Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fsl',...

Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fsl', 'asoc/fix/fsl-esai', 'asoc/fix/intel', 'asoc/fix/mcasp' and 'asoc/fix/pxa' into asoc-linus

MAINTAINERS

@@ -3794,10 +3794,13 @@ 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

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);
 

sound/soc/davinci/davinci-mcasp.c

@@ -403,7 +403,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	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,

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.

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,
 };
 

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" },
 	{}
 };
 

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);
 

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);

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)

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,