From 324752632a2017cc2e2464d110445328ad2a987c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 19 Nov 2013 01:06:15 -0800
Subject: ASoC: rcar: rename GEN2_SRU to GEN2_SCU

Gen2 has SCU. SRU is for Gen1

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/rcar_snd.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h
index 12afab18945d..a818ff76b138 100644
--- a/include/sound/rcar_snd.h
+++ b/include/sound/rcar_snd.h
@@ -18,7 +18,7 @@
 #define RSND_GEN1_ADG	1
 #define RSND_GEN1_SSI	2
 
-#define RSND_GEN2_SRU	0
+#define RSND_GEN2_SCU	0
 #define RSND_GEN2_ADG	1
 #define RSND_GEN2_SSIU	2
 #define RSND_GEN2_SSI	3
-- 
cgit v1.2.3


From 3635bf09a89cf92b80ac44198c5c8f0989624ea6 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <b42378@freescale.com>
Date: Wed, 13 Nov 2013 18:56:24 +0800
Subject: ASoC: soc-pcm: add symmetry for channels and sample bits

Some SoCs can only work in mono or stereo mode at one time. So if
we let them capture a mono stream while playing a stereo stream,
there might be a problem occur to one of these two streams: double
paced or slowed down.

In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate
symmetry. But we don't have one for channels.

Likewise, we can treat symmetric_rate as a solution for those SoCs
or CODECs which can not handle asymmetrical LRCLK. But it's also
impossible for them to handle asymmetrical BCLK. And accodring to
BCLK = LRCLK * channel number * slot size(fixed or sample bits),
sample bits might also be a problem if they are not using a fixed
slot size.

Thus, this patch applys symmetry for channels and sample bits.

Meanwhile, there might be a race between two substreams if starting
simultaneously. Previously, we only added warning to compalin but
still using conservative way to let it carry on. However, this patch
rejects the second stream with any unmatched parameter to make sure
the first existing stream won't be broken.

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/soc-dai.h |   6 +++
 include/sound/soc.h     |   2 +
 sound/soc/soc-pcm.c     | 130 +++++++++++++++++++++++++++++++++++++++---------
 3 files changed, 115 insertions(+), 23 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 800c101bb096..243d3b689699 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -220,6 +220,8 @@ struct snd_soc_dai_driver {
 	struct snd_soc_pcm_stream capture;
 	struct snd_soc_pcm_stream playback;
 	unsigned int symmetric_rates:1;
+	unsigned int symmetric_channels:1;
+	unsigned int symmetric_samplebits:1;
 
 	/* probe ordering - for components with runtime dependencies */
 	int probe_order;
@@ -244,6 +246,8 @@ struct snd_soc_dai {
 	unsigned int capture_active:1;		/* stream is in use */
 	unsigned int playback_active:1;		/* stream is in use */
 	unsigned int symmetric_rates:1;
+	unsigned int symmetric_channels:1;
+	unsigned int symmetric_samplebits:1;
 	struct snd_pcm_runtime *runtime;
 	unsigned int active;
 	unsigned char probed:1;
@@ -258,6 +262,8 @@ struct snd_soc_dai {
 
 	/* Symmetry data - only valid if symmetry is being enforced */
 	unsigned int rate;
+	unsigned int channels;
+	unsigned int sample_bits;
 
 	/* parent platform/codec */
 	struct snd_soc_platform *platform;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 1f741cb24f33..1cda7d343d16 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -879,6 +879,8 @@ struct snd_soc_dai_link {
 
 	/* Symmetry requirements */
 	unsigned int symmetric_rates:1;
+	unsigned int symmetric_channels:1;
+	unsigned int symmetric_samplebits:1;
 
 	/* Do not create a PCM for this DAI link (Backend link) */
 	unsigned int no_pcm:1;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..ed1e077114a2 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -84,30 +84,97 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	int ret;
 
-	if (!soc_dai->driver->symmetric_rates &&
-	    !rtd->dai_link->symmetric_rates)
-		return 0;
+	if (soc_dai->rate && (soc_dai->driver->symmetric_rates ||
+				rtd->dai_link->symmetric_rates)) {
+		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n",
+				soc_dai->rate);
+
+		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_RATE,
+						soc_dai->rate, soc_dai->rate);
+		if (ret < 0) {
+			dev_err(soc_dai->dev,
+				"ASoC: Unable to apply rate constraint: %d\n",
+				ret);
+			return ret;
+		}
+	}
 
-	/* This can happen if multiple streams are starting simultaneously -
-	 * the second can need to get its constraints before the first has
-	 * picked a rate.  Complain and allow the application to carry on.
-	 */
-	if (!soc_dai->rate) {
-		dev_warn(soc_dai->dev,
-			 "ASoC: Not enforcing symmetric_rates due to race\n");
-		return 0;
+	if (soc_dai->channels && (soc_dai->driver->symmetric_channels ||
+				rtd->dai_link->symmetric_channels)) {
+		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d channel(s)\n",
+				soc_dai->channels);
+
+		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_CHANNELS,
+						soc_dai->channels,
+						soc_dai->channels);
+		if (ret < 0) {
+			dev_err(soc_dai->dev,
+				"ASoC: Unable to apply channel symmetry constraint: %d\n",
+				ret);
+			return ret;
+		}
 	}
 
-	dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n", soc_dai->rate);
+	if (soc_dai->sample_bits && (soc_dai->driver->symmetric_samplebits ||
+				rtd->dai_link->symmetric_samplebits)) {
+		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d sample bits\n",
+				soc_dai->sample_bits);
 
-	ret = snd_pcm_hw_constraint_minmax(substream->runtime,
-					   SNDRV_PCM_HW_PARAM_RATE,
-					   soc_dai->rate, soc_dai->rate);
-	if (ret < 0) {
-		dev_err(soc_dai->dev,
-			"ASoC: Unable to apply rate symmetry constraint: %d\n",
-			ret);
-		return ret;
+		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+						soc_dai->sample_bits,
+						soc_dai->sample_bits);
+		if (ret < 0) {
+			dev_err(soc_dai->dev,
+				"ASoC: Unable to apply sample bits symmetry constraint: %d\n",
+				ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	unsigned int rate, channels, sample_bits, symmetry;
+
+	rate = params_rate(params);
+	channels = params_channels(params);
+	sample_bits = snd_pcm_format_physical_width(params_format(params));
+
+	/* reject unmatched parameters when applying symmetry */
+	symmetry = cpu_dai->driver->symmetric_rates ||
+		codec_dai->driver->symmetric_rates ||
+		rtd->dai_link->symmetric_rates;
+	if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
+		dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
+				cpu_dai->rate, rate);
+		return -EINVAL;
+	}
+
+	symmetry = cpu_dai->driver->symmetric_channels ||
+		codec_dai->driver->symmetric_channels ||
+		rtd->dai_link->symmetric_channels;
+	if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
+		dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
+				cpu_dai->channels, channels);
+		return -EINVAL;
+	}
+
+	symmetry = cpu_dai->driver->symmetric_samplebits ||
+		codec_dai->driver->symmetric_samplebits ||
+		rtd->dai_link->symmetric_samplebits;
+	if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) {
+		dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
+				cpu_dai->sample_bits, sample_bits);
+		return -EINVAL;
 	}
 
 	return 0;
@@ -384,11 +451,17 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	codec->active--;
 
 	/* clear the corresponding DAIs rate when inactive */
-	if (!cpu_dai->active)
+	if (!cpu_dai->active) {
 		cpu_dai->rate = 0;
+		cpu_dai->channels = 0;
+		cpu_dai->sample_bits = 0;
+	}
 
-	if (!codec_dai->active)
+	if (!codec_dai->active) {
 		codec_dai->rate = 0;
+		codec_dai->channels = 0;
+		codec_dai->sample_bits = 0;
+	}
 
 	/* Muting the DAC suppresses artifacts caused during digital
 	 * shutdown, for example from stopping clocks.
@@ -525,6 +598,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
+	ret = soc_pcm_params_symmetry(substream, params);
+	if (ret)
+		goto out;
+
 	if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
 		ret = rtd->dai_link->ops->hw_params(substream, params);
 		if (ret < 0) {
@@ -561,9 +638,16 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	/* store the rate for each DAIs */
+	/* store the parameters for each DAIs */
 	cpu_dai->rate = params_rate(params);
+	cpu_dai->channels = params_channels(params);
+	cpu_dai->sample_bits =
+		snd_pcm_format_physical_width(params_format(params));
+
 	codec_dai->rate = params_rate(params);
+	codec_dai->channels = params_channels(params);
+	codec_dai->sample_bits =
+		snd_pcm_format_physical_width(params_format(params));
 
 out:
 	mutex_unlock(&rtd->pcm_mutex);
-- 
cgit v1.2.3


From 8778ac6be25abf0496fc614a3e77ad2ff8300353 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 21 Nov 2013 15:55:06 +0100
Subject: ASoC: Fix build without CONFIG_GPIOLIB

snd_soc_jack_gpio stuff is currently enabled for CONFIG_GPIOLIB
explicitly with ifdef, and this causes build errors on some drivers
such as:
  sound/soc/omap/rx51.c:220:33: error: array type has incomplete element type

Remove ifdef and provide dummy functions for CONFIG_GPIOLIB=n case
instead.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/soc.h | 15 +++++++++++----
 1 file changed, 11 insertions(+), 4 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 1f741cb24f33..f7e1fac51bba 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -334,9 +334,7 @@ struct snd_soc_jack_pin;
 #include <sound/soc-dapm.h>
 #include <sound/soc-dpcm.h>
 
-#ifdef CONFIG_GPIOLIB
 struct snd_soc_jack_gpio;
-#endif
 
 typedef int (*hw_write_t)(void *,const char* ,int);
 
@@ -446,6 +444,17 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 			struct snd_soc_jack_gpio *gpios);
 void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
 			struct snd_soc_jack_gpio *gpios);
+#else
+static inline int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
+					 struct snd_soc_jack_gpio *gpios)
+{
+	return 0;
+}
+
+static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
+					   struct snd_soc_jack_gpio *gpios)
+{
+}
 #endif
 
 /* codec register bit access */
@@ -580,7 +589,6 @@ struct snd_soc_jack_zone {
  *		       to provide more complex checks (eg, reading an
  *		       ADC).
  */
-#ifdef CONFIG_GPIOLIB
 struct snd_soc_jack_gpio {
 	unsigned int gpio;
 	const char *name;
@@ -594,7 +602,6 @@ struct snd_soc_jack_gpio {
 
 	int (*jack_status_check)(void);
 };
-#endif
 
 struct snd_soc_jack {
 	struct mutex mutex;
-- 
cgit v1.2.3


From a3d36bc2aba531328f7311ef57dec7687283ec57 Mon Sep 17 00:00:00 2001
From: Brian Austin <brian.austin@cirrus.com>
Date: Wed, 13 Nov 2013 16:05:40 -0600
Subject: ASoC: cs42l52: Reorganize MICA/B Config and Select

This patch reworks the MICA an MICB config for single-ended or
differential and the selection of which MIC for the single config

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/cs42l52.h    |  6 ------
 sound/soc/codecs/cs42l52.c | 25 ++++---------------------
 2 files changed, 4 insertions(+), 27 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h
index 7c2be4a51894..daa91f327e4f 100644
--- a/include/sound/cs42l52.h
+++ b/include/sound/cs42l52.h
@@ -22,12 +22,6 @@ struct cs42l52_platform_data {
 	/* MICB mode selection 0=Single 1=Differential */
 	unsigned int micb_cfg;
 
-	/* MICA Select 0=MIC1A 1=MIC2A */
-	unsigned int mica_sel;
-
-	/* MICB Select 0=MIC2A 1=MIC2B */
-	unsigned int micb_sel;
-
 	/* Charge Pump Freq. Check datasheet Pg73 */
 	unsigned int chgfreq;
 
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 19ee10b6d6ca..18010639d0c5 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -233,7 +233,7 @@ static const struct soc_enum mic_bias_level_enum =
 	SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
 			ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
 
-static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
+static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" };
 
 static const struct soc_enum mica_enum =
 	SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5,
@@ -243,12 +243,6 @@ static const struct soc_enum micb_enum =
 	SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5,
 			ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
 
-static const struct snd_kcontrol_new mica_mux =
-	SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum);
-
-static const struct snd_kcontrol_new micb_mux =
-	SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum);
-
 static const char * const digital_output_mux_text[] = {"ADC", "DSP"};
 
 static const struct soc_enum digital_output_mux_enum =
@@ -425,6 +419,9 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
 	SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
 
+	SOC_ENUM("MICA Select", mica_enum),
+	SOC_ENUM("MICB Select", micb_enum),
+
 	SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL,
 			      CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv),
 
@@ -550,9 +547,6 @@ static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL,  0,
 			SND_SOC_NOPM, 0, 0),
 
-	SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux),
-	SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux),
-
 	SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1),
 	SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1),
 	SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0),
@@ -1239,17 +1233,6 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 				cs42l52->pdata.micb_cfg <<
 				CS42L52_MIC_CTL_TYPE_SHIFT);
 
-	if (cs42l52->pdata.mica_sel)
-		regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL,
-				   CS42L52_MIC_CTL_MIC_SEL_MASK,
-				cs42l52->pdata.mica_sel <<
-				CS42L52_MIC_CTL_MIC_SEL_SHIFT);
-	if (cs42l52->pdata.micb_sel)
-		regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL,
-				   CS42L52_MIC_CTL_MIC_SEL_MASK,
-				cs42l52->pdata.micb_sel <<
-				CS42L52_MIC_CTL_MIC_SEL_SHIFT);
-
 	if (cs42l52->pdata.chgfreq)
 		regmap_update_bits(cs42l52->regmap, CS42L52_CHARGE_PUMP,
 				   CS42L52_CHARGE_PUMP_MASK,
-- 
cgit v1.2.3


From 44b2ed54036ecec36ad27adf356f0274a72e5f05 Mon Sep 17 00:00:00 2001
From: Brian Austin <brian.austin@cirrus.com>
Date: Thu, 14 Nov 2013 11:46:11 -0600
Subject: ASoC: cs42l52: Make MICA/B mixer dependent on mic config

MICA/B Single-Ended input selection depends on mica/b config so lets
make the mixer controls for them only show for selected mic's

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/cs42l52.h    |  8 ++++----
 sound/soc/codecs/cs42l52.c | 37 ++++++++++++++++++++++++++++++-------
 2 files changed, 34 insertions(+), 11 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h
index daa91f327e4f..bbabf84bdb44 100644
--- a/include/sound/cs42l52.h
+++ b/include/sound/cs42l52.h
@@ -16,11 +16,11 @@ struct cs42l52_platform_data {
 	/* MICBIAS Level. Check datasheet Pg48 */
 	unsigned int micbias_lvl;
 
-	/* MICA mode selection 0=Single 1=Differential */
-	unsigned int mica_cfg;
+	/* MICA mode selection Differential or Single-ended */
+	bool mica_diff_cfg;
 
-	/* MICB mode selection 0=Single 1=Differential */
-	unsigned int micb_cfg;
+	/* MICB mode selection Differential or Single-ended */
+	bool micb_diff_cfg;
 
 	/* Charge Pump Freq. Check datasheet Pg73 */
 	unsigned int chgfreq;
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 18010639d0c5..78d2dd669e89 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -419,9 +419,6 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
 	SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
 
-	SOC_ENUM("MICA Select", mica_enum),
-	SOC_ENUM("MICB Select", micb_enum),
-
 	SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL,
 			      CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv),
 
@@ -528,6 +525,30 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
 };
 
+static const struct snd_kcontrol_new cs42l52_mica_controls[] = {
+	SOC_ENUM("MICA Select", mica_enum),
+};
+
+static const struct snd_kcontrol_new cs42l52_micb_controls[] = {
+	SOC_ENUM("MICB Select", micb_enum),
+};
+
+static int cs42l52_add_mic_controls(struct snd_soc_codec *codec)
+{
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+	struct cs42l52_platform_data *pdata = &cs42l52->pdata;
+
+	if (!pdata->mica_diff_cfg)
+		snd_soc_add_codec_controls(codec, cs42l52_mica_controls,
+				     ARRAY_SIZE(cs42l52_mica_controls));
+
+	if (!pdata->micb_diff_cfg)
+		snd_soc_add_codec_controls(codec, cs42l52_micb_controls,
+				     ARRAY_SIZE(cs42l52_micb_controls));
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = {
 
 	SND_SOC_DAPM_INPUT("AIN1L"),
@@ -1104,6 +1125,8 @@ static int cs42l52_probe(struct snd_soc_codec *codec)
 	}
 	regcache_cache_only(cs42l52->regmap, true);
 
+	cs42l52_add_mic_controls(codec);
+
 	cs42l52_init_beep(codec);
 
 	cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1221,16 +1244,16 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 			reg & 0xFF);
 
 	/* Set Platform Data */
-	if (cs42l52->pdata.mica_cfg)
+	if (cs42l52->pdata.mica_diff_cfg)
 		regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL,
 				   CS42L52_MIC_CTL_TYPE_MASK,
-				cs42l52->pdata.mica_cfg <<
+				cs42l52->pdata.mica_diff_cfg <<
 				CS42L52_MIC_CTL_TYPE_SHIFT);
 
-	if (cs42l52->pdata.micb_cfg)
+	if (cs42l52->pdata.micb_diff_cfg)
 		regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL,
 				   CS42L52_MIC_CTL_TYPE_MASK,
-				cs42l52->pdata.micb_cfg <<
+				cs42l52->pdata.micb_diff_cfg <<
 				CS42L52_MIC_CTL_TYPE_SHIFT);
 
 	if (cs42l52->pdata.chgfreq)
-- 
cgit v1.2.3


From 21585ee848078b12d0d1a513e93936bf96b444a0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 28 Nov 2013 08:50:32 +0100
Subject: ASoC: Add resource managed snd_dmaengine_pcm_register()

For many drivers using the generic dmaengine PCM driver one of the few (or the
only) things left to do in the drivers remove function is to unregister the PCM
device. This patch adds a resource managed version of snd_dmaengine_pcm_register()
which makes it possible to simplify the remove function as well as the error
path in the probe function for those drivers.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/dmaengine_pcm.h |  4 ++++
 sound/soc/soc-devres.c        | 41 +++++++++++++++++++++++++++++++++++++++++
 2 files changed, 45 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index 15017311f2e9..4ef986cab182 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -140,6 +140,10 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	unsigned int flags);
 void snd_dmaengine_pcm_unregister(struct device *dev);
 
+int devm_snd_dmaengine_pcm_register(struct device *dev,
+	const struct snd_dmaengine_pcm_config *config,
+	unsigned int flags);
+
 int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params,
 	struct dma_slave_config *slave_config);
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index b1d732255c02..999861942d28 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -12,6 +12,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
 
 static void devm_component_release(struct device *dev, void *res)
 {
@@ -84,3 +85,43 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
 	return ret;
 }
 EXPORT_SYMBOL_GPL(devm_snd_soc_register_card);
+
+#ifdef CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM
+
+static void devm_dmaengine_pcm_release(struct device *dev, void *res)
+{
+	snd_dmaengine_pcm_unregister(*(struct device **)res);
+}
+
+/**
+ * devm_snd_dmaengine_pcm_register - resource managed dmaengine PCM registration
+ * @dev: The parent device for the PCM device
+ * @config: Platform specific PCM configuration
+ * @flags: Platform specific quirks
+ *
+ * Register a dmaengine based PCM device with automatic unregistration when the
+ * device is unregistered.
+ */
+int devm_snd_dmaengine_pcm_register(struct device *dev,
+	const struct snd_dmaengine_pcm_config *config, unsigned int flags)
+{
+	struct device **ptr;
+	int ret;
+
+	ptr = devres_alloc(devm_dmaengine_pcm_release, sizeof(*ptr), GFP_KERNEL);
+	if (!ptr)
+		return -ENOMEM;
+
+	ret = snd_dmaengine_pcm_register(dev, config, flags);
+	if (ret == 0) {
+		*ptr = dev;
+		devres_add(dev, ptr);
+	} else {
+		devres_free(ptr);
+	}
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register);
+
+#endif
-- 
cgit v1.2.3


From 194c7dea00c68c1b1f8ff26304fa937a006f66dd Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 3 Dec 2013 14:26:34 -0700
Subject: ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config

Add fields to struct snd_dmaengine_pcm_config to allow custom:

- DMA channel names.

  This is useful when the default "tx" and "rx" channel names don't
  apply, for example if a HW module supports multiple channels, each
  having different DMA channel names. This is the case with the FIFOs
  in Tegra's AHUB. This new facility can replace
  SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME.

- DMA device

  This allows requesting DMA channels for a device other than the device
  which is registering the "PCM" driver. This is quite unusual, but is
  currently useful on Tegra. In much HW, and in Tegra20, each DAI HW
  module contains its own FIFOs which DMA writes to. However, in Tegra30,
  the DMA FIFOs were split out AHUB HW module, which then routes the data
  through a cross-bar, and into the DAI HW modules. However, the current
  ASoC driver structure does not expose this detail, and acts as if the
  FIFOs are still part of the DAI HW modules. Consequently, the "PCM"
  driver is registered with the DAI HW module, yet the DMA channels must
  be looked up in the AHUB HW module's device tree node. This new config
  field allows that to happen. Eventually, the Tegra drivers will be
  reworked to fully expose the AHUB, and this config field can be
  removed.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/dmaengine_pcm.h         |  6 ++++++
 sound/soc/soc-generic-dmaengine-pcm.c | 18 ++++++++++++++++--
 2 files changed, 22 insertions(+), 2 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index 4ef986cab182..eb73a3a39ec2 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -114,6 +114,10 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
  * @compat_filter_fn: Will be used as the filter function when requesting a
  *  channel for platforms which do not use devicetree. The filter parameter
  *  will be the DAI's DMA data.
+ * @dma_dev: If set, request DMA channel on this device rather than the DAI
+ *  device.
+ * @chan_names: If set, these custom DMA channel names will be requested at
+ *  registration time.
  * @pcm_hardware: snd_pcm_hardware struct to be used for the PCM.
  * @prealloc_buffer_size: Size of the preallocated audio buffer.
  *
@@ -130,6 +134,8 @@ struct snd_dmaengine_pcm_config {
 			struct snd_soc_pcm_runtime *rtd,
 			struct snd_pcm_substream *substream);
 	dma_filter_fn compat_filter_fn;
+	struct device *dma_dev;
+	const char *chan_names[SNDRV_PCM_STREAM_LAST + 1];
 
 	const struct snd_pcm_hardware *pcm_hardware;
 	unsigned int prealloc_buffer_size;
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 1cb3494cf278..5b70c556fba3 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -288,7 +288,7 @@ static const char * const dmaengine_pcm_dma_channel_names[] = {
 };
 
 static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
-	struct device *dev)
+	struct device *dev, const struct snd_dmaengine_pcm_config *config)
 {
 	unsigned int i;
 	const char *name;
@@ -298,12 +298,26 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 	    !dev->of_node)
 		return;
 
+	if (config->dma_dev) {
+		/*
+		 * If this warning is seen, it probably means that your Linux
+		 * device structure does not match your HW device structure.
+		 * It would be best to refactor the Linux device structure to
+		 * correctly match the HW structure.
+		 */
+		dev_warn(dev, "DMA channels sourced from device %s",
+			 dev_name(config->dma_dev));
+		dev = config->dma_dev;
+	}
+
 	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
 	     i++) {
 		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
 			name = "rx-tx";
 		else
 			name = dmaengine_pcm_dma_channel_names[i];
+		if (config->chan_names[i])
+			name = config->chan_names[i];
 		pcm->chan[i] = dma_request_slave_channel(dev, name);
 		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
 			break;
@@ -346,7 +360,7 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	pcm->config = config;
 	pcm->flags = flags;
 
-	dmaengine_pcm_request_chan_of(pcm, dev);
+	dmaengine_pcm_request_chan_of(pcm, dev, config);
 
 	if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
 		ret = snd_soc_add_platform(dev, &pcm->platform,
-- 
cgit v1.2.3


From e1771bcf99b0dc91f4ba645c1740fd5031702f49 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 10 Dec 2013 12:35:25 -0700
Subject: ASoC: SPEAr: remove custom DMA alloc compat function

spear_pcm_request_chan() is almost identical to
dmaengine_pcm_compat_request_channel(), with the exception that the
latter:

a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data
   pointer rather than some custom type.

b) dma_data->filter_data rather than dma_data should be passed to
   snd_dmaengine_pcm_request_channel() as the filter data.

Make minor changes to the SPEAr DAI drivers so that those two conditions
are met. This allows removal of the custom .compat_request_channel().

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/spear_dma.h   |  1 -
 sound/soc/spear/spdif_in.c  | 10 +++++++---
 sound/soc/spear/spdif_out.c | 10 +++++++---
 sound/soc/spear/spear_pcm.c | 21 +++++++--------------
 sound/soc/spear/spear_pcm.h |  4 +++-
 5 files changed, 24 insertions(+), 22 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h
index 1b365bfdfb37..65aca51fe255 100644
--- a/include/sound/spear_dma.h
+++ b/include/sound/spear_dma.h
@@ -29,7 +29,6 @@ struct spear_dma_data {
 	dma_addr_t addr;
 	u32 max_burst;
 	enum dma_slave_buswidth addr_width;
-	bool (*filter)(struct dma_chan *chan, void *slave);
 };
 
 #endif /* SPEAR_DMA_H */
diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index 4627110f3441..4ab442a63d7e 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -18,6 +18,7 @@
 #include <linux/ioport.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <sound/dmaengine_pcm.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
@@ -38,6 +39,8 @@ struct spdif_in_dev {
 	struct device *dev;
 	void (*reset_perip)(void);
 	int irq;
+	struct snd_dmaengine_dai_dma_data dma_params_rx;
+	struct snd_dmaengine_pcm_config config;
 };
 
 static void spdif_in_configure(struct spdif_in_dev *host)
@@ -54,7 +57,8 @@ static int spdif_in_dai_probe(struct snd_soc_dai *dai)
 {
 	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
 
-	dai->capture_dma_data = &host->dma_params;
+	host->dma_params_rx.filter_data = &host->dma_params;
+	dai->capture_dma_data = &host->dma_params_rx;
 
 	return 0;
 }
@@ -245,7 +249,6 @@ static int spdif_in_probe(struct platform_device *pdev)
 	host->dma_params.addr = res_fifo->start;
 	host->dma_params.max_burst = 16;
 	host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
-	host->dma_params.filter = pdata->filter;
 	host->reset_perip = pdata->reset_perip;
 
 	host->dev = &pdev->dev;
@@ -263,7 +266,8 @@ static int spdif_in_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	return devm_spear_pcm_platform_register(&pdev->dev);
+	return devm_spear_pcm_platform_register(&pdev->dev, &host->config,
+						pdata->filter);
 }
 
 static struct platform_driver spdif_in_driver = {
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index 731a1e0cfeaa..fe99f461aff0 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -18,6 +18,7 @@
 #include <linux/ioport.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <sound/dmaengine_pcm.h>
 #include <sound/soc.h>
 #include <sound/spear_dma.h>
 #include <sound/spear_spdif.h>
@@ -36,6 +37,8 @@ struct spdif_out_dev {
 	struct spdif_out_params saved_params;
 	u32 running;
 	void __iomem *io_base;
+	struct snd_dmaengine_dai_dma_data dma_params_tx;
+	struct snd_dmaengine_pcm_config config;
 };
 
 static void spdif_out_configure(struct spdif_out_dev *host)
@@ -245,7 +248,8 @@ static int spdif_soc_dai_probe(struct snd_soc_dai *dai)
 {
 	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
 
-	dai->playback_dma_data = &host->dma_params;
+	host->dma_params_tx.filter_data = &host->dma_params;
+	dai->playback_dma_data = &host->dma_params_tx;
 
 	return snd_soc_add_dai_controls(dai, spdif_out_controls,
 				ARRAY_SIZE(spdif_out_controls));
@@ -304,7 +308,6 @@ static int spdif_out_probe(struct platform_device *pdev)
 	host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA;
 	host->dma_params.max_burst = 16;
 	host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
-	host->dma_params.filter = pdata->filter;
 
 	dev_set_drvdata(&pdev->dev, host);
 
@@ -313,7 +316,8 @@ static int spdif_out_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	return devm_spear_pcm_platform_register(&pdev->dev);
+	return devm_spear_pcm_platform_register(&pdev->dev, &host->config,
+						pdata->filter);
 }
 
 #ifdef CONFIG_PM
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index f288724961da..0e5a8f35d0ad 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -32,26 +32,19 @@ static const struct snd_pcm_hardware spear_pcm_hardware = {
 	.fifo_size = 0, /* fifo size in bytes */
 };
 
-static struct dma_chan *spear_pcm_request_chan(struct snd_soc_pcm_runtime *rtd,
-	struct snd_pcm_substream *substream)
-{
-	struct spear_dma_data *dma_data;
-
-	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
-	return snd_dmaengine_pcm_request_channel(dma_data->filter, dma_data);
-}
-
 static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = {
 	.pcm_hardware = &spear_pcm_hardware,
-	.compat_request_channel = spear_pcm_request_chan,
 	.prealloc_buffer_size = 16 * 1024,
 };
 
-int devm_spear_pcm_platform_register(struct device *dev)
+int devm_spear_pcm_platform_register(struct device *dev,
+			struct snd_dmaengine_pcm_config *config,
+			bool (*filter)(struct dma_chan *chan, void *slave))
 {
-	return devm_snd_dmaengine_pcm_register(dev,
-		&spear_dmaengine_pcm_config,
+	*config = spear_dmaengine_pcm_config;
+	config->compat_filter_fn = filter;
+
+	return snd_dmaengine_pcm_register(dev, config,
 		SND_DMAENGINE_PCM_FLAG_NO_DT |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h
index 631e2aa1fb33..9b0ca62d6f02 100644
--- a/sound/soc/spear/spear_pcm.h
+++ b/sound/soc/spear/spear_pcm.h
@@ -17,6 +17,8 @@
 #ifndef __SPEAR_PCM_H__
 #define __SPEAR_PCM_H__
 
-int devm_spear_pcm_platform_register(struct device *dev);
+int devm_spear_pcm_platform_register(struct device *dev,
+			struct snd_dmaengine_pcm_config *config,
+			bool (*filter)(struct dma_chan *chan, void *slave));
 
 #endif
-- 
cgit v1.2.3


From 8c5178fca4ce5a57711ea14b807648e19b105d0e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Tue, 24 Dec 2013 12:24:28 +0000
Subject: ALSA: Add params_width() helpers

Add helpers for obtaining the width of a format directly from params
since this is expected to become a common operation in ASoC.

Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/pcm_params.h | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index 37ae12e0ab06..6b1c78f05fab 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -354,4 +354,16 @@ params_period_bytes(const struct snd_pcm_hw_params *p)
 		params_channels(p)) / 8;
 }
 
+static inline int
+params_width(const struct snd_pcm_hw_params *p)
+{
+	return snd_pcm_format_width(params_format(p));
+}
+
+static inline int
+params_physical_width(const struct snd_pcm_hw_params *p)
+{
+	return snd_pcm_format_physical_width(params_format(p));
+}
+
 #endif /* __SOUND_PCM_PARAMS_H */
-- 
cgit v1.2.3


From ef749400434cefd14fe02fe3de9e9f0125b2256d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:28:51 -0800
Subject: ASoC: rsnd: add SRC (Sampling Rate Converter) support

This patch adds SRC support to Renesas sound driver.
SRC converts sampling rate between codec <-> cpu.
It needs special codec chip,
or very simple DA/AD converter to use it.
This patch was tested via ak4554 codec,
and supports Gen1 only at this point.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/rcar_snd.h |   1 +
 sound/soc/sh/rcar/adg.c  |  73 +++++++++++++++++++++++
 sound/soc/sh/rcar/gen.c  |  10 ++++
 sound/soc/sh/rcar/rsnd.h |  18 ++++++
 sound/soc/sh/rcar/scu.c  | 152 ++++++++++++++++++++++++++++++++++++++++++++---
 sound/soc/sh/rcar/ssi.c  |   2 +-
 6 files changed, 248 insertions(+), 8 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h
index a818ff76b138..e147498abe50 100644
--- a/include/sound/rcar_snd.h
+++ b/include/sound/rcar_snd.h
@@ -58,6 +58,7 @@ struct rsnd_ssi_platform_info {
 
 struct rsnd_scu_platform_info {
 	u32 flags;
+	u32 convert_rate; /* sampling rate convert */
 };
 
 /*
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 2e71a7bda4c2..a53235c4d1b0 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -30,6 +30,79 @@ struct rsnd_adg {
 	     i++, (pos) = adg->clk[i])
 #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
 
+static int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
+					 struct rsnd_mod *mod,
+					 unsigned int src_rate,
+					 unsigned int dst_rate)
+{
+	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	int idx, sel, div, shift;
+	u32 mask, val;
+	int id = rsnd_mod_id(mod);
+	unsigned int sel_rate [] = {
+		clk_get_rate(adg->clk[CLKA]),	/* 000: CLKA */
+		clk_get_rate(adg->clk[CLKB]),	/* 001: CLKB */
+		clk_get_rate(adg->clk[CLKC]),	/* 010: CLKC */
+		0,				/* 011: MLBCLK (not used) */
+		adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */
+		adg->rbgb_rate_for_48khz_div_6,	/* 101: RBGB */
+	};
+
+	/* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */
+	for (sel = 0; sel < ARRAY_SIZE(sel_rate); sel++) {
+		for (div  = 128,	idx = 0;
+		     div <= 2048;
+		     div *= 2,		idx++) {
+			if (src_rate == sel_rate[sel] / div) {
+				val = (idx << 4) | sel;
+				goto find_rate;
+			}
+		}
+	}
+	dev_err(dev, "can't find convert src clk\n");
+	return -EINVAL;
+
+find_rate:
+	shift	= (id % 4) * 8;
+	mask	= 0xFF << shift;
+	val	= val << shift;
+
+	dev_dbg(dev, "adg convert src clk = %02x\n", val);
+
+	switch (id / 4) {
+	case 0:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val);
+		break;
+	case 1:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val);
+		break;
+	case 2:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val);
+		break;
+	}
+
+	/*
+	 * Gen1 doesn't need dst_rate settings,
+	 * since it uses SSI WS pin.
+	 * see also rsnd_src_set_route_if_gen1()
+	 */
+
+	return 0;
+}
+
+int rsnd_adg_set_convert_clk(struct rsnd_priv *priv,
+			     struct rsnd_mod *mod,
+			     unsigned int src_rate,
+			     unsigned int dst_rate)
+{
+	if (rsnd_is_gen1(priv))
+		return rsnd_adg_set_convert_clk_gen1(priv, mod,
+						     src_rate, dst_rate);
+
+	return -EINVAL;
+}
+
 static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
 {
 	int id = rsnd_mod_id(mod);
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 862758d3ec06..add088bd4b2a 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -318,13 +318,23 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
 		RSND_GEN1_S_REG(gen, SRU,	SSI_MODE0,	0xD0),
 		RSND_GEN1_S_REG(gen, SRU,	SSI_MODE1,	0xD4),
 		RSND_GEN1_M_REG(gen, SRU,	BUSIF_MODE,	0x20,	0x4),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_ROUTE_MODE0,0x50,	0x8),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_SWRSR,	0x200,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_SRCIR,	0x204,	0x40),
 		RSND_GEN1_M_REG(gen, SRU,	SRC_ADINR,	0x214,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_IFSCR,	0x21c,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_IFSVR,	0x220,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_SRCCR,	0x224,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_MNFSR,	0x228,	0x40),
 
 		RSND_GEN1_S_REG(gen, ADG,	BRRA,		0x00),
 		RSND_GEN1_S_REG(gen, ADG,	BRRB,		0x04),
 		RSND_GEN1_S_REG(gen, ADG,	SSICKR,		0x08),
 		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL0,	0x0c),
 		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL1,	0x10),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL3,	0x18),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL4,	0x1c),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL5,	0x20),
 
 		RSND_GEN1_M_REG(gen, SSI,	SSICR,		0x00,	0x40),
 		RSND_GEN1_M_REG(gen, SSI,	SSISR,		0x04,	0x40),
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 3774dfcfaf0f..4ca66cd899c8 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -41,7 +41,14 @@ enum rsnd_reg {
 	RSND_REG_SSI_MODE1,
 	RSND_REG_BUSIF_MODE,
 	RSND_REG_INT_ENABLE,		/* for Gen2 */
+	RSND_REG_SRC_ROUTE_MODE0,
+	RSND_REG_SRC_SWRSR,
+	RSND_REG_SRC_SRCIR,
 	RSND_REG_SRC_ADINR,
+	RSND_REG_SRC_IFSCR,
+	RSND_REG_SRC_IFSVR,
+	RSND_REG_SRC_SRCCR,
+	RSND_REG_SRC_MNFSR,
 
 	/* ADG */
 	RSND_REG_BRRA,
@@ -50,6 +57,9 @@ enum rsnd_reg {
 	RSND_REG_AUDIO_CLK_SEL0,
 	RSND_REG_AUDIO_CLK_SEL1,
 	RSND_REG_AUDIO_CLK_SEL2,
+	RSND_REG_AUDIO_CLK_SEL3,	/* for Gen1 */
+	RSND_REG_AUDIO_CLK_SEL4,	/* for Gen1 */
+	RSND_REG_AUDIO_CLK_SEL5,	/* for Gen1 */
 
 	/* SSI */
 	RSND_REG_SSICR,
@@ -227,6 +237,10 @@ int rsnd_adg_probe(struct platform_device *pdev,
 		   struct rsnd_priv *priv);
 void rsnd_adg_remove(struct platform_device *pdev,
 		   struct rsnd_priv *priv);
+int rsnd_adg_set_convert_clk(struct rsnd_priv *priv,
+			     struct rsnd_mod *mod,
+			     unsigned int src_rate,
+			     unsigned int dst_rate);
 
 /*
  *	R-Car sound priv
@@ -280,6 +294,10 @@ void rsnd_scu_remove(struct platform_device *pdev,
 		     struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id);
 bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod);
+unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv,
+				   struct rsnd_mod *ssi_mod,
+				   struct snd_pcm_runtime *runtime);
+
 #define rsnd_scu_nr(priv) ((priv)->scu_nr)
 
 /*
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
index 5f4f57206faf..1406dd8d9ed2 100644
--- a/sound/soc/sh/rcar/scu.c
+++ b/sound/soc/sh/rcar/scu.c
@@ -13,9 +13,13 @@
 struct rsnd_scu {
 	struct rsnd_scu_platform_info *info; /* rcar_snd.h */
 	struct rsnd_mod mod;
+	struct clk *clk;
 };
 
 #define rsnd_scu_mode_flags(p) ((p)->info->flags)
+#define rsnd_scu_convert_rate(p) ((p)->info->convert_rate)
+
+#define RSND_SCU_NAME_SIZE 16
 
 /*
  * ADINR
@@ -26,6 +30,15 @@ struct rsnd_scu {
 #define OTBL_18		(6 << 16)
 #define OTBL_16		(8 << 16)
 
+/*
+ *		image of SRC (Sampling Rate Converter)
+ *
+ * 96kHz   <-> +-----+	48kHz	+-----+	 48kHz	+-------+
+ * 48kHz   <-> | SRC | <------>	| SSI |	<----->	| codec |
+ * 44.1kHz <-> +-----+		+-----+		+-------+
+ * ...
+ *
+ */
 
 #define rsnd_mod_to_scu(_mod)	\
 	container_of((_mod), struct rsnd_scu, mod)
@@ -56,7 +69,7 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 		{ 0x3, 28, }, /* 7 */
 		{ 0x3, 30, }, /* 8 */
 	};
-
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
 	u32 mask;
 	u32 val;
 	int shift;
@@ -86,9 +99,18 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 	 */
 	shift	= (id % 4) * 8;
 	mask	= 0x1F << shift;
-	if (8 == id) /* SRU8 is very special */
+
+	/*
+	 * ADG is used as source clock if SRC was used,
+	 * then, SSI WS is used as destination clock.
+	 * SSI WS is used as source clock if SRC is not used
+	 * (when playback, source/destination become reverse when capture)
+	 */
+	if (rsnd_scu_convert_rate(scu))	/* use ADG */
+		val = 0;
+	else if (8 == id)		/* use SSI WS, but SRU8 is special */
 		val = id << shift;
-	else
+	else				/* use SSI WS */
 		val = (id + 1) << shift;
 
 	switch (id / 4) {
@@ -106,14 +128,45 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 	return 0;
 }
 
-static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv,
+unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv,
+				   struct rsnd_mod *ssi_mod,
+				   struct snd_pcm_runtime *runtime)
+{
+	struct rsnd_scu *scu;
+	unsigned int rate;
+
+	/* this function is assuming SSI id = SCU id here */
+	scu = rsnd_mod_to_scu(rsnd_scu_mod_get(priv, rsnd_mod_id(ssi_mod)));
+
+	/*
+	 * return convert rate if SRC is used,
+	 * otherwise, return runtime->rate as usual
+	 */
+	rate = rsnd_scu_convert_rate(scu);
+	if (!rate)
+		rate = runtime->rate;
+
+	return rate;
+}
+
+static int rsnd_scu_convert_rate_ctrl(struct rsnd_priv *priv,
 			      struct rsnd_mod *mod,
 			      struct rsnd_dai *rdai,
 			      struct rsnd_dai_stream *io)
 {
 	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+	u32 convert_rate = rsnd_scu_convert_rate(scu);
 	u32 adinr = runtime->channels;
 
+	/* set/clear soft reset */
+	rsnd_mod_write(mod, SRC_SWRSR, 0);
+	rsnd_mod_write(mod, SRC_SWRSR, 1);
+
+	/* Initialize the operation of the SRC internal circuits */
+	rsnd_mod_write(mod, SRC_SRCIR, 1);
+
+	/* Set channel number and output bit length */
 	switch (runtime->sample_bits) {
 	case 16:
 		adinr |= OTBL_16;
@@ -124,9 +177,42 @@ static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv,
 	default:
 		return -EIO;
 	}
-
 	rsnd_mod_write(mod, SRC_ADINR, adinr);
 
+	if (convert_rate) {
+		u32 fsrate = 0x0400000 / convert_rate * runtime->rate;
+		int ret;
+
+		/* Enable the initial value of IFS */
+		rsnd_mod_write(mod, SRC_IFSCR, 1);
+
+		/* Set initial value of IFS */
+		rsnd_mod_write(mod, SRC_IFSVR, fsrate);
+
+		/* Select SRC mode (fixed value) */
+		rsnd_mod_write(mod, SRC_SRCCR, 0x00010110);
+
+		/* Set the restriction value of the FS ratio (98%) */
+		rsnd_mod_write(mod, SRC_MNFSR, fsrate / 100 * 98);
+
+		if (rsnd_is_gen1(priv)) {
+			/* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */
+		}
+
+		/* set convert clock */
+		ret = rsnd_adg_set_convert_clk(priv, mod,
+					       runtime->rate,
+					       convert_rate);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* Cancel the initialization and operate the SRC function */
+	rsnd_mod_write(mod, SRC_SRCIR, 0);
+
+	/* use DMA transfer */
+	rsnd_mod_write(mod, BUSIF_MODE, 1);
+
 	return 0;
 }
 
@@ -135,6 +221,7 @@ static int rsnd_scu_transfer_start(struct rsnd_priv *priv,
 				   struct rsnd_dai *rdai,
 				   struct rsnd_dai_stream *io)
 {
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
 	int id = rsnd_mod_id(mod);
 	u32 val;
 
@@ -143,7 +230,28 @@ static int rsnd_scu_transfer_start(struct rsnd_priv *priv,
 		rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val);
 	}
 
-	rsnd_mod_write(mod, BUSIF_MODE, 1);
+	if (rsnd_scu_convert_rate(scu))
+		rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1);
+
+	return 0;
+}
+
+static int rsnd_scu_transfer_stop(struct rsnd_priv *priv,
+				  struct rsnd_mod *mod,
+				  struct rsnd_dai *rdai,
+				  struct rsnd_dai_stream *io)
+{
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+	int id = rsnd_mod_id(mod);
+	u32 mask;
+
+	if (rsnd_is_gen1(priv)) {
+		mask = (1 << id);
+		rsnd_mod_bset(mod, SRC_ROUTE_CTRL, mask, 0);
+	}
+
+	if (rsnd_scu_convert_rate(scu))
+		rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0);
 
 	return 0;
 }
@@ -161,6 +269,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 			  struct rsnd_dai_stream *io)
 {
 	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	int ret;
 
@@ -175,13 +284,15 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 		return 0;
 	}
 
+	clk_enable(scu->clk);
+
 	/* it use DMA transter */
 
 	ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io);
 	if (ret < 0)
 		return ret;
 
-	ret = rsnd_scu_rate_ctrl(priv, mod, rdai, io);
+	ret = rsnd_scu_convert_rate_ctrl(priv, mod, rdai, io);
 	if (ret < 0)
 		return ret;
 
@@ -194,9 +305,27 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 	return 0;
 }
 
+static int rsnd_scu_stop(struct rsnd_mod *mod,
+			  struct rsnd_dai *rdai,
+			  struct rsnd_dai_stream *io)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+
+	if (!rsnd_scu_hpbif_is_enable(mod))
+		return 0;
+
+	rsnd_scu_transfer_stop(priv, mod, rdai, io);
+
+	clk_disable(scu->clk);
+
+	return 0;
+}
+
 static struct rsnd_mod_ops rsnd_scu_ops = {
 	.name	= "scu",
 	.start	= rsnd_scu_start,
+	.stop	= rsnd_scu_stop,
 };
 
 struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id)
@@ -212,6 +341,8 @@ int rsnd_scu_probe(struct platform_device *pdev,
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct rsnd_scu *scu;
+	struct clk *clk;
+	char name[RSND_SCU_NAME_SIZE];
 	int i, nr;
 
 	/*
@@ -228,9 +359,16 @@ int rsnd_scu_probe(struct platform_device *pdev,
 	priv->scu	= scu;
 
 	for_each_rsnd_scu(scu, priv, i) {
+		snprintf(name, RSND_SCU_NAME_SIZE, "scu.%d", i);
+
+		clk = devm_clk_get(dev, name);
+		if (IS_ERR(clk))
+			return PTR_ERR(clk);
+
 		rsnd_mod_init(priv, &scu->mod,
 			      &rsnd_scu_ops, i);
 		scu->info = &info->scu_info[i];
+		scu->clk = clk;
 
 		dev_dbg(dev, "SCU%d probed\n", i);
 	}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 2db9711549f5..b7cd06be9436 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -200,7 +200,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
 		1, 2, 4, 8, 16, 6, 12,
 	};
 	unsigned int main_rate;
-	unsigned int rate = runtime->rate;
+	unsigned int rate = rsnd_scu_get_ssi_rate(priv, &ssi->mod, runtime);
 
 	/*
 	 * Find best clock, and try to start ADG
-- 
cgit v1.2.3