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.

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);
         }