[PATCH] ASoC: define playback and capture streams in dummy codec

[PATCH] ASoC: define playback and capture streams in dummy codec

[Stas Sergeev]

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Hi.

This patch adds a playback and capture streams to the dummy codec DAI
configuration. Most permissive set of sampling rates and formats is used.

Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>

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Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Mark Brown]

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On Fri, Apr 12, 2013 at 02:53:56PM +0400, Stas Sergeev wrote:
> Hi.
> 
> This patch adds a playback and capture streams to the dummy codec DAI
> configuration. Most permissive set of sampling rates and formats is used.

The changelog ought to explain why this is required...

> Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>

Please send patches as documented in SubmittingPatches - even as an
attachment this isn't coming out as plain text for me.

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Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Stas Sergeev]

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On Fri, 12 Apr 2013 11:59:44 +0100
Mark Brown <broonie@opensource.wolfsonmicro.com> wrote:
> > This patch adds a playback and capture streams to the dummy codec DAI
> > configuration. Most permissive set of sampling rates and formats is used.
> The changelog ought to explain why this is required...
I can only tell that without this patch I am not getting any
sound at all... which is probably a rather poor statement for
the changelog. Could you please correct the changelog with the
better knowledge of the details?

> > Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>
> Please send patches as documented in SubmittingPatches - even as an
> attachment this isn't coming out as plain text for me.
OK, now it should be as a plain text.

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>From 22813b2124d81d8de5cde19a4e2f9490bd23baab Mon Sep 17 00:00:00 2001
From: Stas Sergeev <stsp@users.sourceforge.net>
Date: Thu, 11 Apr 2013 19:07:20 +0400
Subject: [PATCH] ASoC: define playback and capture streams in dummy codec

This patch adds a playback and capture streams to the dummy codec DAI
configuration. Most permissive set of sampling rates and formats is used.

Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>
---
 sound/soc/soc-utils.c | 25 +++++++++++++++++++++++++
 1 file changed, 25 insertions(+)

diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index fe4541d..bf1f92a 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -90,8 +90,33 @@ static struct snd_soc_platform_driver dummy_platform = {
 };
 
 static struct snd_soc_codec_driver dummy_codec;
+
+#define STUB_RATES	SNDRV_PCM_RATE_8000_192000
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
+			SNDRV_PCM_FMTBIT_U8 | \
+			SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_U16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
+			SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | \
+			SNDRV_PCM_FMTBIT_U32_LE | \
+			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
 static struct snd_soc_dai_driver dummy_dai = {
 	.name = "snd-soc-dummy-dai",
+	.playback 	= {
+		.stream_name	= "Playback",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates		= STUB_RATES,
+		.formats	= STUB_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates = STUB_RATES,
+		.formats = STUB_FORMATS,
+	 },
 };
 
 static int snd_soc_dummy_probe(struct platform_device *pdev)
-- 
1.7.11.7


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Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Gabriel M. Beddingfield]

On 04/12/2013 07:19 AM, Stas Sergeev wrote:
> On Fri, 12 Apr 2013 11:59:44 +0100
> Mark Brown <broonie@opensource.wolfsonmicro.com> wrote:
>>> This patch adds a playback and capture streams to the dummy codec DAI
>>> configuration. Most permissive set of sampling rates and formats is used.
>> The changelog ought to explain why this is required...
> I can only tell that without this patch I am not getting any
> sound at all... which is probably a rather poor statement for
> the changelog. Could you please correct the changelog with the
> better knowledge of the details?

He means that the commit message.  Your commit message says what the 
patch does.  It does not say why the patch is needed.

-gabriel

Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Stas Sergeev]

12.04.2013 19:13, Gabriel M. Beddingfield пишет:
> On 04/12/2013 07:19 AM, Stas Sergeev wrote:
>> On Fri, 12 Apr 2013 11:59:44 +0100
>> Mark Brown <broonie@opensource.wolfsonmicro.com> wrote:
>>>> This patch adds a playback and capture streams to the dummy codec DAI
>>>> configuration. Most permissive set of sampling rates and formats is
>>>> used.
>>> The changelog ought to explain why this is required...
>> I can only tell that without this patch I am not getting any
>> sound at all... which is probably a rather poor statement for
>> the changelog. Could you please correct the changelog with the
>> better knowledge of the details?
>
> He means that the commit message. Your commit message says what the
> patch does. It does not say why the patch is needed.
Yes.
But what should I add to the commit message, if the
only thing I know about this patch, is that I am not getting
any sound without it? :) The reason should be obvious for
whoever is familiar with that subsystem, but its not me.
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Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Gabriel M. Beddingfield]

On 04/12/2013 08:20 AM, Stas Sergeev wrote:
>> He means that the commit message. Your commit message says what the
>> patch does. It does not say why the patch is needed.
> Yes.
> But what should I add to the commit message, if the
> only thing I know about this patch, is that I am not getting
> any sound without it? :) The reason should be obvious for
> whoever is familiar with that subsystem, but its not me.
>

Yep, just formalize the "I am not getting any sound with it" statement. 
:-)  Maybe something like: "Because snd-soc-dummy didn't declare any 
DAI, it was unable to be used for any audio streams."

-gabriel

Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Stas Sergeev]

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12.04.2013 19:24, Gabriel M. Beddingfield пишет:
> On 04/12/2013 08:20 AM, Stas Sergeev wrote:
>>> He means that the commit message. Your commit message says what the
>>> patch does. It does not say why the patch is needed.
>> Yes.
>> But what should I add to the commit message, if the
>> only thing I know about this patch, is that I am not getting
>> any sound without it? :) The reason should be obvious for
>> whoever is familiar with that subsystem, but its not me.
>>
>
> Yep, just formalize the "I am not getting any sound with it"
> statement. :-)
Hi, Would something like the attached patch be good to
help formalizing this? With that patch I am getting:
---
[    2.800190] asoc: no playback or capture streams
available                  
[    2.805158] asoc: can't create pcm
Pri_Dai                                  
[    2.809241] asoc: failed to instantiate card I2S: -22
---
which is much better than before.
If I get any ACKs on this one, I'll submit it too.

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diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 394fb3b..fb3a401 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2140,6 +2140,11 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	if (codec_dai->driver->capture.channels_min)
 		capture = 1;
 
+	if (playback + capture == 0) {
+		pr_err("asoc: no playback or capture streams available\n");
+		return -EINVAL;
+	}
+
 	dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
 	ret = snd_pcm_new(rtd->card->snd_card, new_name,
 			num, playback, capture, &pcm);

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Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Lars-Peter Clausen]

On 04/12/2013 05:20 PM, Stas Sergeev wrote:
> 12.04.2013 19:13, Gabriel M. Beddingfield пишет:
>> On 04/12/2013 07:19 AM, Stas Sergeev wrote:
>>> On Fri, 12 Apr 2013 11:59:44 +0100
>>> Mark Brown <broonie@opensource.wolfsonmicro.com> wrote:
>>>>> This patch adds a playback and capture streams to the dummy codec DAI
>>>>> configuration. Most permissive set of sampling rates and formats is
>>>>> used.
>>>> The changelog ought to explain why this is required...
>>> I can only tell that without this patch I am not getting any
>>> sound at all... which is probably a rather poor statement for
>>> the changelog. Could you please correct the changelog with the
>>> better knowledge of the details?
>>
>> He means that the commit message. Your commit message says what the
>> patch does. It does not say why the patch is needed.
> Yes.
> But what should I add to the commit message, if the
> only thing I know about this patch, is that I am not getting
> any sound without it? :) The reason should be obvious for
> whoever is familiar with that subsystem, but its not me.

Well, if not explicitly initialized a field is set to 0. Which is kind of the
most restrictive option for many of the fields. E.g. channels_max, rates,
formats, etc. When the ASoC core creates a new PCM device it will take the
intersection of the CPU DAI and CODEC DAI fields to initialize the fields the
PCM. So if for example channels_max is 0 for the CODEC DAI, channels_max will
also be 0 for the PCM, no matter what channels_max is set to for the CPU DAI.
Same goes for formats and rates. So a dummy CODEC should have set its fields in
a way that it is most permissive, so that the intersection of the CODEC DAI
fields with the CPU DAI fields will be equal to the CPU DAI fields.

- Lars
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Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Stas Sergeev]

On Fri, 12 Apr 2013 17:31:19 +0200
Lars-Peter Clausen <lars@metafoo.de> wrote:
> Well, if not explicitly initialized a field is set to 0. Which is kind of the
> most restrictive option for many of the fields. E.g. channels_max, rates,
> formats, etc. When the ASoC core creates a new PCM device it will take the
> intersection of the CPU DAI and CODEC DAI fields to initialize the fields the
> PCM. So if for example channels_max is 0 for the CODEC DAI, channels_max will
> also be 0 for the PCM, no matter what channels_max is set to for the CPU DAI.
> Same goes for formats and rates. So a dummy CODEC should have set its fields in
> a way that it is most permissive, so that the intersection of the CODEC DAI
> fields with the CPU DAI fields will be equal to the CPU DAI fields.
Thanks for an explanation!
But what happens when the DAI is considered mute
by the effect of such intersection?
>From what I can see, no callbacks from the DAI driver
are called in this case (this is expected), but no
error is returned to userspace, and, more importantly,
the playback speed is still correct, so the userspace
can get a playback position as if the playback is fine.
While during the normal playback, if I stop calling
snd_pcm_period_elapsed(), userspace is no longer getting
the right position.
That's why I am confused, I can't easily explain this
"no sound but otherwise fine playback" effect, so I am
a bit reluctant to try explaining this in a commit msg.
Are there are some fallbacks? Such as when the DAI is
considered mute, it gets somehow emulated, with the use
of the system clock for correct timing etc?

Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Lars-Peter Clausen]

On 04/12/2013 06:37 PM, Stas Sergeev wrote:
> On Fri, 12 Apr 2013 17:31:19 +0200
> Lars-Peter Clausen <lars@metafoo.de> wrote:
>> Well, if not explicitly initialized a field is set to 0. Which is kind of the
>> most restrictive option for many of the fields. E.g. channels_max, rates,
>> formats, etc. When the ASoC core creates a new PCM device it will take the
>> intersection of the CPU DAI and CODEC DAI fields to initialize the fields the
>> PCM. So if for example channels_max is 0 for the CODEC DAI, channels_max will
>> also be 0 for the PCM, no matter what channels_max is set to for the CPU DAI.
>> Same goes for formats and rates. So a dummy CODEC should have set its fields in
>> a way that it is most permissive, so that the intersection of the CODEC DAI
>> fields with the CPU DAI fields will be equal to the CPU DAI fields.
> Thanks for an explanation!
> But what happens when the DAI is considered mute
> by the effect of such intersection?
> From what I can see, no callbacks from the DAI driver
> are called in this case (this is expected), but no
> error is returned to userspace, and, more importantly,
> the playback speed is still correct, so the userspace
> can get a playback position as if the playback is fine.
> While during the normal playback, if I stop calling
> snd_pcm_period_elapsed(), userspace is no longer getting
> the right position.
> That's why I am confused, I can't easily explain this
> "no sound but otherwise fine playback" effect, so I am
> a bit reluctant to try explaining this in a commit msg.
> Are there are some fallbacks? Such as when the DAI is
> considered mute, it gets somehow emulated, with the use
> of the system clock for correct timing etc?

That's actually a bit strange indeed, since you should get an error when you
try to open the pcm device.

- Lars

Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Stas Sergeev]

12.04.2013 20:55, Lars-Peter Clausen пишет:
> On 04/12/2013 06:37 PM, Stas Sergeev wrote:
>> Thanks for an explanation!
>> But what happens when the DAI is considered mute
>> by the effect of such intersection?
>>  From what I can see, no callbacks from the DAI driver
>> are called in this case (this is expected), but no
>> error is returned to userspace, and, more importantly,
>> the playback speed is still correct, so the userspace
>> can get a playback position as if the playback is fine.
>> While during the normal playback, if I stop calling
>> snd_pcm_period_elapsed(), userspace is no longer getting
>> the right position.
>> That's why I am confused, I can't easily explain this
>> "no sound but otherwise fine playback" effect, so I am
>> a bit reluctant to try explaining this in a commit msg.
>> Are there are some fallbacks? Such as when the DAI is
>> considered mute, it gets somehow emulated, with the use
>> of the system clock for correct timing etc?
> That's actually a bit strange indeed, since you should get an error when you
> try to open the pcm device.
Ah great!
Next week I'll double-check that the fall-back is actually
in userspace (in an android media stack perhaps), and
that will be enough info for the reasonable commit msg. :)
Thanks for your help!
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[PATCH] ASoC: define playback and capture streams in dummy codec

[Stas Sergeev]

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Hello.

This patch adds a playback and capture streams to the dummy codec DAI
configuration. Most permissive set of sampling rates and formats is used.

This patch is needed for playback and capturing on a codec-less systems,
as otherwise the PCM device nodes are not even created.

Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>

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>From 9a498275e59c2c6ef933fe7d0cff79a13ba790ff Mon Sep 17 00:00:00 2001
From: Stas Sergeev <stsp@users.sourceforge.net>
Date: Thu, 11 Apr 2013 19:07:20 +0400
Subject: [PATCH] ASoC: define playback and capture streams in dummy codec

This patch adds a playback and capture streams to the dummy codec DAI
configuration. Most permissive set of sampling rates and formats is used.

This patch is needed for playback and capturing on a codec-less systems,
as otherwise the PCM device nodes are not even created.

Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>
---
 sound/soc/soc-utils.c | 25 +++++++++++++++++++++++++
 1 file changed, 25 insertions(+)

diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index fe4541d..4b3be6c 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -90,8 +90,33 @@ static struct snd_soc_platform_driver dummy_platform = {
 };
 
 static struct snd_soc_codec_driver dummy_codec;
+
+#define STUB_RATES	SNDRV_PCM_RATE_8000_192000
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
+			SNDRV_PCM_FMTBIT_U8 | \
+			SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_U16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
+			SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | \
+			SNDRV_PCM_FMTBIT_U32_LE | \
+			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
 static struct snd_soc_dai_driver dummy_dai = {
 	.name = "snd-soc-dummy-dai",
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates		= STUB_RATES,
+		.formats	= STUB_FORMATS,
+	},
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates = STUB_RATES,
+		.formats = STUB_FORMATS,
+	 },
 };
 
 static int snd_soc_dummy_probe(struct platform_device *pdev)
-- 
1.7.11.7


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Re: [PATCH] ASoC: define playback and capture streams in dummy codec

[Mark Brown]

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On Wed, Apr 17, 2013 at 04:34:06PM +0400, Stas Sergeev wrote:

> This patch adds a playback and capture streams to the dummy codec DAI
> configuration. Most permissive set of sampling rates and formats is used.
> 
> This patch is needed for playback and capturing on a codec-less systems,
> as otherwise the PCM device nodes are not even created.
> 
> Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>

> From 9a498275e59c2c6ef933fe7d0cff79a13ba790ff Mon Sep 17 00:00:00 2001
> From: Stas Sergeev <stsp@users.sourceforge.net>
> Date: Thu, 11 Apr 2013 19:07:20 +0400
> Subject: [PATCH] ASoC: define playback and capture streams in dummy codec

Applied, but please do send patches as covered in SubmittingPatches -
send patches inline, not as attachments.  That makes them harder to
apply.

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