diff options
author | Takashi Iwai <tiwai@suse.de> | 2012-12-17 15:40:55 +0100 |
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committer | Takashi Iwai <tiwai@suse.de> | 2012-12-17 15:40:55 +0100 |
commit | 6be7f5344b4fca35f1955aa73f0de825316a3236 (patch) | |
tree | 25ccce9c837fbc7beef026b46852197b637be901 /sound | |
parent | ALSA: usb-audio: ignore-quirk for HP Wireless Audio (diff) | |
parent | Merge remote-tracking branch 'asoc/topic/tpa6130a2' into asoc-next (diff) | |
download | linux-6be7f5344b4fca35f1955aa73f0de825316a3236.tar.xz linux-6be7f5344b4fca35f1955aa73f0de825316a3236.zip |
Merge tag 'asoc-3.8p1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.8
Nothing terribly exciting here, just small localised changes.
As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/codecs/cs42l73.c | 116 | ||||
-rw-r--r-- | sound/soc/codecs/sigmadsp.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/tpa6130a2.c | 23 | ||||
-rw-r--r-- | sound/soc/soc-compress.c | 2 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 10 | ||||
-rw-r--r-- | sound/soc/soc-pcm.c | 12 |
6 files changed, 108 insertions, 57 deletions
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index a0791ecf6d95..6361dab48bd1 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[] = { @@ -588,7 +589,60 @@ 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"), SND_SOC_DAPM_INPUT("LINEINA"), SND_SOC_DAPM_INPUT("LINEINB"), SND_SOC_DAPM_INPUT("MIC1"), @@ -604,9 +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), @@ -632,8 +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), @@ -649,7 +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), @@ -674,16 +725,20 @@ 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"), @@ -705,7 +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"}, @@ -727,7 +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"}, @@ -770,8 +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"}, @@ -779,7 +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"}, @@ -795,6 +850,8 @@ 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"}, @@ -819,21 +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 { @@ -1167,6 +1221,14 @@ 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; } diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 5be42bf56996..4068f2491232 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); } diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8d75aa152c8c..c58bee8346ce 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -398,7 +398,8 @@ static int 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 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 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 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; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 967d0e173e1b..5fbfb06e8083 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-core.c b/sound/soc/soc-core.c index 9c768bcb98a6..91d592ff67b7 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; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5c3ca2a34661..d7711fce119b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -334,11 +334,11 @@ static void close_delayed_work(struct work_struct *work) dev_dbg(rtd->dev, "ASoC: 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); } @@ -408,7 +408,7 @@ 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)); } @@ -480,8 +480,8 @@ 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); } |