From d06080cf08e6b59971959d9be3d0587c6e033292 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 7 Dec 2012 16:32:26 +0530 Subject: ASoC: tpa6130a2: Use devm_* APIs Converted to use devm_gpio_request and devm_regulator_get APIs. These are device managed and make error handling and cleanup a bit simpler. Cc: Peter Ujfalusi Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 23 ++++++----------------- 1 file changed, 6 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 565ff39..ec78073 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -398,7 +398,8 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, TPA6130A2_MUTE_L; if (data->power_gpio >= 0) { - ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + ret = devm_gpio_request(dev, data->power_gpio, + "tpa6130a2 enable"); if (ret < 0) { dev_err(dev, "Failed to request power GPIO (%d)\n", data->power_gpio); @@ -419,16 +420,16 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, break; } - data->supply = regulator_get(dev, regulator); + data->supply = devm_regulator_get(dev, regulator); if (IS_ERR(data->supply)) { ret = PTR_ERR(data->supply); dev_err(dev, "Failed to request supply: %d\n", ret); - goto err_regulator; + goto err_gpio; } ret = tpa6130a2_power(1); if (ret != 0) - goto err_power; + goto err_gpio; /* Read version */ @@ -440,15 +441,10 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, /* Disable the chip */ ret = tpa6130a2_power(0); if (ret != 0) - goto err_power; + goto err_gpio; return 0; -err_power: - regulator_put(data->supply); -err_regulator: - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); err_gpio: tpa6130a2_client = NULL; @@ -457,14 +453,7 @@ err_gpio: static int __devexit tpa6130a2_remove(struct i2c_client *client) { - struct tpa6130a2_data *data = i2c_get_clientdata(client); - tpa6130a2_power(0); - - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); - - regulator_put(data->supply); tpa6130a2_client = NULL; return 0; -- cgit v1.1 From a3adb1432d7a3ad86bb17a1638e44414537e4118 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 7 Dec 2012 18:30:51 +0100 Subject: ASoC: sigmadsp: Fix endianness conversion issue The 'addr' field of the sigma_action struct is stored as big endian in the firmware file. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sigmadsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 5be42bf..4068f24 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -225,7 +225,7 @@ EXPORT_SYMBOL(process_sigma_firmware); static int sigma_action_write_regmap(void *control_data, const struct sigma_action *sa, size_t len) { - return regmap_raw_write(control_data, le16_to_cpu(sa->addr), + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), sa->payload, len - 2); } -- cgit v1.1 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 --- sound/soc/codecs/cs42l73.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') 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 v1.1 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 --- sound/soc/codecs/cs42l73.c | 80 +++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 72 insertions(+), 8 deletions(-) (limited to 'sound') 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 v1.1 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 --- sound/soc/codecs/cs42l73.c | 32 +++++++++++++------------------- 1 file changed, 13 insertions(+), 19 deletions(-) (limited to 'sound') 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 v1.1 From c871bd0b2e627ff387d0ff055d8175879c80d01f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Dec 2012 16:19:52 +0900 Subject: ASoC: core: Fix splitting of log messages Don't wrap log messages over multiple lines, it makes them hard to grep for. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cee37ee..0d42afb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4155,9 +4155,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, ret = of_property_read_string_index(np, propname, 2 * i, &routes[i].sink); if (ret) { - dev_err(card->dev, "ASoC: Property '%s' index %d" - " could not be read: %d\n", propname, 2 * i, - ret); + dev_err(card->dev, + "ASoC: Property '%s' index %d could not be read: %d\n", + propname, 2 * i, ret); kfree(routes); return -EINVAL; } @@ -4165,8 +4165,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, (2 * i) + 1, &routes[i].source); if (ret) { dev_err(card->dev, - "ASoC: Property '%s' index %d could not be" - " read: %d\n", propname, (2 * i) + 1, ret); + "ASoC: Property '%s' index %d could not be read: %d\n", + propname, (2 * i) + 1, ret); kfree(routes); return -EINVAL; } -- cgit v1.1 From 9bffb1fb7c22c96d51d4ba06e2e023dd568a5872 Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Thu, 13 Dec 2012 12:23:05 -0600 Subject: ASoC: Prevent pop_wait overwrite pop_wait is used to determine if a deferred playback close needs to be cancelled when the a PCM is open or if after the power-down delay expires it needs to run. pop_wait is associated with the CODEC DAI, so the CODEC DAI must be unique. This holds true for most CODECs, except for the dummy CODEC and its DAI. In DAI links with non-unique dummy CODECs (e.g. front-ends), pop_wait can be overwritten by another DAI link using also a dummy CODEC. Failure to cancel a deferred close can cause mute due to the DAPM STOP event sent in the deferred work. One scenario where pop_wait is overwritten and causing mute is below (where hw:0,0 and hw:0,1 are two front-ends with default pmdown_time = 5 secs): aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1 sleep 1 aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 & aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE Since CODECs may not be unique, pop_wait is moved to the PCM runtime structure. Creating separate dummy CODECs for each DAI link can also solve the problem, but at this point it's only pop_wait variable in the CODEC DAI that has negative effects by not being unique. Signed-off-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 12 ++++++------ 2 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 967d0e1..5fbfb06 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -113,7 +113,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } else - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } else { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..3a2423b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -333,11 +333,11 @@ static void close_delayed_work(struct work_struct *work) pr_debug("pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); + rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; + if (rtd->pop_wait == 1) { + rtd->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } @@ -407,7 +407,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } @@ -478,8 +478,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; + rtd->pop_wait) { + rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); } -- cgit v1.1