From a1ad500e369183796820bffb4012b876a8685219 Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Fri, 7 Dec 2012 14:53:42 -0600 Subject: ASoC: cs42l73: Add DMIC's as DAPM inputs. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 2c08c4c..4766795 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -589,6 +589,8 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { }; static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), SND_SOC_DAPM_INPUT("LINEINA"), SND_SOC_DAPM_INPUT("LINEINB"), SND_SOC_DAPM_INPUT("MIC1"), @@ -795,6 +797,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"ADC Left", NULL, "PGA Left"}, {"ADC Right", NULL, "PGA Right"}, + {"DMIC Left", NULL, "DMICA"}, + {"DMIC Right", NULL, "DMICB"}, {"Input Left Capture", "ADC Left Input", "ADC Left"}, {"Input Right Capture", "ADC Right Input", "ADC Right"}, -- cgit v0.10.2 From 41df0829cee9e4c4ba68de33b4ca26cb18ac8ed7 Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Fri, 7 Dec 2012 14:53:43 -0600 Subject: ASoC: cs42l73: Add DAPM events for power down. Add power down delays between setting PDN and MCLKDIS for spk amp, spklo amp, and ear amp. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 4766795..fb9b178 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("DMICA"), SND_SOC_DAPM_INPUT("DMICB"), @@ -676,16 +728,20 @@ 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"), @@ -1171,6 +1227,14 @@ 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; } -- cgit v0.10.2 From 7f3dd4a8e31cdaed5f80f24b798cedcab644830b Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Fri, 7 Dec 2012 14:53:44 -0600 Subject: ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT Since VSP only has one power bit, only provide one DAPM widget. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index fb9b178..dd0d9b2 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 { -- cgit v0.10.2