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ASoC: es8328 codec improvements

PATCH In review John Keeping 2 participants 23 messages May 9, 2016 – May 13, 2016

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John Keeping

author
#0 lore raw

cover ASoC: es8328 codec improvements

The first 4 patches are bug fixes, then the next 4 add support for more sample rates and formats.

The final patch I'm slightly unsure about since the hardware does allow different playback and capture rates, but it only allows 8kHz and 48kHz or 8.0182kHz and 44.1kHz, all other rates have to be symmetric.

All of these patches have been tested on a Radxa Rock2 Square with rockchip-i2s.

John Keeping (9):
ASoC: es8328: Move clock setup to hw_params
ASoC: es8328: Fix ADC format setup
ASoC: es8328: Fix mask for VMIDSEL
ASoC: es8328: Use single R/W for regmap
ASoC: es8328: Use more suitable definition for mic bias
ASoC: es8328: Move sample size setup to hw_params
ASoC: es8328: Support more sample formats
ASoC: es8328: Support more sample rates
ASoC: es8328: Set symmetric rates

 sound/soc/codecs/es8328.c | 198 +++++++++++++++++++++++++++++++++++-----------
 sound/soc/codecs/es8328.h |  23 ++++--
 2 files changed, 168 insertions(+), 53 deletions(-)
Signature · 1 line
2.8.0.rc4.238.g874082a
author
#1 lore raw

PATCH 1/9 ASoC: es8328: Move clock setup to hw_params

This ensures that the clock is setup after its frequency has been set; the existing code in set_dai_fmt may be called before the clock rate has been set resulting in an incorrect configuration.

Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.c | 31 ++++++++++++++++---------------
 1 file changed, 16 insertions(+), 15 deletions(-)
sound/soc/codecs/es8328.c +16 -15
index afa6c5db9dcc..3ca89ae32889 100644
@@ -445,9 +445,10 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate;
+	int clk_rate = clk_get_rate(es8328->clk);
 	int i;
 	int reg;
+	int val;
 	u8 ratio;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

@@ -455,16 +456,24 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 	else
 		reg = ES8328_ADCCONTROL5;
 
-	clk_rate = clk_get_rate(es8328->clk);
-
-	if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
-		(clk_rate != ES8328_SYSCLK_RATE_2X)) {
+	switch (clk_rate) {
+	case ES8328_SYSCLK_RATE_1X:
+		val = 0;
+		break;
+	case ES8328_SYSCLK_RATE_2X:
+		val = ES8328_MASTERMODE_MCLKDIV2;
+		break;
+	default:
 		dev_err(codec->dev,
 			"%s: clock is running at %d Hz, not %d or %d Hz\n",
 			 __func__, clk_rate,
 			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
 		return -EINVAL;
 	}
+	ret = snd_soc_update_bits(codec, ES8328_MASTERMODE,
+			ES8328_MASTERMODE_MCLKDIV2, val);
+	if (ret < 0)
+		return ret;
 
 	/* find master mode MCLK to sampling frequency ratio */
 	ratio = mclk_ratios[0].rate;

@@ -484,8 +493,6 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate;
 	u8 mode = ES8328_DACCONTROL1_DACWL_16;
 
 	/* set master/slave audio interface */

@@ -515,14 +522,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
 
 	/* Master serial port mode, with BCLK generated automatically */
-	clk_rate = clk_get_rate(es8328->clk);
-	if (clk_rate == ES8328_SYSCLK_RATE_1X)
-		snd_soc_write(codec, ES8328_MASTERMODE,
-				ES8328_MASTERMODE_MSC);
-	else
-		snd_soc_write(codec, ES8328_MASTERMODE,
-				ES8328_MASTERMODE_MCLKDIV2 |
-				ES8328_MASTERMODE_MSC);
+	snd_soc_update_bits(codec, ES8328_MASTERMODE,
+			ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC);
 
 	return 0;
 }
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#2 lore raw

Applied "ASoC: es8328: Move clock setup to hw_params" to the asoc tree

The patch

ASoC: es8328: Move clock setup to hw_params

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From 420c470d6b5c2924a3182edf5b002870ff770331 Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Mon, 9 May 2016 12:24:29 +0100
Subject: [PATCH] ASoC: es8328: Move clock setup to hw_params

This ensures that the clock is setup after its frequency has been set; the existing code in set_dai_fmt may be called before the clock rate has been set resulting in an incorrect configuration.

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 sound/soc/codecs/es8328.c | 31 ++++++++++++++++---------------
 1 file changed, 16 insertions(+), 15 deletions(-)
sound/soc/codecs/es8328.c +16 -15
index afa6c5db9dcc..3ca89ae32889 100644
@@ -445,9 +445,10 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate;
+	int clk_rate = clk_get_rate(es8328->clk);
 	int i;
 	int reg;
+	int val;
 	u8 ratio;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

@@ -455,16 +456,24 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 	else
 		reg = ES8328_ADCCONTROL5;
 
-	clk_rate = clk_get_rate(es8328->clk);
-
-	if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
-		(clk_rate != ES8328_SYSCLK_RATE_2X)) {
+	switch (clk_rate) {
+	case ES8328_SYSCLK_RATE_1X:
+		val = 0;
+		break;
+	case ES8328_SYSCLK_RATE_2X:
+		val = ES8328_MASTERMODE_MCLKDIV2;
+		break;
+	default:
 		dev_err(codec->dev,
 			"%s: clock is running at %d Hz, not %d or %d Hz\n",
 			 __func__, clk_rate,
 			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
 		return -EINVAL;
 	}
+	ret = snd_soc_update_bits(codec, ES8328_MASTERMODE,
+			ES8328_MASTERMODE_MCLKDIV2, val);
+	if (ret < 0)
+		return ret;
 
 	/* find master mode MCLK to sampling frequency ratio */
 	ratio = mclk_ratios[0].rate;

@@ -484,8 +493,6 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate;
 	u8 mode = ES8328_DACCONTROL1_DACWL_16;
 
 	/* set master/slave audio interface */

@@ -515,14 +522,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
 
 	/* Master serial port mode, with BCLK generated automatically */
-	clk_rate = clk_get_rate(es8328->clk);
-	if (clk_rate == ES8328_SYSCLK_RATE_1X)
-		snd_soc_write(codec, ES8328_MASTERMODE,
-				ES8328_MASTERMODE_MSC);
-	else
-		snd_soc_write(codec, ES8328_MASTERMODE,
-				ES8328_MASTERMODE_MCLKDIV2 |
-				ES8328_MASTERMODE_MSC);
+	snd_soc_update_bits(codec, ES8328_MASTERMODE,
+			ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC);
 
 	return 0;
 }
Signature · 1 line
2.8.1
author
#3 lore raw

PATCH 2/9 ASoC: es8328: Fix ADC format setup

In reply to John Keeping

The ADCCONTROL4 and DACCONTROL1 registers are similar but not identical, with the DACCONTROL1 having each field starting one bit higher than ADCCONTROL4.

Instead of introducing a magic shift, add new constants for the values in ADCCONTROL4 and use a second variable to setup the ADC.

Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.c | 16 ++++++++++------
 sound/soc/codecs/es8328.h | 15 +++++++++++++++
 2 files changed, 25 insertions(+), 6 deletions(-)
sound/soc/codecs/es8328.c +10 -6
index 3ca89ae32889..63e82628222c 100644
@@ -493,7 +493,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	u8 mode = ES8328_DACCONTROL1_DACWL_16;
+	u8 dac_mode = ES8328_DACCONTROL1_DACWL_16;
+	u8 adc_mode = ES8328_ADCCONTROL4_ADCWL_16;
 
 	/* set master/slave audio interface */
 	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)

@@ -502,13 +503,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	/* interface format */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST;
 		break;
 	default:
 		return -EINVAL;

@@ -518,8 +522,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
 		return -EINVAL;
 
-	snd_soc_write(codec, ES8328_DACCONTROL1, mode);
-	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+	snd_soc_write(codec, ES8328_DACCONTROL1, dac_mode);
+	snd_soc_write(codec, ES8328_ADCCONTROL4, adc_mode);
 
 	/* Master serial port mode, with BCLK generated automatically */
 	snd_soc_update_bits(codec, ES8328_MASTERMODE,
sound/soc/codecs/es8328.h +15 -0
index 156c748c89c7..5a4af014e516 100644
@@ -84,7 +84,22 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL1	0x09
 #define ES8328_ADCCONTROL2	0x0a
 #define ES8328_ADCCONTROL3	0x0b
+
 #define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
+#define ES8328_ADCCONTROL4_ADCWL_24 (0 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_20 (1 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5)
+
 #define ES8328_ADCCONTROL5	0x0d
 #define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#4 lore raw

Applied "ASoC: es8328: Fix ADC format setup" to the asoc tree

The patch

ASoC: es8328: Fix ADC format setup

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From 57e41f3fb32a359753a3b2679c2502b2750bf6af Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Mon, 9 May 2016 12:24:30 +0100
Subject: [PATCH] ASoC: es8328: Fix ADC format setup

The ADCCONTROL4 and DACCONTROL1 registers are similar but not identical, with the DACCONTROL1 having each field starting one bit higher than ADCCONTROL4.

Instead of introducing a magic shift, add new constants for the values in ADCCONTROL4 and use a second variable to setup the ADC.

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 sound/soc/codecs/es8328.c | 16 ++++++++++------
 sound/soc/codecs/es8328.h | 15 +++++++++++++++
 2 files changed, 25 insertions(+), 6 deletions(-)
sound/soc/codecs/es8328.c +10 -6
index 3ca89ae32889..63e82628222c 100644
@@ -493,7 +493,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	u8 mode = ES8328_DACCONTROL1_DACWL_16;
+	u8 dac_mode = ES8328_DACCONTROL1_DACWL_16;
+	u8 adc_mode = ES8328_ADCCONTROL4_ADCWL_16;
 
 	/* set master/slave audio interface */
 	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)

@@ -502,13 +503,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	/* interface format */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST;
 		break;
 	default:
 		return -EINVAL;

@@ -518,8 +522,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
 		return -EINVAL;
 
-	snd_soc_write(codec, ES8328_DACCONTROL1, mode);
-	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+	snd_soc_write(codec, ES8328_DACCONTROL1, dac_mode);
+	snd_soc_write(codec, ES8328_ADCCONTROL4, adc_mode);
 
 	/* Master serial port mode, with BCLK generated automatically */
 	snd_soc_update_bits(codec, ES8328_MASTERMODE,
sound/soc/codecs/es8328.h +15 -0
index 156c748c89c7..5a4af014e516 100644
@@ -84,7 +84,22 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL1	0x09
 #define ES8328_ADCCONTROL2	0x0a
 #define ES8328_ADCCONTROL3	0x0b
+
 #define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
+#define ES8328_ADCCONTROL4_ADCWL_24 (0 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_20 (1 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5)
+
 #define ES8328_ADCCONTROL5	0x0d
 #define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
Signature · 1 line
2.8.1
author
#5 lore raw

PATCH 3/9 ASoC: es8328: Fix mask for VMIDSEL

In reply to John Keeping

This is always used along with ES8328_CONTROL1_ENREF so there is no change in the generated code as a result of this fix.

Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)
sound/soc/codecs/es8328.h +1 -1
index 5a4af014e516..8bc79fff0218 100644
@@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
 #define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
 #define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
-#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0)
 #define ES8328_CONTROL1_ENREF (1 << 2)
 #define ES8328_CONTROL1_SEQEN (1 << 3)
 #define ES8328_CONTROL1_SAMEFS (1 << 4)
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#6 lore raw

Applied "ASoC: es8328: Fix mask for VMIDSEL" to the asoc tree

The patch

ASoC: es8328: Fix mask for VMIDSEL

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From 2da1ab667a506cc6a7dea88b70e6df3d281458f8 Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Mon, 9 May 2016 12:24:31 +0100
Subject: [PATCH] ASoC: es8328: Fix mask for VMIDSEL

This is always used along with ES8328_CONTROL1_ENREF so there is no change in the generated code as a result of this fix.

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 sound/soc/codecs/es8328.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)
sound/soc/codecs/es8328.h +1 -1
index 5a4af014e516..8bc79fff0218 100644
@@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
 #define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
 #define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
-#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0)
 #define ES8328_CONTROL1_ENREF (1 << 2)
 #define ES8328_CONTROL1_SEQEN (1 << 3)
 #define ES8328_CONTROL1_SAMEFS (1 << 4)
Signature · 1 line
2.8.1
author
#7 lore raw

PATCH 4/9 ASoC: es8328: Use single R/W for regmap

In reply to John Keeping

The chip only supports single reads and writes.

Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.c | 1 +
 1 file changed, 1 insertion(+)
sound/soc/codecs/es8328.c +1 -0
index 63e82628222c..d580300d9220 100644
@@ -713,6 +713,7 @@ const struct regmap_config es8328_regmap_config = {
 	.val_bits	= 8,
 	.max_register	= ES8328_REG_MAX,
 	.cache_type	= REGCACHE_RBTREE,
+	.use_single_rw	= true,
 };
 EXPORT_SYMBOL_GPL(es8328_regmap_config);
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#8 lore raw

Applied "ASoC: es8328: Use single R/W for regmap" to the asoc tree

The patch

ASoC: es8328: Use single R/W for regmap

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From f2ed04a4317e5c8074d98a5c1da175596811a2d8 Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Mon, 9 May 2016 12:24:32 +0100
Subject: [PATCH] ASoC: es8328: Use single R/W for regmap

The chip only supports single reads and writes.

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 sound/soc/codecs/es8328.c | 1 +
 1 file changed, 1 insertion(+)
sound/soc/codecs/es8328.c +1 -0
index 63e82628222c..d580300d9220 100644
@@ -713,6 +713,7 @@ const struct regmap_config es8328_regmap_config = {
 	.val_bits	= 8,
 	.max_register	= ES8328_REG_MAX,
 	.cache_type	= REGCACHE_RBTREE,
+	.use_single_rw	= true,
 };
 EXPORT_SYMBOL_GPL(es8328_regmap_config);
Signature · 1 line
2.8.1
author
#9 lore raw

PATCH 5/9 ASoC: es8328: Use more suitable definition for mic bias

In reply to John Keeping
Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)
sound/soc/codecs/es8328.c +2 -2
index d580300d9220..8661d1837422 100644
@@ -287,8 +287,8 @@ static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = {
 	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ES8328_ADCPOWER,
 			ES8328_ADCPOWER_ADCL_OFF, 1),
 
-	SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER,
-			ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_MIC_BIAS_OFF, 1),
 	SND_SOC_DAPM_SUPPLY("Mic Bias Gen", ES8328_ADCPOWER,
 			ES8328_ADCPOWER_ADC_BIAS_GEN_OFF, 1, NULL, 0),
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#10 lore raw

Re: [PATCH 5/9] ASoC: es8328: Use more suitable definition for mic bias

sound/soc/codecs/es8328.c static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = { · line 291

Quoted 4 lines · John Keeping
sound/soc/codecs/es8328.c +2 -2
-	SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER,
-			ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_MIC_BIAS_OFF, 1),

No, this is broken. The specialist MICBIAS widget is legacy and should not be used.

author
#11 lore raw

Re: [PATCH 5/9] ASoC: es8328: Use more suitable definition for mic bias

Quoted 9 lines · Mark Brown

Review

Quoted 4 lines · John Keeping
(quoted diff) +2 -2
-	SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER,
-			ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_MIC_BIAS_OFF, 1),

No, this is broken. The specialist MICBIAS widget is legacy and should not be used.

OK, this can be dropped. Would you consider doing something like the patch below?

The only reference I can find to MICBIAS being legacy is in commit 086d7f804e26 ("ASoC: Convert WM8962 MICBIAS to a supply widget", 2011-09-23) and some following commits, there don't seem to be any comments in the documentation indicating that it's not to be used.

-- >8 --

include/sound/soc-dapm.h +2 -1
index 97069466c38d..3101d53468aa 100644
@@ -100,6 +100,7 @@ struct device;
 {       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+/* DEPRECATED: use SND_SOC_DAPM_SUPPLY */
 #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_micbias, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \

@@ -473,7 +474,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_out_drv,			/* output driver */
 	snd_soc_dapm_adc,			/* analog to digital converter */
 	snd_soc_dapm_dac,			/* digital to analog converter */
-	snd_soc_dapm_micbias,		/* microphone bias (power) */
+	snd_soc_dapm_micbias,		/* microphone bias (power) - DEPRECATED: use snd_soc_dapm_supply */
 	snd_soc_dapm_mic,			/* microphone */
 	snd_soc_dapm_hp,			/* headphones */
 	snd_soc_dapm_spk,			/* speaker */
maintainer
#12 lore raw

Re: [PATCH 5/9] ASoC: es8328: Use more suitable definition for mic bias

Quoted 2 lines · John Keeping

OK, this can be dropped. Would you consider doing something like the patch below?

Documentation/SubmittingPatches.

author
#13 lore raw

PATCH 1/1 ASoC: dapm: deprecate MICBIAS widget type

Commit 086d7f804e26 ("ASoC: Convert WM8962 MICBIAS to a supply widget", 2011-09-23) says:

A supply widget is generally clearer than a MICBIAS widget and a mic bias is just a type of supply so use a supply widget for the MICBIAS. This also avoids confusion with the routing when connected to multiple inputs.

but this has never been documented as a policy. Add some comments to make it clear.

Signed-off-by John Keeping <[email protected]>
---
 include/sound/soc-dapm.h | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)
include/sound/soc-dapm.h +2 -1
index 97069466c38d..3101d53468aa 100644
@@ -100,6 +100,7 @@ struct device;
 {       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+/* DEPRECATED: use SND_SOC_DAPM_SUPPLY */
 #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_micbias, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \

@@ -473,7 +474,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_out_drv,			/* output driver */
 	snd_soc_dapm_adc,			/* analog to digital converter */
 	snd_soc_dapm_dac,			/* digital to analog converter */
-	snd_soc_dapm_micbias,		/* microphone bias (power) */
+	snd_soc_dapm_micbias,		/* microphone bias (power) - DEPRECATED: use snd_soc_dapm_supply */
 	snd_soc_dapm_mic,			/* microphone */
 	snd_soc_dapm_hp,			/* headphones */
 	snd_soc_dapm_spk,			/* speaker */
Signature · 1 line
2.8.2.565.gdb84f68.dirty
maintainer
#14 lore raw

Applied "ASoC: dapm: deprecate MICBIAS widget type" to the asoc tree

In reply to John Keeping

The patch

ASoC: dapm: deprecate MICBIAS widget type

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From 32902177f7f6ae70e1d5e71d935aa1bfcae7f01c Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Thu, 12 May 2016 13:55:53 +0100
Subject: [PATCH] ASoC: dapm: deprecate MICBIAS widget type

Commit 086d7f804e26 ("ASoC: Convert WM8962 MICBIAS to a supply widget", 2011-09-23) says:

A supply widget is generally clearer than a MICBIAS widget and a mic bias is just a type of supply so use a supply widget for the MICBIAS. This also avoids confusion with the routing when connected to multiple inputs.

but this has never been documented as a policy. Add some comments to make it clear.

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 include/sound/soc-dapm.h | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)
include/sound/soc-dapm.h +2 -1
index 97069466c38d..3101d53468aa 100644
@@ -100,6 +100,7 @@ struct device;
 {       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+/* DEPRECATED: use SND_SOC_DAPM_SUPPLY */
 #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_micbias, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \

@@ -473,7 +474,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_out_drv,			/* output driver */
 	snd_soc_dapm_adc,			/* analog to digital converter */
 	snd_soc_dapm_dac,			/* digital to analog converter */
-	snd_soc_dapm_micbias,		/* microphone bias (power) */
+	snd_soc_dapm_micbias,		/* microphone bias (power) - DEPRECATED: use snd_soc_dapm_supply */
 	snd_soc_dapm_mic,			/* microphone */
 	snd_soc_dapm_hp,			/* headphones */
 	snd_soc_dapm_spk,			/* speaker */
Signature · 1 line
2.8.1
author
#15 lore raw

PATCH 6/9 ASoC: es8328: Move sample size setup to hw_params

In reply to John Keeping

This is a refactor in preparation for supporting more sample sizes which has no functional change.

Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.c | 19 ++++++++++++++-----
 sound/soc/codecs/es8328.h |  4 ++++
 2 files changed, 18 insertions(+), 5 deletions(-)
sound/soc/codecs/es8328.c +14 -5
index 8661d1837422..31c90ef9abf2 100644
@@ -482,9 +482,16 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 			ratio = mclk_ratios[i].ratio;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+				ES8328_DACCONTROL1_DACWL_MASK,
+				ES8328_DACCONTROL1_DACWL_16);
+
 		es8328->playback_fs = params_rate(params);
 		es8328_set_deemph(codec);
-	}
+	} else
+		snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+				ES8328_ADCCONTROL4_ADCWL_MASK,
+				ES8328_ADCCONTROL4_ADCWL_16);
 
 	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
 }

@@ -493,8 +500,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	u8 dac_mode = ES8328_DACCONTROL1_DACWL_16;
-	u8 adc_mode = ES8328_ADCCONTROL4_ADCWL_16;
+	u8 dac_mode = 0;
+	u8 adc_mode = 0;
 
 	/* set master/slave audio interface */
 	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)

@@ -522,8 +529,10 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
 		return -EINVAL;
 
-	snd_soc_write(codec, ES8328_DACCONTROL1, dac_mode);
-	snd_soc_write(codec, ES8328_ADCCONTROL4, adc_mode);
+	snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+			ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode);
+	snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+			ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode);
 
 	/* Master serial port mode, with BCLK generated automatically */
 	snd_soc_update_bits(codec, ES8328_MASTERMODE,
sound/soc/codecs/es8328.h +4 -0
index 8bc79fff0218..9c33d8bda859 100644
@@ -86,6 +86,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL3	0x0b
 
 #define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)

@@ -95,6 +96,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2)
 #define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2)
 #define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
 #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
 #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
 #define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)

@@ -124,6 +126,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL14	0x16
 
 #define ES8328_DACCONTROL1	0x17
+#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)

@@ -133,6 +136,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
 #define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
 #define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
 #define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#16 lore raw

Applied "ASoC: es8328: Move sample size setup to hw_params" to the asoc tree

The patch

ASoC: es8328: Move sample size setup to hw_params

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From 8865c95e43257e6676bc0f6b042ecce17eff74fe Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Mon, 9 May 2016 12:24:34 +0100
Subject: [PATCH] ASoC: es8328: Move sample size setup to hw_params

This is a refactor in preparation for supporting more sample sizes which has no functional change.

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 sound/soc/codecs/es8328.c | 19 ++++++++++++++-----
 sound/soc/codecs/es8328.h |  4 ++++
 2 files changed, 18 insertions(+), 5 deletions(-)
sound/soc/codecs/es8328.c +14 -5
index d580300d9220..c5a36e65fc40 100644
@@ -482,9 +482,16 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 			ratio = mclk_ratios[i].ratio;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+				ES8328_DACCONTROL1_DACWL_MASK,
+				ES8328_DACCONTROL1_DACWL_16);
+
 		es8328->playback_fs = params_rate(params);
 		es8328_set_deemph(codec);
-	}
+	} else
+		snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+				ES8328_ADCCONTROL4_ADCWL_MASK,
+				ES8328_ADCCONTROL4_ADCWL_16);
 
 	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
 }

@@ -493,8 +500,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	u8 dac_mode = ES8328_DACCONTROL1_DACWL_16;
-	u8 adc_mode = ES8328_ADCCONTROL4_ADCWL_16;
+	u8 dac_mode = 0;
+	u8 adc_mode = 0;
 
 	/* set master/slave audio interface */
 	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)

@@ -522,8 +529,10 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
 		return -EINVAL;
 
-	snd_soc_write(codec, ES8328_DACCONTROL1, dac_mode);
-	snd_soc_write(codec, ES8328_ADCCONTROL4, adc_mode);
+	snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+			ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode);
+	snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+			ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode);
 
 	/* Master serial port mode, with BCLK generated automatically */
 	snd_soc_update_bits(codec, ES8328_MASTERMODE,
sound/soc/codecs/es8328.h +4 -0
index 8bc79fff0218..9c33d8bda859 100644
@@ -86,6 +86,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL3	0x0b
 
 #define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)

@@ -95,6 +96,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2)
 #define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2)
 #define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
 #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
 #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
 #define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)

@@ -124,6 +126,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL14	0x16
 
 #define ES8328_DACCONTROL1	0x17
+#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)

@@ -133,6 +136,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
 #define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
 #define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
 #define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
Signature · 1 line
2.8.1
author
#17 lore raw

PATCH 7/9 ASoC: es8328: Support more sample formats

In reply to John Keeping

The values are the same for the DAC and ADC so remove the specific values and use values with shifts.

Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.c | 35 +++++++++++++++++++++++++++++------
 sound/soc/codecs/es8328.h | 12 ++----------
 2 files changed, 31 insertions(+), 16 deletions(-)
sound/soc/codecs/es8328.c +29 -6
index 31c90ef9abf2..636177641173 100644
@@ -60,7 +60,11 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
 #define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_22050 | \
 		SNDRV_PCM_RATE_11025)
-#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S18_3LE | \
+		SNDRV_PCM_FMTBIT_S20_3LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S32_LE)
 
 struct es8328_priv {
 	struct regmap *regmap;

@@ -449,6 +453,7 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 	int i;
 	int reg;
 	int val;
+	int wl;
 	u8 ratio;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

@@ -470,10 +475,28 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
 		return -EINVAL;
 	}
-	ret = snd_soc_update_bits(codec, ES8328_MASTERMODE,
+	snd_soc_update_bits(codec, ES8328_MASTERMODE,
 			ES8328_MASTERMODE_MCLKDIV2, val);
-	if (ret < 0)
-		return ret;
+
+	switch (params_width(params)) {
+	case 16:
+		wl = 3;
+		break;
+	case 18:
+		wl = 2;
+		break;
+	case 20:
+		wl = 1;
+		break;
+	case 24:
+		wl = 0;
+		break;
+	case 32:
+		wl = 4;
+		break;
+	default:
+		return -EINVAL;
+	}
 
 	/* find master mode MCLK to sampling frequency ratio */
 	ratio = mclk_ratios[0].rate;

@@ -484,14 +507,14 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		snd_soc_update_bits(codec, ES8328_DACCONTROL1,
 				ES8328_DACCONTROL1_DACWL_MASK,
-				ES8328_DACCONTROL1_DACWL_16);
+				wl << ES8328_DACCONTROL1_DACWL_SHIFT);
 
 		es8328->playback_fs = params_rate(params);
 		es8328_set_deemph(codec);
 	} else
 		snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
 				ES8328_ADCCONTROL4_ADCWL_MASK,
-				ES8328_ADCCONTROL4_ADCWL_16);
+				wl << ES8328_ADCCONTROL4_ADCWL_SHIFT);
 
 	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
 }
sound/soc/codecs/es8328.h +2 -10
index 9c33d8bda859..1a736e72a929 100644
@@ -91,11 +91,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
-#define ES8328_ADCCONTROL4_ADCWL_24 (0 << 2)
-#define ES8328_ADCCONTROL4_ADCWL_20 (1 << 2)
-#define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2)
-#define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2)
-#define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2
 #define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
 #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
 #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)

@@ -131,11 +127,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
-#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
-#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
-#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
-#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
-#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACWL_SHIFT 3
 #define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#18 lore raw

Applied "ASoC: es8328: Support more sample formats" to the asoc tree

The patch

ASoC: es8328: Support more sample formats

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From 779e86a31402c3f33f20bb02e99a5b75595bdf7f Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Mon, 9 May 2016 12:24:35 +0100
Subject: [PATCH] ASoC: es8328: Support more sample formats

The values are the same for the DAC and ADC so remove the specific values and use values with shifts.

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 sound/soc/codecs/es8328.c | 35 +++++++++++++++++++++++++++++------
 sound/soc/codecs/es8328.h | 12 ++----------
 2 files changed, 31 insertions(+), 16 deletions(-)
sound/soc/codecs/es8328.c +29 -6
index c5a36e65fc40..a66c21c7b5a0 100644
@@ -60,7 +60,11 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
 #define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_22050 | \
 		SNDRV_PCM_RATE_11025)
-#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S18_3LE | \
+		SNDRV_PCM_FMTBIT_S20_3LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S32_LE)
 
 struct es8328_priv {
 	struct regmap *regmap;

@@ -449,6 +453,7 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 	int i;
 	int reg;
 	int val;
+	int wl;
 	u8 ratio;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

@@ -470,10 +475,28 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
 		return -EINVAL;
 	}
-	ret = snd_soc_update_bits(codec, ES8328_MASTERMODE,
+	snd_soc_update_bits(codec, ES8328_MASTERMODE,
 			ES8328_MASTERMODE_MCLKDIV2, val);
-	if (ret < 0)
-		return ret;
+
+	switch (params_width(params)) {
+	case 16:
+		wl = 3;
+		break;
+	case 18:
+		wl = 2;
+		break;
+	case 20:
+		wl = 1;
+		break;
+	case 24:
+		wl = 0;
+		break;
+	case 32:
+		wl = 4;
+		break;
+	default:
+		return -EINVAL;
+	}
 
 	/* find master mode MCLK to sampling frequency ratio */
 	ratio = mclk_ratios[0].rate;

@@ -484,14 +507,14 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		snd_soc_update_bits(codec, ES8328_DACCONTROL1,
 				ES8328_DACCONTROL1_DACWL_MASK,
-				ES8328_DACCONTROL1_DACWL_16);
+				wl << ES8328_DACCONTROL1_DACWL_SHIFT);
 
 		es8328->playback_fs = params_rate(params);
 		es8328_set_deemph(codec);
 	} else
 		snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
 				ES8328_ADCCONTROL4_ADCWL_MASK,
-				ES8328_ADCCONTROL4_ADCWL_16);
+				wl << ES8328_ADCCONTROL4_ADCWL_SHIFT);
 
 	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
 }
sound/soc/codecs/es8328.h +2 -10
index 9c33d8bda859..1a736e72a929 100644
@@ -91,11 +91,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
 #define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
-#define ES8328_ADCCONTROL4_ADCWL_24 (0 << 2)
-#define ES8328_ADCCONTROL4_ADCWL_20 (1 << 2)
-#define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2)
-#define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2)
-#define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2)
+#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2
 #define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
 #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
 #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)

@@ -131,11 +127,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
-#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
-#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
-#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
-#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
-#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACWL_SHIFT 3
 #define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
Signature · 1 line
2.8.1
author
#19 lore raw

PATCH 8/9 ASoC: es8328: Support more sample rates

In reply to John Keeping
Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.c | 135 ++++++++++++++++++++++++++++++++++------------
 1 file changed, 100 insertions(+), 35 deletions(-)
sound/soc/codecs/es8328.c +100 -35
index 636177641173..6ec15c8822d3 100644
@@ -26,18 +26,30 @@
 #include <sound/tlv.h>
 #include "es8328.h"
 
-#define ES8328_SYSCLK_RATE_1X 11289600
-#define ES8328_SYSCLK_RATE_2X 22579200
+static const unsigned int rates_12288[] = {
+	8000, 12000, 16000, 24000, 32000, 48000, 96000,
+};
 
-/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
-static struct {
-	int rate;
-	u8 ratio;
-} mclk_ratios[] = {
-	{ 8000, 9 },
-	{11025, 7 },
-	{22050, 4 },
-	{44100, 2 },
+static const int ratios_12288[] = {
+	10, 7, 6, 4, 3, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
+	.count	= ARRAY_SIZE(rates_12288),
+	.list	= rates_12288,
+};
+
+static const unsigned int rates_11289[] = {
+	8018, 11025, 22050, 44100, 88200,
+};
+
+static const int ratios_11289[] = {
+	9, 7, 4, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
+	.count	= ARRAY_SIZE(rates_11289),
+	.list	= rates_11289,
 };
 
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */

@@ -57,9 +69,14 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
 	"HPVDD",
 };
 
-#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \
+		SNDRV_PCM_RATE_48000 | \
+		SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_32000 | \
 		SNDRV_PCM_RATE_22050 | \
-		SNDRV_PCM_RATE_11025)
+		SNDRV_PCM_RATE_16000 | \
+		SNDRV_PCM_RATE_11025 | \
+		SNDRV_PCM_RATE_8000)
 #define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
 		SNDRV_PCM_FMTBIT_S18_3LE | \
 		SNDRV_PCM_FMTBIT_S20_3LE | \

@@ -71,6 +88,9 @@ struct es8328_priv {
 	struct clk *clk;
 	int playback_fs;
 	bool deemph;
+	int mclkdiv2;
+	const struct snd_pcm_hw_constraint_list *sysclk_constraints;
+	const int *mclk_ratios;
 	struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
 };
 

@@ -443,40 +463,55 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute)
 			mute ? ES8328_DACCONTROL3_DACMUTE : 0);
 }
 
+static int es8328_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+	if (es8328->sysclk_constraints)
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				es8328->sysclk_constraints);
+
+	return 0;
+}
+
 static int es8328_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params,
 	struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate = clk_get_rate(es8328->clk);
 	int i;
 	int reg;
-	int val;
 	int wl;
-	u8 ratio;
+	int ratio;
+
+	if (!es8328->sysclk_constraints) {
+		dev_err(codec->dev, "No MCLK configured\n");
+		return -EINVAL;
+	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		reg = ES8328_DACCONTROL2;
 	else
 		reg = ES8328_ADCCONTROL5;
 
-	switch (clk_rate) {
-	case ES8328_SYSCLK_RATE_1X:
-		val = 0;
-		break;
-	case ES8328_SYSCLK_RATE_2X:
-		val = ES8328_MASTERMODE_MCLKDIV2;
-		break;
-	default:
-		dev_err(codec->dev,
-			"%s: clock is running at %d Hz, not %d or %d Hz\n",
-			 __func__, clk_rate,
-			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+	for (i = 0; i < es8328->sysclk_constraints->count; i++)
+		if (es8328->sysclk_constraints->list[i] == params_rate(params))
+			break;
+
+	if (i == es8328->sysclk_constraints->count) {
+		dev_err(codec->dev, "LRCLK %d unsupported with current clock\n",
+			params_rate(params));
 		return -EINVAL;
 	}
+
+	ratio = es8328->mclk_ratios[i];
 	snd_soc_update_bits(codec, ES8328_MASTERMODE,
-			ES8328_MASTERMODE_MCLKDIV2, val);
+			ES8328_MASTERMODE_MCLKDIV2,
+			es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0);
 
 	switch (params_width(params)) {
 	case 16:

@@ -498,12 +533,6 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	/* find master mode MCLK to sampling frequency ratio */
-	ratio = mclk_ratios[0].rate;
-	for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
-		if (params_rate(params) <= mclk_ratios[i].rate)
-			ratio = mclk_ratios[i].ratio;
-
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		snd_soc_update_bits(codec, ES8328_DACCONTROL1,
 				ES8328_DACCONTROL1_DACWL_MASK,

@@ -519,6 +548,40 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
 }
 
+static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int mclkdiv2 = 0;
+
+	switch (freq) {
+	case 0:
+		es8328->sysclk_constraints = NULL;
+		es8328->mclk_ratios = NULL;
+		break;
+	case 22579200:
+		mclkdiv2 = 1;
+		/* fallthru */
+	case 11289600:
+		es8328->sysclk_constraints = &constraints_11289;
+		es8328->mclk_ratios = ratios_11289;
+		break;
+	case 24576000:
+		mclkdiv2 = 1;
+		/* fallthru */
+	case 12288000:
+		es8328->sysclk_constraints = &constraints_12288;
+		es8328->mclk_ratios = ratios_12288;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	es8328->mclkdiv2 = mclkdiv2;
+	return 0;
+}
+
 static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {

@@ -616,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 }
 
 static const struct snd_soc_dai_ops es8328_dai_ops = {
+	.startup	= es8328_startup,
 	.hw_params	= es8328_hw_params,
 	.digital_mute	= es8328_mute,
+	.set_sysclk	= es8328_set_sysclk,
 	.set_fmt	= es8328_set_dai_fmt,
 };
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#20 lore raw

Applied "ASoC: es8328: Support more sample rates" to the asoc tree

The patch

ASoC: es8328: Support more sample rates

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From 45749c918129e409c44777f051dc0a5afb689459 Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Mon, 9 May 2016 12:24:36 +0100
Subject: [PATCH] ASoC: es8328: Support more sample rates

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 sound/soc/codecs/es8328.c | 135 ++++++++++++++++++++++++++++++++++------------
 1 file changed, 100 insertions(+), 35 deletions(-)
sound/soc/codecs/es8328.c +100 -35
index a66c21c7b5a0..b8ca214a5332 100644
@@ -26,18 +26,30 @@
 #include <sound/tlv.h>
 #include "es8328.h"
 
-#define ES8328_SYSCLK_RATE_1X 11289600
-#define ES8328_SYSCLK_RATE_2X 22579200
+static const unsigned int rates_12288[] = {
+	8000, 12000, 16000, 24000, 32000, 48000, 96000,
+};
 
-/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
-static struct {
-	int rate;
-	u8 ratio;
-} mclk_ratios[] = {
-	{ 8000, 9 },
-	{11025, 7 },
-	{22050, 4 },
-	{44100, 2 },
+static const int ratios_12288[] = {
+	10, 7, 6, 4, 3, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
+	.count	= ARRAY_SIZE(rates_12288),
+	.list	= rates_12288,
+};
+
+static const unsigned int rates_11289[] = {
+	8018, 11025, 22050, 44100, 88200,
+};
+
+static const int ratios_11289[] = {
+	9, 7, 4, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
+	.count	= ARRAY_SIZE(rates_11289),
+	.list	= rates_11289,
 };
 
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */

@@ -57,9 +69,14 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
 	"HPVDD",
 };
 
-#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \
+		SNDRV_PCM_RATE_48000 | \
+		SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_32000 | \
 		SNDRV_PCM_RATE_22050 | \
-		SNDRV_PCM_RATE_11025)
+		SNDRV_PCM_RATE_16000 | \
+		SNDRV_PCM_RATE_11025 | \
+		SNDRV_PCM_RATE_8000)
 #define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
 		SNDRV_PCM_FMTBIT_S18_3LE | \
 		SNDRV_PCM_FMTBIT_S20_3LE | \

@@ -71,6 +88,9 @@ struct es8328_priv {
 	struct clk *clk;
 	int playback_fs;
 	bool deemph;
+	int mclkdiv2;
+	const struct snd_pcm_hw_constraint_list *sysclk_constraints;
+	const int *mclk_ratios;
 	struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
 };
 

@@ -443,40 +463,55 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute)
 			mute ? ES8328_DACCONTROL3_DACMUTE : 0);
 }
 
+static int es8328_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+	if (es8328->sysclk_constraints)
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				es8328->sysclk_constraints);
+
+	return 0;
+}
+
 static int es8328_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params,
 	struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate = clk_get_rate(es8328->clk);
 	int i;
 	int reg;
-	int val;
 	int wl;
-	u8 ratio;
+	int ratio;
+
+	if (!es8328->sysclk_constraints) {
+		dev_err(codec->dev, "No MCLK configured\n");
+		return -EINVAL;
+	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		reg = ES8328_DACCONTROL2;
 	else
 		reg = ES8328_ADCCONTROL5;
 
-	switch (clk_rate) {
-	case ES8328_SYSCLK_RATE_1X:
-		val = 0;
-		break;
-	case ES8328_SYSCLK_RATE_2X:
-		val = ES8328_MASTERMODE_MCLKDIV2;
-		break;
-	default:
-		dev_err(codec->dev,
-			"%s: clock is running at %d Hz, not %d or %d Hz\n",
-			 __func__, clk_rate,
-			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+	for (i = 0; i < es8328->sysclk_constraints->count; i++)
+		if (es8328->sysclk_constraints->list[i] == params_rate(params))
+			break;
+
+	if (i == es8328->sysclk_constraints->count) {
+		dev_err(codec->dev, "LRCLK %d unsupported with current clock\n",
+			params_rate(params));
 		return -EINVAL;
 	}
+
+	ratio = es8328->mclk_ratios[i];
 	snd_soc_update_bits(codec, ES8328_MASTERMODE,
-			ES8328_MASTERMODE_MCLKDIV2, val);
+			ES8328_MASTERMODE_MCLKDIV2,
+			es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0);
 
 	switch (params_width(params)) {
 	case 16:

@@ -498,12 +533,6 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	/* find master mode MCLK to sampling frequency ratio */
-	ratio = mclk_ratios[0].rate;
-	for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
-		if (params_rate(params) <= mclk_ratios[i].rate)
-			ratio = mclk_ratios[i].ratio;
-
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		snd_soc_update_bits(codec, ES8328_DACCONTROL1,
 				ES8328_DACCONTROL1_DACWL_MASK,

@@ -519,6 +548,40 @@ static int es8328_hw_params(struct snd_pcm_substream *substream,
 	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
 }
 
+static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int mclkdiv2 = 0;
+
+	switch (freq) {
+	case 0:
+		es8328->sysclk_constraints = NULL;
+		es8328->mclk_ratios = NULL;
+		break;
+	case 22579200:
+		mclkdiv2 = 1;
+		/* fallthru */
+	case 11289600:
+		es8328->sysclk_constraints = &constraints_11289;
+		es8328->mclk_ratios = ratios_11289;
+		break;
+	case 24576000:
+		mclkdiv2 = 1;
+		/* fallthru */
+	case 12288000:
+		es8328->sysclk_constraints = &constraints_12288;
+		es8328->mclk_ratios = ratios_12288;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	es8328->mclkdiv2 = mclkdiv2;
+	return 0;
+}
+
 static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {

@@ -616,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 }
 
 static const struct snd_soc_dai_ops es8328_dai_ops = {
+	.startup	= es8328_startup,
 	.hw_params	= es8328_hw_params,
 	.digital_mute	= es8328_mute,
+	.set_sysclk	= es8328_set_sysclk,
 	.set_fmt	= es8328_set_dai_fmt,
 };
Signature · 1 line
2.8.1
author
#21 lore raw

PATCH 9/9 ASoC: es8328: Set symmetric rates

In reply to John Keeping

Although the ES8328 does support different rates for capture and playback, only very limited combinations are supported (8kHz and 48kHz or 8.0182kHz and 44.1kHz) with most rates required to be symmetric.

Instead of adding a lot of complexity for little gain, let's enforce symmetric rates.

Signed-off-by John Keeping <[email protected]>
---
 sound/soc/codecs/es8328.c | 1 +
 1 file changed, 1 insertion(+)
sound/soc/codecs/es8328.c +1 -0
index 6ec15c8822d3..26d46e5cb402 100644
@@ -703,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = {
 		.formats = ES8328_FORMATS,
 	},
 	.ops = &es8328_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static int es8328_suspend(struct snd_soc_codec *codec)
Signature · 1 line
2.8.0.rc4.238.g874082a
maintainer
#22 lore raw

Applied "ASoC: es8328: Set symmetric rates" to the asoc tree

The patch

ASoC: es8328: Set symmetric rates

has been applied to the asoc tree at

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks,
Mark

Quoted 1 line

From ca0d8797397c5daa6260a6c67b845d79f65140f5 Mon Sep 17 00:00:00 2001

From: John Keeping <[email protected]>
Date: Mon, 9 May 2016 12:24:37 +0100
Subject: [PATCH] ASoC: es8328: Set symmetric rates

Although the ES8328 does support different rates for capture and playback, only very limited combinations are supported (8kHz and 48kHz or 8.0182kHz and 44.1kHz) with most rates required to be symmetric.

Instead of adding a lot of complexity for little gain, let's enforce symmetric rates.

Signed-off-by John Keeping <[email protected]>
Signed-off-by Mark Brown <[email protected]>
---
 sound/soc/codecs/es8328.c | 1 +
 1 file changed, 1 insertion(+)
sound/soc/codecs/es8328.c +1 -0
index b8ca214a5332..2086d7107622 100644
@@ -703,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = {
 		.formats = ES8328_FORMATS,
 	},
 	.ops = &es8328_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static int es8328_suspend(struct snd_soc_codec *codec)
Signature · 1 line
2.8.1