codec config pulse generation */
static bool ams_delta_muted = 1;
@@ -512,9 +489,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
ams_delta_ops.shutdown = ams_delta_shutdown;
}
- /* Set codec bias level */
- ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
-
/* Add hook switch - can be used to control the codec from userspace
* even if line discipline fails */
ret = snd_soc_jack_new(rtd->codec, "hook_switch",
@@ -598,7 +572,6 @@ static struct snd_soc_card ams_delta_audio_card = {
.owner = THIS_MODULE,
.dai_link = &ams_delta_dai_link,
.num_links = 1,
- .set_bias_level = ams_delta_set_bias_level,
};
/* Module init/exit */
@@ -635,7 +608,7 @@ err:
platform_device_put(ams_delta_audio_platform_device);
return ret;
}
-module_init(ams_delta_module_init);
+late_initcall(ams_delta_module_init);
static void __exit ams_delta_module_exit(void)
{
@@ -647,11 +620,6 @@ static void __exit ams_delta_module_exit(void)
ARRAY_SIZE(ams_delta_hook_switch_gpios),
ams_delta_hook_switch_gpios);
- /* Keep modem power on */
- ams_delta_set_bias_level(&ams_delta_audio_card,
- &ams_delta_audio_card.rtd[0].codec->dapm,
- SND_SOC_BIAS_STANDBY);
-
platform_device_unregister(cx20442_platform_device);
platform_device_unregister(ams_delta_audio_platform_device);
}
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 24bdb321269..321d51134e4 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -367,7 +367,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -376,7 +376,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.stream_name = "Voice",
.cpu_dai_name = "dfbmcs320-pcm",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-001a",
+ .codec_name = "wm8753.0-001a",
.ops = &neo1973_voice_ops,
},
};
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index dcd11609f93..6241490fff3 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3238,9 +3238,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
* standby.
*/
if (powerdown) {
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
+ if (dapm->bias_level == SND_SOC_BIAS_ON)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_PREPARE);
dapm_seq_run(dapm, &down_list, 0, false);
- snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_STANDBY);
}
}
@@ -3253,7 +3257,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
list_for_each_entry(codec, &card->codec_dev_list, list) {
soc_dapm_shutdown_codec(&codec->dapm);
- snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(&codec->dapm,
+ SND_SOC_BIAS_OFF);
}
}
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2025-01-08 14:50:31 +0000