[PATCH] ALSA: ASoC: support more sample rates on raumfeld devices

[PATCH] ALSA: ASoC: support more sample rates on raumfeld devices

[Daniel Mack]

Add support for sample rates other than 44100Khz on raumfeld audio
devices. Force the codec's supported sample rates as the table of
supported rates which is calculated at run-time does not suffice: if we
call snd_soc_dai_set_sysclk() at stream startup, half of the possible
rates are missing.

Use the external audio clock generator to provide double data rate
clocks as the PXA's internal baud generator does anything but what's
described in the datasheets.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Timur Tabi <timur@freescale.com>

Signed-off-by: Daniel Mack <daniel@caiaq.de>
---
 sound/soc/pxa/raumfeld.c |   61 ++++++++++++++++++++++++++++++----------------
 1 files changed, 40 insertions(+), 21 deletions(-)

diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index e713181..8d65a49 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -41,7 +41,9 @@ static struct i2c_board_info max9486_hwmon_info = {
 };
 
 #define MAX9485_MCLK_FREQ_112896 0x22
-#define	MAX9485_MCLK_FREQ_122880 0x23
+#define MAX9485_MCLK_FREQ_122880 0x23
+#define MAX9485_MCLK_FREQ_225792 0x32
+#define MAX9485_MCLK_FREQ_245760 0x33
 
 static void set_max9485_clk(char clk)
 {
@@ -71,9 +73,18 @@ static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
-	set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+	codec_dai->playback.rates = SNDRV_PCM_RATE_44100 |
+				    SNDRV_PCM_RATE_48000 |
+				    SNDRV_PCM_RATE_88200 |
+				    SNDRV_PCM_RATE_96000;
+	codec_dai->playback.rate_min = 44100;
+	codec_dai->playback.rate_max = 48000;
 
-	return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0);
+	codec_dai->capture.rates = codec_dai->playback.rates;
+	codec_dai->capture.rate_min = codec_dai->playback.rate_min;
+	codec_dai->capture.rate_max = codec_dai->playback.rate_max;
+
+	return 0;
 }
 
 static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
@@ -86,20 +97,24 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 
 	switch (params_rate(params)) {
-	case 8000:
-	case 16000:
+	case 44100:
+		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+		clk = 11289600;
+		break;
 	case 48000:
-	case 96000:
 		set_max9485_clk(MAX9485_MCLK_FREQ_122880);
 		clk = 12288000;
 		break;
-	case 11025:
-	case 22050:
-	case 44100:
 	case 88200:
-		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
-		clk = 11289600;
+		set_max9485_clk(MAX9485_MCLK_FREQ_225792);
+		clk = 22579200;
+		break;
+	case 96000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_245760);
+		clk = 24576000;
 		break;
+	default:
+		return -EINVAL;
 	}
 
 	fmt = SND_SOC_DAIFMT_I2S |
@@ -128,7 +143,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1);
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
 	if (ret < 0)
 		return ret;
 
@@ -181,20 +196,24 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream,
 	int fmt, ret = 0, clk = 0;
 
 	switch (params_rate(params)) {
-	case 8000:
-	case 16000:
+	case 44100:
+		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+		clk = 11289600;
+		break;
 	case 48000:
-	case 96000:
 		set_max9485_clk(MAX9485_MCLK_FREQ_122880);
 		clk = 12288000;
 		break;
-	case 11025:
-	case 22050:
-	case 44100:
 	case 88200:
-		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
-		clk = 11289600;
+		set_max9485_clk(MAX9485_MCLK_FREQ_225792);
+		clk = 22579200;
+		break;
+	case 96000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_245760);
+		clk = 24576000;
 		break;
+	default:
+		return -EINVAL;
 	}
 
 	fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF;
@@ -217,7 +236,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1);
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
 	if (ret < 0)
 		return ret;
 
-- 
1.6.5.2

Re: [PATCH] ALSA: ASoC: support more sample rates on raumfeld devices

[Mark Brown]

On Wed, Jan 13, 2010 at 11:25:38AM +0100, Daniel Mack wrote:
> Add support for sample rates other than 44100Khz on raumfeld audio
> devices. Force the codec's supported sample rates as the table of
> supported rates which is calculated at run-time does not suffice: if we
> call snd_soc_dai_set_sysclk() at stream startup, half of the possible
> rates are missing.

I'm not massively enthusiastic about this - I'd much rather be able to
give the CODEC a hook to let it know what MCLK rates it can have and a
callback to update them.  It's a perfect case for the clock API but of
course we can't use that off-SoC :/

Re: [PATCH] ALSA: ASoC: support more sample rates on raumfeld devices

[Timur Tabi]

Mark Brown wrote:
> On Wed, Jan 13, 2010 at 11:25:38AM +0100, Daniel Mack wrote:
>> Add support for sample rates other than 44100Khz on raumfeld audio
>> devices. Force the codec's supported sample rates as the table of
>> supported rates which is calculated at run-time does not suffice: if we
>> call snd_soc_dai_set_sysclk() at stream startup, half of the possible
>> rates are missing.
> 
> I'm not massively enthusiastic about this - I'd much rather be able to
> give the CODEC a hook to let it know what MCLK rates it can have and a
> callback to update them.  It's a perfect case for the clock API but of
> course we can't use that off-SoC :/

This is how I was planning on doing it for the P1022DS.

-- 
Timur Tabi
Linux kernel developer at Freescale

Re: [PATCH] ALSA: ASoC: support more sample rates on raumfeld devices

[Daniel Mack]

On Wed, Jan 13, 2010 at 01:02:09PM +0000, Mark Brown wrote:
> On Wed, Jan 13, 2010 at 11:25:38AM +0100, Daniel Mack wrote:
> > Add support for sample rates other than 44100Khz on raumfeld audio
> > devices. Force the codec's supported sample rates as the table of
> > supported rates which is calculated at run-time does not suffice: if we
> > call snd_soc_dai_set_sysclk() at stream startup, half of the possible
> > rates are missing.
> 
> I'm not massively enthusiastic about this - 

Me neither, and I agree to your objections.

> I'd much rather be able to
> give the CODEC a hook to let it know what MCLK rates it can have and a
> callback to update them.  It's a perfect case for the clock API but of
> course we can't use that off-SoC :/

I thought about this too and disliked it as well. It would make the
codec be in charge of dealing with clock changes, which is - in this
case particularily at least - not apropriate as the codec is slave to
all clocks.

For this application, the codec driver would need to be as dumb as
possible: I would just get the information about which MCLK and LRCLK it
runs with and it could reject invalid configurations by return value.
But that would imply major changes to the clocking logic in the cs4270
driver, which is fine as-is when it acts as master.

I'm uncertain and hence I just went the other way which bypasses this
logic for now, also not to break existing platforms.

Daniel

Re: [PATCH] ALSA: ASoC: support more sample rates on raumfeld devices

[Mark Brown]

On Wed, Jan 13, 2010 at 04:43:20PM +0100, Daniel Mack wrote:

> For this application, the codec driver would need to be as dumb as
> possible: I would just get the information about which MCLK and LRCLK it
> runs with and it could reject invalid configurations by return value.
> But that would imply major changes to the clocking logic in the cs4270
> driver, which is fine as-is when it acts as master.

That's not quite true.  The problem you're seeing is that the CODEC
driver has variable constraints depending on the master clock, so it at
least needs to know what master clocks it might get so it can specify
the constraints.  Even more ideally than what I suggested we'd be able
to use a clock API with constraint handling which we could use to figure
stuff out but obviously tackling that is very much an uphill struggle.

> I'm uncertain and hence I just went the other way which bypasses this
> logic for now, also not to break existing platforms.

It doesn't do anything for maintainability of the stack, though,
especially if other people try to copy it.

If you're going to do anything that doesn't involve the core API I'd
have the CODEC driver manually apply the constraints from MCLK in the
startup function iff an MCLK is currently set.  Machines with fixed
MCLKs can set the MCLK on init() and machines with dynamic MCLKs can set
the value in hw_params() and reset it when shutting down, imposing any
constraints that do exist themselves.  The rates fixed in the DAI would
be all the rates that can possibly be used.

[PATCH 1/2] ALSA: ASoC: cs4270: allow passing freq=0 in set_dai_sysclk()

[Daniel Mack]

For setups with variable MCLKs, the current logic of limiting the
available sampling rates at startup time is not sufficient. We need to
be able to change the setting at a later point, and so the codec must
offer all possible rates until the hw_params are given.

This patches allows that by passing 0 as 'freq' argument to
cs4270_set_dai_sysclk().

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Timur Tabi <timur@freescale.com>
---
 sound/soc/codecs/cs4270.c |   40 ++++++++++++++++++++++++++--------------
 1 files changed, 26 insertions(+), 14 deletions(-)

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index ffe122d..37ed60d 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -192,6 +192,11 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = {
  * This function must be called by the machine driver's 'startup' function,
  * otherwise the list of supported sample rates will not be available in
  * time for ALSA.
+ *
+ * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause
+ * theoretically possible sample rates to be enabled. Call it again with a
+ * proper value set one the external clock is set (most probably you would do
+ * that from a machine's driver 'hw_param' hook.
  */
 static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
@@ -205,20 +210,27 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 
 	cs4270->mclk = freq;
 
-	for (i = 0; i < NUM_MCLK_RATIOS; i++) {
-		unsigned int rate = freq / cs4270_mode_ratios[i].ratio;
-		rates |= snd_pcm_rate_to_rate_bit(rate);
-		if (rate < rate_min)
-			rate_min = rate;
-		if (rate > rate_max)
-			rate_max = rate;
-	}
-	/* FIXME: soc should support a rate list */
-	rates &= ~SNDRV_PCM_RATE_KNOT;
+	if (cs4270->mclk) {
+		for (i = 0; i < NUM_MCLK_RATIOS; i++) {
+			unsigned int rate = freq / cs4270_mode_ratios[i].ratio;
+			rates |= snd_pcm_rate_to_rate_bit(rate);
+			if (rate < rate_min)
+				rate_min = rate;
+			if (rate > rate_max)
+				rate_max = rate;
+		}
+		/* FIXME: soc should support a rate list */
+		rates &= ~SNDRV_PCM_RATE_KNOT;
 
-	if (!rates) {
-		dev_err(codec->dev, "could not find a valid sample rate\n");
-		return -EINVAL;
+		if (!rates) {
+			dev_err(codec->dev, "could not find a valid sample rate\n");
+			return -EINVAL;
+		}
+	} else {
+		/* enable all possible rates */
+		rates = SNDRV_PCM_RATE_8000_192000;
+		rate_min = 8000;
+		rate_max = 192000;
 	}
 
 	codec_dai->playback.rates = rates;
-- 
1.6.5.2

[PATCH 2/2] ALSA: ASoC: support more sample rates on raumfeld devices

[Daniel Mack]

Add support for sample rates other than 44100Khz on raumfeld audio
devices. At startup time, call snd_soc_dai_set_sysclk() with 0 as 'freq'
argument so it offers all the sample rates. Later, the function is
called again to give proper constraints.

Use the external audio clock generator to provide double data rate
clocks as the PXA's internal baud generator does anything but what's
described in the datasheets.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Timur Tabi <timur@freescale.com>
---
 sound/soc/pxa/raumfeld.c |   61 ++++++++++++++++++++++++++++++----------------
 1 files changed, 40 insertions(+), 21 deletions(-)

diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index e713181..66e0dd8 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -41,7 +41,9 @@ static struct i2c_board_info max9486_hwmon_info = {
 };
 
 #define MAX9485_MCLK_FREQ_112896 0x22
-#define	MAX9485_MCLK_FREQ_122880 0x23
+#define MAX9485_MCLK_FREQ_122880 0x23
+#define MAX9485_MCLK_FREQ_225792 0x32
+#define MAX9485_MCLK_FREQ_245760 0x33
 
 static void set_max9485_clk(char clk)
 {
@@ -71,9 +73,17 @@ static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
-	set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+	/* set freq to 0 to enable all possible codec sample rates */
+	return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+}
 
-	return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0);
+static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+	/* set freq to 0 to enable all possible codec sample rates */
+	snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
 }
 
 static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
@@ -86,20 +96,24 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 
 	switch (params_rate(params)) {
-	case 8000:
-	case 16000:
+	case 44100:
+		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+		clk = 11289600;
+		break;
 	case 48000:
-	case 96000:
 		set_max9485_clk(MAX9485_MCLK_FREQ_122880);
 		clk = 12288000;
 		break;
-	case 11025:
-	case 22050:
-	case 44100:
 	case 88200:
-		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
-		clk = 11289600;
+		set_max9485_clk(MAX9485_MCLK_FREQ_225792);
+		clk = 22579200;
 		break;
+	case 96000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_245760);
+		clk = 24576000;
+		break;
+	default:
+		return -EINVAL;
 	}
 
 	fmt = SND_SOC_DAIFMT_I2S |
@@ -128,7 +142,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1);
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
 	if (ret < 0)
 		return ret;
 
@@ -137,6 +151,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
 
 static struct snd_soc_ops raumfeld_cs4270_ops = {
 	.startup = raumfeld_cs4270_startup,
+	.shutdown = raumfeld_cs4270_shutdown,
 	.hw_params = raumfeld_cs4270_hw_params,
 };
 
@@ -181,20 +196,24 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream,
 	int fmt, ret = 0, clk = 0;
 
 	switch (params_rate(params)) {
-	case 8000:
-	case 16000:
+	case 44100:
+		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+		clk = 11289600;
+		break;
 	case 48000:
-	case 96000:
 		set_max9485_clk(MAX9485_MCLK_FREQ_122880);
 		clk = 12288000;
 		break;
-	case 11025:
-	case 22050:
-	case 44100:
 	case 88200:
-		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
-		clk = 11289600;
+		set_max9485_clk(MAX9485_MCLK_FREQ_225792);
+		clk = 22579200;
+		break;
+	case 96000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_245760);
+		clk = 24576000;
 		break;
+	default:
+		return -EINVAL;
 	}
 
 	fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF;
@@ -217,7 +236,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1);
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
 	if (ret < 0)
 		return ret;
 
-- 
1.6.5.2

Re: [PATCH 1/2] ALSA: ASoC: cs4270: allow passing freq=0 in set_dai_sysclk()

[Timur Tabi]

Daniel Mack wrote:
> For setups with variable MCLKs, the current logic of limiting the
> available sampling rates at startup time is not sufficient. We need to
> be able to change the setting at a later point, and so the codec must
> offer all possible rates until the hw_params are given.
> 
> This patches allows that by passing 0 as 'freq' argument to
> cs4270_set_dai_sysclk().
> 
> Signed-off-by: Daniel Mack <daniel@caiaq.de>
> Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
> Cc: Timur Tabi <timur@freescale.com>
> ---

Acked-by: Timur Tabi <timur@freescale.com>

-- 
Timur Tabi
Linux kernel developer at Freescale

Re: [PATCH 1/2] ALSA: ASoC: cs4270: allow passing freq=0 in set_dai_sysclk()

[Mark Brown]

On Fri, Jan 15, 2010 at 11:08:56AM -0600, Timur Tabi wrote:
> Daniel Mack wrote:

> > For setups with variable MCLKs, the current logic of limiting the
> > available sampling rates at startup time is not sufficient. We need to
> > be able to change the setting at a later point, and so the codec must
> > offer all possible rates until the hw_params are given.

> > Signed-off-by: Daniel Mack <daniel@caiaq.de>

> Acked-by: Timur Tabi <timur@freescale.com>

Applied both patches, thanks.