<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux/include/sound, branch v2.6.35</title>
<subtitle>Linux kernel source tree</subtitle>
<id>https://git.amat.us/linux/atom/include/sound?h=v2.6.35</id>
<link rel='self' href='https://git.amat.us/linux/atom/include/sound?h=v2.6.35'/>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/'/>
<updated>2010-05-20T16:41:44Z</updated>
<entry>
<title>Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6</title>
<updated>2010-05-20T16:41:44Z</updated>
<author>
<name>Linus Torvalds</name>
<email>torvalds@linux-foundation.org</email>
</author>
<published>2010-05-20T16:41:44Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=7f06a8b26aba1dc03b42272dc0089a800372c575'/>
<id>urn:sha1:7f06a8b26aba1dc03b42272dc0089a800372c575</id>
<content type='text'>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits)
  ALSA: hda: Storage class should be before const qualifier
  ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
  ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
  ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
  ASoC: TWL6040: Enable earphone path in codec
  ASoC: SDP4430: Add support for Earphone speaker
  ASoC: SDP4430: Add sdp4430 machine driver
  ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
  ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
  ALSA: sound/pci/asihpi: Use kzalloc
  ALSA: hdmi - dont fail on extra nodes
  ALSA: intelhdmi - add id for the CougarPoint chipset
  ALSA: intelhdmi - user friendly codec name
  ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
  ALSA: asihpi: incorrect range check
  ALSA: asihpi: testing the wrong variable
  ALSA: es1688: add pedantic range checks
  ARM: McBSP: Add support for omap4 in McBSP driver
  ARM: McBSP: Fix request for irq in OMAP4
  OMAP: McBSP: Add 32-bit mode support
  ...
</content>
</entry>
<entry>
<title>Merge branch 'topic/asoc' into for-linus</title>
<updated>2010-05-20T10:00:43Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2010-05-20T10:00:43Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=d71f4cece4bd97d05592836202fc04ff2e7817e3'/>
<id>urn:sha1:d71f4cece4bd97d05592836202fc04ff2e7817e3</id>
<content type='text'>
Conflicts:
	sound/soc/codecs/ad1938.c
</content>
</entry>
<entry>
<title>Merge branch 'topic/jack' into for-linus</title>
<updated>2010-05-20T09:59:37Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2010-05-20T09:59:37Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=20406f9b67e6fde4fff4639225c7a0e5ea6eaa9b'/>
<id>urn:sha1:20406f9b67e6fde4fff4639225c7a0e5ea6eaa9b</id>
<content type='text'>
</content>
</entry>
<entry>
<title>Merge branch 'topic/misc' into for-linus</title>
<updated>2010-05-20T09:59:29Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2010-05-20T09:59:29Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=5e8aa85253513b9c1ade8bd71dc341218a752a65'/>
<id>urn:sha1:5e8aa85253513b9c1ade8bd71dc341218a752a65</id>
<content type='text'>
</content>
</entry>
<entry>
<title>ALSA: include/sound/asound.h whitespace fixups</title>
<updated>2010-05-11T20:41:50Z</updated>
<author>
<name>Daniel Mack</name>
<email>daniel@caiaq.de</email>
</author>
<published>2010-05-11T16:57:37Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=89485d4931769d40353ea49bff1596accff8f06e'/>
<id>urn:sha1:89485d4931769d40353ea49bff1596accff8f06e</id>
<content type='text'>
This fixes some whitespace/indentation flaws I stumbled over.

Signed-off-by: Daniel Mack &lt;daniel@caiaq.de&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>ASoC: core: Fix for the volume limiting when invert is in use</title>
<updated>2010-05-11T08:34:11Z</updated>
<author>
<name>Peter Ujfalusi</name>
<email>peter.ujfalusi@nokia.com</email>
</author>
<published>2010-05-10T11:39:24Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=d11bb4a925613fa814ed4ae350440eb24ebff336'/>
<id>urn:sha1:d11bb4a925613fa814ed4ae350440eb24ebff336</id>
<content type='text'>
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.

Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.

Signed-off-by: Peter Ujfalusi &lt;peter.ujfalusi@nokia.com&gt;
Acked-by: Liam Girdwood &lt;lrg@slimlogic.co.uk&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
<entry>
<title>PM QOS update</title>
<updated>2010-05-10T21:08:19Z</updated>
<author>
<name>Mark Gross</name>
<email>mgross@linux.intel.com</email>
</author>
<published>2010-05-05T23:59:26Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=ed77134bfccf5e75b6cbadab268e559dbe6a4ebb'/>
<id>urn:sha1:ed77134bfccf5e75b6cbadab268e559dbe6a4ebb</id>
<content type='text'>
This patch changes the string based list management to a handle base
implementation to help with the hot path use of pm-qos, it also renames
much of the API to use "request" as opposed to "requirement" that was
used in the initial implementation.  I did this because request more
accurately represents what it actually does.

Also, I added a string based ABI for users wanting to use a string
interface.  So if the user writes 0xDDDDDDDD formatted hex it will be
accepted by the interface.  (someone asked me for it and I don't think
it hurts anything.)

This patch updates some documentation input I got from Randy.

Signed-off-by: markgross &lt;mgross@linux.intel.com&gt;
Signed-off-by: Rafael J. Wysocki &lt;rjw@sisk.pl&gt;
</content>
</entry>
<entry>
<title>ASoC: Allow DAI links to be kept active over suspend</title>
<updated>2010-05-10T09:37:13Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2010-05-09T12:25:43Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=3efab7dcc0f4d0087f73ff975eaa2fddc02ffc69'/>
<id>urn:sha1:3efab7dcc0f4d0087f73ff975eaa2fddc02ffc69</id>
<content type='text'>
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked.  This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link.  It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.

Tested-by: Peter Ujfalusi &lt;peter.ujfalusi@nokia.com&gt;
Acked-by: Liam Girdwood &lt;lrg@slimlogic.co.uk&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
<entry>
<title>ASoC: Support leaving paths enabled over system suspend</title>
<updated>2010-05-10T09:36:48Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2010-05-07T20:11:40Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=1547aba993c990e5a316751431328145b38e1fea'/>
<id>urn:sha1:1547aba993c990e5a316751431328145b38e1fea</id>
<content type='text'>
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.

Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.

When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.

Tested-by: Peter Ujfalusi &lt;peter.ujfalusi@nokia.com&gt;
Acked-by: Liam Girdwood &lt;lrg@slimlogic.co.uk&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
<entry>
<title>ASoC: Remove unused DAPM suspend flag</title>
<updated>2010-05-10T09:35:55Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2010-05-07T17:40:54Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=50ae8384cde9a67714ff03010493c5052690624e'/>
<id>urn:sha1:50ae8384cde9a67714ff03010493c5052690624e</id>
<content type='text'>
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.

Tested-by: Peter Ujfalusi &lt;peter.ujfalusi@nokia.com&gt;
Acked-by: Liam Girdwood &lt;lrg@slimlogic.co.uk&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
</content>
</entry>
</feed>
