<feed xmlns='http://www.w3.org/2005/Atom'>
<title>kernel.git/include/sound/soc.h, branch linux-2.6.27.y</title>
<subtitle>Hosts the 0x221E linux distro kernel.
</subtitle>
<id>https://git.0xinfinity.dev/distro/kernel.git/atom?h=linux-2.6.27.y</id>
<link rel='self' href='https://git.0xinfinity.dev/distro/kernel.git/atom?h=linux-2.6.27.y'/>
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<updated>2008-07-10T07:32:50Z</updated>
<entry>
<title>ALSA: asoc: core - add Digital Audio Interface (DAI) control functions.</title>
<updated>2008-07-10T07:32:50Z</updated>
<author>
<name>Liam Girdwood</name>
</author>
<published>2008-07-08T12:19:13Z</published>
<link rel='alternate' type='text/html' href='https://git.0xinfinity.dev/distro/kernel.git/commit/?id=8c6529dbf881303920a415c2d14a500218661949'/>
<id>urn:sha1:8c6529dbf881303920a415c2d14a500218661949</id>
<content type='text'>
This patch adds several functions for DAI control and config
and replaces the current method of calling function pointers within
the DAI struct.

Signed-off-by: Liam Girdwood &lt;lg@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
</content>
</entry>
<entry>
<title>ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.</title>
<updated>2008-07-10T07:32:32Z</updated>
<author>
<name>Liam Girdwood</name>
</author>
<published>2008-07-07T15:07:17Z</published>
<link rel='alternate' type='text/html' href='https://git.0xinfinity.dev/distro/kernel.git/commit/?id=3c4b266fe642bcaebe2b95edb56c3f8802924ff9'/>
<id>urn:sha1:3c4b266fe642bcaebe2b95edb56c3f8802924ff9</id>
<content type='text'>
This patch series merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai in preparation for further
ASoC v2 patches.

This merger removes duplication in both DAI structures and simplifies
the API for other users.

Signed-off-by: Liam Girdwood &lt;lg@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
</content>
</entry>
<entry>
<title>ALSA: ASoC: Don't block system resume</title>
<updated>2008-06-16T07:32:31Z</updated>
<author>
<name>Andy Green</name>
</author>
<published>2008-06-13T15:24:05Z</published>
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<id>urn:sha1:6ed2597883b1b03ca94f62f0cfe908314cba6d6b</id>
<content type='text'>
On OpenMoko soc-audio resume is taking 700ms of the whole resume time of
1.3s, dominated by writes to the codec over I2C.  This patch shunts the
resume guts into a workqueue which then is done asynchronously.

The "card" is locked using the ALSA power state APIs as suggested by
Mark Brown.

[Added fix for race with resume to suspend and fixed a couple of nits
from checkpatch -- broonie.]

Signed-off-by: Andy Green &lt;andy@openmoko.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
</content>
</entry>
<entry>
<title>ALSA: ASoC: Add SOC_SINGLE_EXT_TLV control type</title>
<updated>2008-06-13T14:48:13Z</updated>
<author>
<name>Mike Montour</name>
</author>
<published>2008-06-11T12:47:13Z</published>
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<id>urn:sha1:10144c09a0d6a62e1d56e25f142743c7a00e5dba</id>
<content type='text'>
Signed-off-by: Mike Montour &lt;mail@mmontour.net&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
</content>
</entry>
<entry>
<title>ALSA: ASoC: Pass the DAI being configured into CPU DAI probe and remove</title>
<updated>2008-06-13T14:48:01Z</updated>
<author>
<name>Mark Brown</name>
</author>
<published>2008-06-11T12:47:10Z</published>
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<id>urn:sha1:bdb92876f0a9d2b431199e385732ede89ff0b97d</id>
<content type='text'>
This allows per-DAI initialisation to be done by the CPU DAI drivers.

Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
</content>
</entry>
<entry>
<title>[ALSA] ASoC: Add SOC_DOUBLE_S8_TLV control type</title>
<updated>2008-05-29T06:10:49Z</updated>
<author>
<name>Mark Brown</name>
</author>
<published>2008-05-28T16:58:05Z</published>
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<id>urn:sha1:e13ac2e9b18bde51cf32c69c2209df25791ab3e5</id>
<content type='text'>
The SOC_DOUBLE_S8_TLV control type was originally implemented in the
UDA1380 driver by Philipp Zabel and was moved into the core by me.

Signed-off-by: Philipp Zabel &lt;philipp.zabel@gmail.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>[ALSA] ASoC: Clarify API for bias configuration</title>
<updated>2008-05-19T15:28:43Z</updated>
<author>
<name>Mark Brown</name>
</author>
<published>2008-05-19T10:31:28Z</published>
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<id>urn:sha1:0be9898adb6f58fee44f0fec0bbc0eac997ea9eb</id>
<content type='text'>
Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.

Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.

Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.

Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Cc: Jarkko Nikula &lt;jarkko.nikula@nokia.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
</content>
</entry>
<entry>
<title>[ALSA] ASoC: Make CPU and codec DAI operations have same type</title>
<updated>2008-05-19T11:19:20Z</updated>
<author>
<name>Mark Brown</name>
</author>
<published>2008-05-19T10:31:55Z</published>
<link rel='alternate' type='text/html' href='https://git.0xinfinity.dev/distro/kernel.git/commit/?id=1ef6ab75c7deef931d6308af282ed7d8e480e77f'/>
<id>urn:sha1:1ef6ab75c7deef931d6308af282ed7d8e480e77f</id>
<content type='text'>
The CPU and codec DAI operations differ only in the presence of the
digital mute operation for the codec so they may as well be the same
type.

Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
</content>
</entry>
<entry>
<title>[ALSA] ASoC: build fix for snd_soc_info_bool_ext</title>
<updated>2008-05-13T12:47:44Z</updated>
<author>
<name>Philipp Zabel</name>
</author>
<published>2008-05-13T12:03:40Z</published>
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<id>urn:sha1:392abe9cb36f04ccdda9c96938077b0d43f2b50b</id>
<content type='text'>
I suspect that snd_ctl_boolean_mono should have been
snd_ctl_boolean_mono_info instead. This fixes the build for magician.

Signed-off-by: Philipp Zabel &lt;philipp.zabel@gmail.com&gt;
Acked-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>[ALSA] Bump ASoC core version number</title>
<updated>2008-01-31T16:29:51Z</updated>
<author>
<name>Mark Brown</name>
</author>
<published>2008-01-10T13:53:48Z</published>
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<id>urn:sha1:0a22b87de0467bbc2ce863d54b55c3a7b649ba74</id>
<content type='text'>
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
</content>
</entry>
</feed>
