Merge branch 'for-linus' into topic/hda-cleanup

Syncing the HD-audio updates for further cleanup works.

13 files changed, 67 insertions, 63 deletions

diff --git a/MAINTAINERS b/MAINTAINERS
index aefa94841ff3..8a17e7cf9446 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -3843,10 +3843,13 @@ F:	drivers/tty/serial/ucc_uart.c
 
 FREESCALE SOC SOUND DRIVERS
 M:	Timur Tabi <timur@tabi.org>
+M:	Nicolin Chen <nicoleotsuka@gmail.com>
+M:	Xiubo Li <Li.Xiubo@freescale.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 L:	linuxppc-dev@lists.ozlabs.org
 S:	Maintained
 F:	sound/soc/fsl/fsl*
+F:	sound/soc/fsl/imx*
 F:	sound/soc/fsl/mpc8610_hpcd.c
 
 FREEVXFS FILESYSTEM
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6b38ec3c6e57..d71270a3f73f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -181,6 +181,8 @@ static void alc_fix_pll(struct hda_codec *codec)
 			    spec->pll_coef_idx);
 	val = snd_hda_codec_read(codec, spec->pll_nid, 0,
 				 AC_VERB_GET_PROC_COEF, 0);
+	if (val == -1)
+		return;
 	snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
 			    spec->pll_coef_idx);
 	snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF,
@@ -2806,6 +2808,8 @@ static void alc286_shutup(struct hda_codec *codec)
 static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up)
 {
 	int val = alc_read_coef_idx(codec, 0x04);
+	if (val == -1)
+		return;
 	if (power_up)
 		val |= 1 << 11;
 	else
@@ -3264,6 +3268,15 @@ static int alc269_resume(struct hda_codec *codec)
 	snd_hda_codec_resume_cache(codec);
 	alc_inv_dmic_sync(codec, true);
 	hda_call_check_power_status(codec, 0x01);
+
+	/* on some machine, the BIOS will clear the codec gpio data when enter
+	 * suspend, and won't restore the data after resume, so we restore it
+	 * in the driver.
+	 */
+	if (spec->gpio_led)
+		snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA,
+			    spec->gpio_led);
+
 	if (spec->has_alc5505_dsp)
 		alc5505_dsp_resume(codec);
 
@@ -5311,27 +5324,30 @@ static void alc269_fill_coef(struct hda_codec *codec)
 	if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
 		val = alc_read_coef_idx(codec, 0x04);
 		/* Power up output pin */
-		alc_write_coef_idx(codec, 0x04, val | (1<<11));
+		if (val != -1)
+			alc_write_coef_idx(codec, 0x04, val | (1<<11));
 	}
 
 	if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		val = alc_read_coef_idx(codec, 0xd);
-		if ((val & 0x0c00) >> 10 != 0x1) {
+		if (val != -1 && (val & 0x0c00) >> 10 != 0x1) {
 			/* Capless ramp up clock control */
 			alc_write_coef_idx(codec, 0xd, val | (1<<10));
 		}
 		val = alc_read_coef_idx(codec, 0x17);
-		if ((val & 0x01c0) >> 6 != 0x4) {
+		if (val != -1 && (val & 0x01c0) >> 6 != 0x4) {
 			/* Class D power on reset */
 			alc_write_coef_idx(codec, 0x17, val | (1<<7));
 		}
 	}
 
 	val = alc_read_coef_idx(codec, 0xd); /* Class D */
-	alc_write_coef_idx(codec, 0xd, val | (1<<14));
+	if (val != -1)
+		alc_write_coef_idx(codec, 0xd, val | (1<<14));
 
 	val = alc_read_coef_idx(codec, 0x4); /* HP */
-	alc_write_coef_idx(codec, 0x4, val | (1<<11));
+	if (val != -1)
+		alc_write_coef_idx(codec, 0x4, val | (1<<11));
 }
 
 /*
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);