Merge tag 'asoc-v3.15' into asoc-intel

ASoC: Updates for v3.15

Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:

 - Lots of cleanups from Takashi for enumerations; the original API for
   these was error prone so he's refactored lots of code to use more
   modern APIs which avoid issues.
 - Elimination of the ASoC level wrappers for I2C and SPI moving us
   closer to converting to regmap completely and avoiding some
   randconfig hassle.
 - Provide both manually and transparently locked DAPM APIs rather than
   a mix of the two fixing some concurrency issues.
 - Start converting CODEC drivers to use separate bus interface drivers
   rather than having them all in one file helping avoid dependency
   issues.
 - DPCM support for Intel Haswell and Bay Trail platforms.
 - Lots of work on improvements for simple-card, DaVinci and the Renesas
   rcar drivers.
 - New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
   CSR SiRF SoC.

Conflicts:
	sound/soc/intel/Kconfig

6 files changed, 65 insertions, 127 deletions

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 454f41cfc82..f2e289180e4 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -59,7 +59,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750
 	select SND_SOC_WM8750
 	select SND_S3C2412_SOC_I2S
 	help
-	  Sat Y if you want to add support for SoC audio on the Jive.
+	  Say Y if you want to add support for SoC audio on the Jive.
 
 config SND_SOC_SAMSUNG_SMDK_WM8580
 	tristate "SoC I2S Audio support for WM8580 on SMDK"
@@ -117,7 +117,7 @@ config SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23
 	tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards"
 	depends on SND_SOC_SAMSUNG && ARCH_S3C24XX
 	select SND_S3C24XX_I2S
-	select SND_SOC_TLV320AIC23
+	select SND_SOC_TLV320AIC23_I2C
 	select SND_SOC_SAMSUNG_SIMTEC
 
 config SND_SOC_SAMSUNG_SIMTEC_HERMES
@@ -145,11 +145,11 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380
 
 config SND_SOC_SAMSUNG_SMDK_WM9713
 	tristate "SoC AC97 Audio support for SMDK with WM9713"
-	depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210)
+	depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110)
 	select SND_SOC_WM9713
 	select SND_SAMSUNG_AC97
 	help
-	  Sat Y if you want to add support for SoC audio on the SMDK.
+	  Say Y if you want to add support for SoC audio on the SMDK.
 
 config SND_SOC_SMARTQ
 	tristate "SoC I2S Audio support for SmartQ board"
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index fbced589d07..88b09e02250 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -66,10 +66,6 @@ static int h1940_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
-	runtime->hw.rate_min = hw_rates.list[0];
-	runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
-	runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
-
 	return snd_pcm_hw_constraint_list(runtime, 0,
 					SNDRV_PCM_HW_PARAM_RATE,
 					&hw_rates);
@@ -94,7 +90,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream,
 			div++;
 		break;
 	default:
-		dev_err(&rtd->dev, "%s: rate %d is not supported\n",
+		dev_err(rtd->dev, "%s: rate %d is not supported\n",
 			__func__, rate);
 		return -EINVAL;
 	}
@@ -181,7 +177,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
 
 	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
 	snd_soc_dapm_enable_pin(dapm, "Speaker");
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 98a04c11202..b0800337b79 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -192,44 +192,6 @@ static struct snd_soc_ops neo1973_voice_ops = {
 	.hw_free = neo1973_voice_hw_free,
 };
 
-/* Shared routes and controls */
-
-static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
-	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
-	SND_SOC_DAPM_LINE("GSM Line In", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_MIC("Handset Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
-	/* Connections to the GSM Module */
-	{"GSM Line Out", NULL, "MONO1"},
-	{"GSM Line Out", NULL, "MONO2"},
-	{"RXP", NULL, "GSM Line In"},
-	{"RXN", NULL, "GSM Line In"},
-
-	/* Connections to Headset */
-	{"MIC1", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Headset Mic"},
-
-	/* Call Mic */
-	{"MIC2", NULL, "Mic Bias"},
-	{"MIC2N", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Handset Mic"},
-
-	/* Connect the ALC pins */
-	{"ACIN", NULL, "ACOP"},
-};
-
-static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
-	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
-	SOC_DAPM_PIN_SWITCH("GSM Line In"),
-	SOC_DAPM_PIN_SWITCH("Headset Mic"),
-	SOC_DAPM_PIN_SWITCH("Handset Mic"),
-};
-
-/* GTA02 specific routes and controls */
-
 static int gta02_speaker_enabled;
 
 static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
@@ -257,7 +219,34 @@ static int lm4853_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
+static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+	SND_SOC_DAPM_LINE("GSM Line In", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Handset Mic", NULL),
+	SND_SOC_DAPM_SPK("Handset Spk", NULL),
+	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
+};
+
+static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
+	/* Connections to the GSM Module */
+	{"GSM Line Out", NULL, "MONO1"},
+	{"GSM Line Out", NULL, "MONO2"},
+	{"RXP", NULL, "GSM Line In"},
+	{"RXN", NULL, "GSM Line In"},
+
+	/* Connections to Headset */
+	{"MIC1", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Headset Mic"},
+
+	/* Call Mic */
+	{"MIC2", NULL, "Mic Bias"},
+	{"MIC2N", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Handset Mic"},
+
+	/* Connect the ALC pins */
+	{"ACIN", NULL, "ACOP"},
+
 	/* Connections to the amp */
 	{"Stereo Out", NULL, "LOUT1"},
 	{"Stereo Out", NULL, "ROUT1"},
@@ -267,7 +256,11 @@ static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
 	{"Handset Spk", NULL, "ROUT2"},
 };
 
-static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
+static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
+	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+	SOC_DAPM_PIN_SWITCH("GSM Line In"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Handset Mic"),
 	SOC_DAPM_PIN_SWITCH("Handset Spk"),
 	SOC_DAPM_PIN_SWITCH("Stereo Out"),
 
@@ -276,86 +269,32 @@ static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
 		lm4853_set_spk),
 };
 
-static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = {
-	SND_SOC_DAPM_SPK("Handset Spk", NULL),
-	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
-};
-
-static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret;
-
-	ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets,
-			ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets));
-	if (ret)
-		return ret;
-
-	ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes,
-			ARRAY_SIZE(neo1973_gta02_routes));
-	if (ret)
-		return ret;
-
-	ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls,
-			ARRAY_SIZE(neo1973_gta02_wm8753_controls));
-	if (ret)
-		return ret;
-
-	snd_soc_dapm_disable_pin(dapm, "Stereo Out");
-	snd_soc_dapm_disable_pin(dapm, "Handset Spk");
-	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
-	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
-
-	return 0;
-}
-
 static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret;
+	struct snd_soc_card *card = rtd->card;
 
 	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(dapm, "OUT3");
-	snd_soc_dapm_nc_pin(dapm, "OUT4");
-	snd_soc_dapm_nc_pin(dapm, "LINE1");
-	snd_soc_dapm_nc_pin(dapm, "LINE2");
-
-	/* Add neo1973 specific widgets */
-	ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets,
-			ARRAY_SIZE(neo1973_wm8753_dapm_widgets));
-	if (ret)
-		return ret;
-
-	/* add neo1973 specific controls */
-	ret = snd_soc_add_card_controls(rtd->card, neo1973_wm8753_controls,
-			ARRAY_SIZE(neo1973_wm8753_controls));
-	if (ret)
-		return ret;
-
-	/* set up neo1973 specific audio routes */
-	ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes,
-			ARRAY_SIZE(neo1973_wm8753_routes));
-	if (ret)
-		return ret;
+	snd_soc_dapm_nc_pin(&codec->dapm, "OUT3");
+	snd_soc_dapm_nc_pin(&codec->dapm, "OUT4");
+	snd_soc_dapm_nc_pin(&codec->dapm, "LINE1");
+	snd_soc_dapm_nc_pin(&codec->dapm, "LINE2");
 
 	/* set endpoints to default off mode */
-	snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-	snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-	snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+	snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out");
+	snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In");
+	snd_soc_dapm_disable_pin(&card->dapm, "Headset Mic");
+	snd_soc_dapm_disable_pin(&card->dapm, "Handset Mic");
+	snd_soc_dapm_disable_pin(&card->dapm, "Stereo Out");
+	snd_soc_dapm_disable_pin(&card->dapm, "Handset Spk");
 
 	/* allow audio paths from the GSM modem to run during suspend */
-	snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
-	snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
-	snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
-	snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
-
-	if (machine_is_neo1973_gta02()) {
-		ret = neo1973_gta02_wm8753_init(codec);
-		if (ret)
-			return ret;
-	}
+	snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line Out");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line In");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Mic");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Stereo Out");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Spk");
 
 	return 0;
 }
@@ -409,6 +348,13 @@ static struct snd_soc_card neo1973 = {
 	.num_aux_devs = ARRAY_SIZE(neo1973_aux_devs),
 	.codec_conf = neo1973_codec_conf,
 	.num_configs = ARRAY_SIZE(neo1973_codec_conf),
+
+	.controls = neo1973_wm8753_controls,
+	.num_controls = ARRAY_SIZE(neo1973_wm8753_controls),
+	.dapm_widgets = neo1973_wm8753_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets),
+	.dapm_routes = neo1973_wm8753_routes,
+	.num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes),
 };
 
 static struct platform_device *neo1973_snd_device;
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 06ebdc06177..2982d9e7f26 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -131,10 +131,6 @@ static int rx1950_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
-	runtime->hw.rate_min = hw_rates.list[0];
-	runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
-	runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
-
 	return snd_pcm_hw_constraint_list(runtime, 0,
 					SNDRV_PCM_HW_PARAM_RATE,
 					&hw_rates);
@@ -226,7 +222,6 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
 
 	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
 	snd_soc_dapm_enable_pin(dapm, "Speaker");
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index d38ae98e2f3..682eb4f7ba0 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -202,7 +202,7 @@ static int smdk_audio_probe(struct platform_device *pdev)
 
 static struct platform_driver smdk_audio_driver = {
 	.driver		= {
-		.name	= "smdk-audio-wm8894",
+		.name	= "smdk-audio-wm8994",
 		.owner	= THIS_MODULE,
 		.of_match_table = of_match_ptr(samsung_wm8994_of_match),
 		.pm	= &snd_soc_pm_ops,
diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c
index f21ff608a81..1807b75ccc1 100644
--- a/sound/soc/samsung/tobermory.c
+++ b/sound/soc/samsung/tobermory.c
@@ -44,6 +44,8 @@ static int tobermory_set_bias_level(struct snd_soc_card *card,
 						     SND_SOC_CLOCK_IN);
 			if (ret < 0) {
 				pr_err("Failed to set SYSCLK: %d\n", ret);
+				snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+						    0, 0, 0);
 				return ret;
 			}
 		}