Merge tag 'sound-3.17-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A bunch of ASoC fixes with a few HD-audio fixes in this pull request.

  All fairly small, boring and device-specific fixes, in addition to
  MAINTAINERS update for better reviewing"

* tag 'sound-3.17-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/hdmi - apply Valleyview fix-ups to Cherryview display codec
  ALSA: hda/hdmi - set depop_delay for haswell plus
  ALSA: hda - restore the gpio led after resume
  ALSA: hda/realtek - Avoid setting wrong COEF on ALC269 & co
  ASoC: pxa-ssp: drop SNDRV_PCM_FMTBIT_S24_LE
  ASoC: fsl-esai: Revert .xlate_tdm_slot_mask() support
  ASoC: mcasp: Fix implicit BLCK divider setting
  ASoC: arizona: Fix TDM slot length handling in arizona_hw_params
  ASoC: pcm512x: Correct Digital Playback control names
  ASoC: dapm: Fix uninitialized variable in snd_soc_dapm_get_enum_double()
  ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset
  ASoC: Intel: Wait Baytrail ADSP boot at resume_early stage
  ASoC: Intel: Merge Baytrail ADSP suspend_noirq into suspend_late
  MAINTAINERS: Add i.MX maintainers and paths to Freescale ASoC entry
  ASoC: Intel: Update Baytrail ADSP firmware name

11 files changed, 43 insertions, 58 deletions

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index bd41ee4da078..2c71f16bd661 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	else
 		rates = &arizona_48k_bclk_rates[0];
 
+	wl = snd_pcm_format_width(params_format(params));
+
 	if (tdm_slots) {
 		arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n",
 				tdm_slots, tdm_width);
@@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 		channels = tdm_slots;
 	} else {
 		bclk_target = snd_soc_params_to_bclk(params);
+		tdm_width = wl;
 	}
 
 	if (chan_limit && chan_limit < channels) {
@@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
 			rates[bclk], rates[bclk] / lrclk);
 
-	wl = snd_pcm_format_width(params_format(params));
-	frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+	frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width;
 
 	reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame);
 
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 163ec3855fd4..0c8aefab404c 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds =
 			pcm512x_ramp_step_text);
 
 static const struct snd_kcontrol_new pcm512x_controls[] = {
-SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2,
 		 PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
 SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL,
 	       PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv),
 SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST,
 	       PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv),
-SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
 	   PCM512x_RQMR_SHIFT, 1, 1),
 
 SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index c28508da34cf..6a6b2ff7d7d7 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -403,7 +403,8 @@ out:
 	return ret;
 }
 
-static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+				      int div, bool explicit)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
@@ -420,7 +421,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
 			       ACLKXDIV(div - 1), ACLKXDIV_MASK);
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
 			       ACLKRDIV(div - 1), ACLKRDIV_MASK);
-		mcasp->bclk_div = div;
+		if (explicit)
+			mcasp->bclk_div = div;
 		break;
 
 	case 2:		/* BCLK/LRCLK ratio */
@@ -434,6 +436,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
 	return 0;
 }
 
+static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+				    int div)
+{
+	return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1);
+}
+
 static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 				    unsigned int freq, int dir)
 {
@@ -738,7 +746,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 				 "Inaccurate BCLK: %u Hz / %u != %u Hz\n",
 				 mcasp->sysclk_freq, div, bclk_freq);
 		}
-		davinci_mcasp_set_clkdiv(cpu_dai, 1, div);
+		__davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0);
 	}
 
 	ret = mcasp_common_hw_param(mcasp, substream->stream,
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f54a8fc99291..f3012b645b51 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI
 	tristate "Enhanced Serial Audio Interface (ESAI) module support"
 	select REGMAP_MMIO
 	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
-	select SND_SOC_FSL_UTILS
 	help
 	  Say Y if you want to add Enhanced Synchronous Audio Interface
 	  (ESAI) support for the Freescale CPUs.
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 72d154e7dd03..a3b29ed84963 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -18,7 +18,6 @@
 
 #include "fsl_esai.h"
 #include "imx-pcm.h"
-#include "fsl_utils.h"
 
 #define FSL_ESAI_RATES		SNDRV_PCM_RATE_8000_192000
 #define FSL_ESAI_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
@@ -607,7 +606,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = {
 	.hw_params = fsl_esai_hw_params,
 	.set_sysclk = fsl_esai_set_dai_sysclk,
 	.set_fmt = fsl_esai_set_dai_fmt,
-	.xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask,
 	.set_tdm_slot = fsl_esai_set_dai_tdm_slot,
 };
 
diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c
index 42edc6f4fc4a..03d0a166b635 100644
--- a/sound/soc/intel/sst-acpi.c
+++ b/sound/soc/intel/sst-acpi.c
@@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = {
 };
 
 static struct sst_acpi_mach baytrail_machines[] = {
-	{ "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" },
-	{ "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" },
+	{ "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
+	{ "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
 	{}
 };
 
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
index 67673a2c0f41..b4ad98c43e5c 100644
--- a/sound/soc/intel/sst-baytrail-ipc.c
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -817,7 +817,7 @@ static struct sst_dsp_device byt_dev = {
 	.ops = &sst_byt_ops,
 };
 
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
+int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
 {
 	struct sst_byt *byt = pdata->dsp;
 
@@ -826,14 +826,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
 	sst_byt_drop_all(byt);
 	dev_dbg(byt->dev, "dsp in reset\n");
 
-	return 0;
-}
-EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq);
-
-int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
-{
-	struct sst_byt *byt = pdata->dsp;
-
 	dev_dbg(byt->dev, "free all blocks and unload fw\n");
 	sst_fw_unload(byt->fw);
 
diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h
index 06a4d202689b..8faff6dcf25d 100644
--- a/sound/soc/intel/sst-baytrail-ipc.h
+++ b/sound/soc/intel/sst-baytrail-ipc.h
@@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt,
 int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata);
 void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata);
 struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt);
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata);
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 599401c0c655..eab1c7d85187 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -59,6 +59,9 @@ struct sst_byt_priv_data {
 
 	/* DAI data */
 	struct sst_byt_pcm_data pcm[BYT_PCM_COUNT];
+
+	/* flag indicating is stream context restore needed after suspend */
+	bool restore_stream;
 };
 
 /* this may get called several times by oss emulation */
@@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		sst_byt_stream_start(byt, pcm_data->stream, 0);
 		break;
 	case SNDRV_PCM_TRIGGER_RESUME:
-		schedule_work(&pcm_data->work);
+		if (pdata->restore_stream == true)
+			schedule_work(&pcm_data->work);
+		else
+			sst_byt_stream_resume(byt, pcm_data->stream);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		sst_byt_stream_resume(byt, pcm_data->stream);
@@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		sst_byt_stream_stop(byt, pcm_data->stream);
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
+		pdata->restore_stream = false;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		sst_byt_stream_pause(byt, pcm_data->stream);
 		break;
@@ -404,26 +411,10 @@ static const struct snd_soc_component_driver byt_dai_component = {
 };
 
 #ifdef CONFIG_PM
-static int sst_byt_pcm_dev_suspend_noirq(struct device *dev)
-{
-	struct sst_pdata *sst_pdata = dev_get_platdata(dev);
-	int ret;
-
-	dev_dbg(dev, "suspending noirq\n");
-
-	/* at this point all streams will be stopped and context saved */
-	ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata);
-	if (ret < 0) {
-		dev_err(dev, "failed to suspend %d\n", ret);
-		return ret;
-	}
-
-	return ret;
-}
-
 static int sst_byt_pcm_dev_suspend_late(struct device *dev)
 {
 	struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+	struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev);
 	int ret;
 
 	dev_dbg(dev, "suspending late\n");
@@ -434,34 +425,30 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev)
 		return ret;
 	}
 
+	priv_data->restore_stream = true;
+
 	return ret;
 }
 
 static int sst_byt_pcm_dev_resume_early(struct device *dev)
 {
 	struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+	int ret;
 
 	dev_dbg(dev, "resume early\n");
 
 	/* load fw and boot DSP */
-	return sst_byt_dsp_boot(dev, sst_pdata);
-}
-
-static int sst_byt_pcm_dev_resume(struct device *dev)
-{
-	struct sst_pdata *sst_pdata = dev_get_platdata(dev);
-
-	dev_dbg(dev, "resume\n");
+	ret = sst_byt_dsp_boot(dev, sst_pdata);
+	if (ret)
+		return ret;
 
 	/* wait for FW to finish booting */
 	return sst_byt_dsp_wait_for_ready(dev, sst_pdata);
 }
 
 static const struct dev_pm_ops sst_byt_pm_ops = {
-	.suspend_noirq = sst_byt_pcm_dev_suspend_noirq,
 	.suspend_late = sst_byt_pcm_dev_suspend_late,
 	.resume_early = sst_byt_pcm_dev_resume_early,
-	.resume = sst_byt_pcm_dev_resume,
 };
 
 #define SST_BYT_PM_OPS	(&sst_byt_pm_ops)
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 0109f6c2334e..a8e097433074 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai)
 			  SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 |	\
 			  SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
-#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
-			    SNDRV_PCM_FMTBIT_S24_LE |	\
-			    SNDRV_PCM_FMTBIT_S32_LE)
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
 	.startup	= pxa_ssp_startup,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8348352dc2c6..177bd8639ef9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int reg_val, val;
-	int ret = 0;
 
-	if (e->reg != SND_SOC_NOPM)
-		ret = soc_dapm_read(dapm, e->reg, &reg_val);
-	else
+	if (e->reg != SND_SOC_NOPM) {
+		int ret = soc_dapm_read(dapm, e->reg, &reg_val);
+		if (ret)
+			return ret;
+	} else {
 		reg_val = dapm_kcontrol_get_value(kcontrol);
+	}
 
 	val = (reg_val >> e->shift_l) & e->mask;
 	ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val);
@@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
 		ucontrol->value.enumerated.item[1] = val;
 	}
 
-	return ret;
+	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);