* [PATCH] ASoC: cs42l73: Add DAPM events for power down.
2012-12-07 20:53 [PATCH 1/3] ASoC: cs42l73: Add DMIC's as DAPM inputs Paul Handrigan
@ 2012-12-07 20:53 ` Paul Handrigan
2012-12-07 20:53 ` [PATCH 3/3] ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT Paul Handrigan
2012-12-10 3:27 ` [PATCH 1/3] ASoC: cs42l73: Add DMIC's as DAPM inputs Mark Brown
2 siblings, 0 replies; 4+ messages in thread
From: Paul Handrigan @ 2012-12-07 20:53 UTC (permalink / raw)
To: alsa-devel; +Cc: brian.austin, broonie, lrg, Paul Handrigan
Add power down delays between setting PDN and MCLKDIS for spk amp, spklo amp, and ear amp.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
---
sound/soc/codecs/cs42l73.c | 80 +++++++++++++++++++++++++++++++++++++++----
1 files changed, 72 insertions(+), 8 deletions(-)
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 2c08c4c..9447200 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -40,6 +40,7 @@ struct cs42l73_private {
u32 sysclk;
u8 mclksel;
u32 mclk;
+ int shutdwn_delay;
};
static const struct reg_default cs42l73_reg_defaults[] = {
@@ -588,6 +589,57 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
SOC_ENUM("XSPOUT Mono/Stereo Select", xsp_output_mux_enum),
};
+static int cs42l73_spklo_spk_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMD:
+ /* 150 ms delay between setting PDN and MCLKDIS */
+ priv->shutdwn_delay = 150;
+ break;
+ default:
+ pr_err("Invalid event = 0x%x\n", event);
+ }
+ return 0;
+}
+
+static int cs42l73_ear_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMD:
+ /* 50 ms delay between setting PDN and MCLKDIS */
+ if (priv->shutdwn_delay < 50)
+ priv->shutdwn_delay = 50;
+ break;
+ default:
+ pr_err("Invalid event = 0x%x\n", event);
+ }
+ return 0;
+}
+
+
+static int cs42l73_hp_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMD:
+ /* 30 ms delay between setting PDN and MCLKDIS */
+ if (priv->shutdwn_delay < 30)
+ priv->shutdwn_delay = 30;
+ break;
+ default:
+ pr_err("Invalid event = 0x%x\n", event);
+ }
+ return 0;
+}
+
static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("LINEINA"),
SND_SOC_DAPM_INPUT("LINEINB"),
@@ -674,16 +726,20 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1,
- &hp_amp_ctl),
+ SND_SOC_DAPM_SWITCH_E("HP Amp", CS42L73_PWRCTL3, 0, 1,
+ &hp_amp_ctl, cs42l73_hp_amp_event,
+ SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1,
&lo_amp_ctl),
- SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1,
- &spk_amp_ctl),
- SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1,
- &ear_amp_ctl),
- SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1,
- &spklo_amp_ctl),
+ SND_SOC_DAPM_SWITCH_E("SPK Amp", CS42L73_PWRCTL3, 2, 1,
+ &spk_amp_ctl, cs42l73_spklo_spk_amp_event,
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SWITCH_E("EAR Amp", CS42L73_PWRCTL3, 3, 1,
+ &ear_amp_ctl, cs42l73_ear_amp_event,
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SWITCH_E("SPKLO Amp", CS42L73_PWRCTL3, 4, 1,
+ &spklo_amp_ctl, cs42l73_spklo_spk_amp_event,
+ SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_OUTPUT("HPOUTA"),
SND_SOC_DAPM_OUTPUT("HPOUTB"),
@@ -1167,6 +1223,14 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_OFF:
snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1);
+ if (cs42l73->shutdwn_delay > 0) {
+ mdelay(cs42l73->shutdwn_delay);
+ cs42l73->shutdwn_delay = 0;
+ } else {
+ mdelay(15); /* Min amount of time requred to power
+ * down.
+ */
+ }
snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1);
break;
}
--
1.7.0.4
^ permalink raw reply related [flat|nested] 4+ messages in thread* [PATCH 3/3] ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
2012-12-07 20:53 [PATCH 1/3] ASoC: cs42l73: Add DMIC's as DAPM inputs Paul Handrigan
2012-12-07 20:53 ` [PATCH] ASoC: cs42l73: Add DAPM events for power down Paul Handrigan
@ 2012-12-07 20:53 ` Paul Handrigan
2012-12-10 3:27 ` [PATCH 1/3] ASoC: cs42l73: Add DMIC's as DAPM inputs Mark Brown
2 siblings, 0 replies; 4+ messages in thread
From: Paul Handrigan @ 2012-12-07 20:53 UTC (permalink / raw)
To: alsa-devel; +Cc: brian.austin, broonie, lrg, Paul Handrigan
Since VSP only has one power bit, only provide one DAPM widget.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
---
sound/soc/codecs/cs42l73.c | 32 +++++++++++++-------------------
1 files changed, 13 insertions(+), 19 deletions(-)
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 1a03c2b..fbeaef1 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -658,9 +658,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
CS42L73_PWRCTL2, 3, 1),
SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0,
CS42L73_PWRCTL2, 3, 1),
- SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0,
- CS42L73_PWRCTL2, 4, 1),
- SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0,
+ SND_SOC_DAPM_AIF_OUT("VSPINOUT", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -686,8 +684,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("VSP Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0,
CS42L73_PWRCTL2, 0, 1),
@@ -703,7 +700,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0,
CS42L73_PWRCTL2, 2, 1),
- SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0,
+ SND_SOC_DAPM_AIF_IN("VSPINOUT", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -763,7 +760,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"},
{"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"},
- {"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"},
+ {"ESL DAC", "ESL-VSP Mono Volume", "VSPINOUT"},
/* Loopback */
{"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"},
{"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"},
@@ -785,7 +782,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"},
{"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"},
- {"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"},
+ {"SPK DAC", "SPK-VSP Mono Volume", "VSPINOUT"},
/* Loopback */
{"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"},
{"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"},
@@ -828,8 +825,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"HL Right Mixer", NULL, "ASPINR"},
{"HL Left Mixer", NULL, "XSPINL"},
{"HL Right Mixer", NULL, "XSPINR"},
- {"HL Left Mixer", NULL, "VSPIN"},
- {"HL Right Mixer", NULL, "VSPIN"},
+ {"HL Left Mixer", NULL, "VSPINOUT"},
+ {"HL Right Mixer", NULL, "VSPINOUT"},
{"ASPINL", NULL, "ASP Playback"},
{"ASPINM", NULL, "ASP Playback"},
@@ -837,7 +834,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"XSPINL", NULL, "XSP Playback"},
{"XSPINM", NULL, "XSP Playback"},
{"XSPINR", NULL, "XSP Playback"},
- {"VSPIN", NULL, "VSP Playback"},
+ {"VSPINOUT", NULL, "VSP Playback"},
/* Capture Paths */
{"MIC1", NULL, "MIC1 Bias"},
@@ -879,21 +876,18 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"XSPOUTR", NULL, "XSPR Output Mixer"},
/* Voice Capture */
- {"VSPL Output Mixer", NULL, "Input Left Capture"},
- {"VSPR Output Mixer", NULL, "Input Left Capture"},
+ {"VSP Output Mixer", NULL, "Input Left Capture"},
+ {"VSP Output Mixer", NULL, "Input Right Capture"},
- {"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"},
- {"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"},
+ {"VSPINOUT", "VSP-IP Volume", "VSP Output Mixer"},
- {"VSPOUTL", NULL, "VSPL Output Mixer"},
- {"VSPOUTR", NULL, "VSPR Output Mixer"},
+ {"VSPINOUT", NULL, "VSP Output Mixer"},
{"ASP Capture", NULL, "ASPOUTL"},
{"ASP Capture", NULL, "ASPOUTR"},
{"XSP Capture", NULL, "XSPOUTL"},
{"XSP Capture", NULL, "XSPOUTR"},
- {"VSP Capture", NULL, "VSPOUTL"},
- {"VSP Capture", NULL, "VSPOUTR"},
+ {"VSP Capture", NULL, "VSPINOUT"},
};
struct cs42l73_mclk_div {
--
1.7.0.4
^ permalink raw reply related [flat|nested] 4+ messages in thread