<feed xmlns='http://www.w3.org/2005/Atom'>
<title>kernel/include/sound/soc-dai.h, branch linux-3.11.y</title>
<subtitle>Hosts the 0x221E linux distro kernel.</subtitle>
<id>https://universe.0xinfinity.dev/distro/kernel/atom?h=linux-3.11.y</id>
<link rel='self' href='https://universe.0xinfinity.dev/distro/kernel/atom?h=linux-3.11.y'/>
<link rel='alternate' type='text/html' href='https://universe.0xinfinity.dev/distro/kernel/'/>
<updated>2013-03-26T21:57:54Z</updated>
<entry>
<title>ASoC: snd_soc_[un]register_dai[s]() become non global function</title>
<updated>2013-03-26T21:57:54Z</updated>
<author>
<name>Kuninori Morimoto</name>
<email>kuninori.morimoto.gx@renesas.com</email>
</author>
<published>2013-03-21T10:38:30Z</published>
<link rel='alternate' type='text/html' href='https://universe.0xinfinity.dev/distro/kernel/commit/?id=f53179c026b11bef674d75154f5ea47ca3248ca9'/>
<id>urn:sha1:f53179c026b11bef674d75154f5ea47ca3248ca9</id>
<content type='text'>
All drivers are using snd_soc_register_component()
instead of snd_soc_register_dai[s]()
snd_soc_[un]register_dai[s]() are no longer needed

Signed-off-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
<entry>
<title>Merge remote-tracking branch 'asoc/topic/of' into asoc-next</title>
<updated>2013-02-11T11:06:38Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2013-02-11T11:06:38Z</published>
<link rel='alternate' type='text/html' href='https://universe.0xinfinity.dev/distro/kernel/commit/?id=556d8b55d083ad487067680feaf179664172bd4e'/>
<id>urn:sha1:556d8b55d083ad487067680feaf179664172bd4e</id>
<content type='text'>
</content>
</entry>
<entry>
<title>Merge remote-tracking branch 'asoc/topic/core' into asoc-next</title>
<updated>2013-02-11T11:06:30Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2013-02-11T11:06:30Z</published>
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<id>urn:sha1:d289323286d6b4e738458c31533da51d294d28a0</id>
<content type='text'>
</content>
</entry>
<entry>
<title>ASoC: core: Allow digital mute for capture</title>
<updated>2013-02-08T11:08:44Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2013-02-06T15:44:07Z</published>
<link rel='alternate' type='text/html' href='https://universe.0xinfinity.dev/distro/kernel/commit/?id=da18396f949ecaa45007d3aeb1b81bd6da092811'/>
<id>urn:sha1:da18396f949ecaa45007d3aeb1b81bd6da092811</id>
<content type='text'>
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.

The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.

Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.

Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Acked-by Vinod Koul &lt;vinod.koul@intel.com&gt;
Acked-by: Liam Girdwood &lt;liam.r.girdwood@linux.intel.com&gt;
</content>
</entry>
<entry>
<title>ASoC: SND_SOC_DAIFMT_GATED become 0 as default settings</title>
<updated>2013-01-30T08:42:24Z</updated>
<author>
<name>Kuninori Morimoto</name>
<email>kuninori.morimoto.gx@renesas.com</email>
</author>
<published>2013-01-30T05:03:13Z</published>
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<id>urn:sha1:eef28e10821fb671ba797a41e7cf44e3d244e32e</id>
<content type='text'>
Current soc-dai.h defines SND_SOC_DAIFMT_GATED as (2 &lt;&lt; 4),
but gated clock should be default settings (= 0).
This patch fixup SND_SOC_DAIFMT_GATED as (0 &lt;&lt; 4).

Signed-off-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
<entry>
<title>ASoC: SND_SOC_DAIFMT_NB_NF become 0 as default settings</title>
<updated>2013-01-16T11:28:24Z</updated>
<author>
<name>Kuninori Morimoto</name>
<email>kuninori.morimoto.gx@renesas.com</email>
</author>
<published>2013-01-16T04:18:23Z</published>
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<id>urn:sha1:5d163336a77af9c1b4d6d08cbc8b1279df5f579e</id>
<content type='text'>
Current soc-dai.h defines SND_SOC_DAIFMT_NB_NF as (1 &lt;&lt; 8),
but normal bit clock / normal frame should be
default settings (= 0).
This patch fixup SND_SOC_DAIFMT_NB_NF as (0 &lt;&lt; 8).

Signed-off-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
<entry>
<title>ASoC: Prevent pop_wait overwrite</title>
<updated>2012-12-15T14:45:09Z</updated>
<author>
<name>Misael Lopez Cruz</name>
<email>misael.lopez@ti.com</email>
</author>
<published>2012-12-13T18:23:05Z</published>
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<id>urn:sha1:9bffb1fb7c22c96d51d4ba06e2e023dd568a5872</id>
<content type='text'>
pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.

In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.

One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):

aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &amp;
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE

Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.

Signed-off-by: Misael Lopez Cruz &lt;misael.lopez@ti.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
<entry>
<title>ASoC: add definations for compressed operations</title>
<updated>2012-08-20T19:50:37Z</updated>
<author>
<name>Vinod Koul</name>
<email>vinod.koul@linux.intel.com</email>
</author>
<published>2012-08-16T11:40:40Z</published>
<link rel='alternate' type='text/html' href='https://universe.0xinfinity.dev/distro/kernel/commit/?id=4968107786e75f5aaba3c1c8e959ccbae929457f'/>
<id>urn:sha1:4968107786e75f5aaba3c1c8e959ccbae929457f</id>
<content type='text'>
Here we update the asoc structures to add compress stream definations
First the struct snd_soc_dai_driver adds a new member to indicate if the dai is
compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in
the struct snd_soc_dai_link. This is to be used for machine driver to perform
any opertaions required for setting up compressed audio streams

next is the compressed data operations, they are added using struct
snd_compr_ops in the struct snd_soc_platform_driver.

Signed-off-by: Namarta Kohli &lt;namartax.kohli@intel.com&gt;
Signed-off-by: Ramesh Babu K V &lt;ramesh.babu@intel.com&gt;
Signed-off-by: Vinod Koul &lt;vinod.koul@linux.intel.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
<entry>
<title>ASoC: core: Allow DAIs to specify a base address</title>
<updated>2012-05-02T14:42:27Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2012-05-01T19:03:32Z</published>
<link rel='alternate' type='text/html' href='https://universe.0xinfinity.dev/distro/kernel/commit/?id=623682941a11e6e09dcf88c4b2e3e88cf6a255a5'/>
<id>urn:sha1:623682941a11e6e09dcf88c4b2e3e88cf6a255a5</id>
<content type='text'>
Devices with many DAIs are becoming more and more common, and generally
the more modern devices have consistent register layouts between DAIs.
Rather than have drivers open code lookups based on the DAI ID or cause
uglification in UI by having register addresses for IDs provide a base
address field they can use.

Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Acked-by: Liam Girdwood &lt;lrg@ti.com&gt;
</content>
</entry>
<entry>
<title>ASoC: dpcm: Add bespoke trigger()</title>
<updated>2012-04-26T16:48:42Z</updated>
<author>
<name>Liam Girdwood</name>
<email>lrg@ti.com</email>
</author>
<published>2012-04-25T11:12:52Z</published>
<link rel='alternate' type='text/html' href='https://universe.0xinfinity.dev/distro/kernel/commit/?id=07bf84aaf736781a283b1bd36eaa911453b14574'/>
<id>urn:sha1:07bf84aaf736781a283b1bd36eaa911453b14574</id>
<content type='text'>
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.

A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.

Signed-off-by: Liam Girdwood &lt;lrg@ti.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
</feed>
