<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux/sound, branch v3.3.2</title>
<subtitle>Linux kernel source tree</subtitle>
<id>https://git.amat.us/linux/atom/sound?h=v3.3.2</id>
<link rel='self' href='https://git.amat.us/linux/atom/sound?h=v3.3.2'/>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/'/>
<updated>2012-04-13T16:13:57Z</updated>
<entry>
<title>ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS</title>
<updated>2012-04-13T16:13:57Z</updated>
<author>
<name>Stephen Warren</name>
<email>swarren@nvidia.com</email>
</author>
<published>2012-04-06T05:11:16Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=4b06fc96136303af61cd835bb1504f50cb23a615'/>
<id>urn:sha1:4b06fc96136303af61cd835bb1504f50cb23a615</id>
<content type='text'>
commit 8abe05c6eb358967f16bce8a02c88d57c82cfbd6 upstream.

Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev-&gt;id" changed
the prototype of tegra_i2s_debug_add, but didn't update the dummy inline
used when !CONFIG_DEBUG_FS. Fix that.

Signed-off-by: Stephen Warren &lt;swarren@nvidia.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ASoC: ak4642: fixup: mute needs +1 step</title>
<updated>2012-04-13T16:13:57Z</updated>
<author>
<name>Kuninori Morimoto</name>
<email>kuninori.morimoto.gx@renesas.com</email>
</author>
<published>2012-04-05T06:28:01Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=3ea212aa78fda0c2892595da96adc315da361225'/>
<id>urn:sha1:3ea212aa78fda0c2892595da96adc315da361225</id>
<content type='text'>
commit 1f99e44cf059d2ed43c5a0724fa738b83800f725 upstream.

ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute.
But current settings didn't care +1 step for mute.
This patch adds it

Signed-off-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ASoC: wm8994: Update WM8994 DCS calibration</title>
<updated>2012-04-13T16:13:50Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2012-03-21T13:22:40Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=11b604c87d574d65c18b7da541b204d8b7f07eae'/>
<id>urn:sha1:11b604c87d574d65c18b7da541b204d8b7f07eae</id>
<content type='text'>
commit e16605855d58803fe0608417150c7a618b4f8243 upstream.

Based on latest production information.

Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ALSA: hda/realtek - Fix ADC assignment with a shared HP/Mic pin</title>
<updated>2012-04-13T16:13:50Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2012-03-22T13:36:50Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=17c7ceb39193493ccf12d489fdfdaf2cdccb0099'/>
<id>urn:sha1:17c7ceb39193493ccf12d489fdfdaf2cdccb0099</id>
<content type='text'>
commit 26acaf08556a3c64ebf8ea3654b51e6acbb0a26c upstream.

The recent Realtek driver tries to assign an extra input via the
headphone plug when only a single input source is found.  The code
worked on Samsung Q1, but it broke ASUS 1015 where the mic is a
digital-mic and only a specific ADC works.

This patch fixes the assignment of ADC in the shared mic/hp case.
Instead of assuming the single ADC at first, reduce the ADCs after
trying to assign both mic and HP pins.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42973

Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ALSA: hda - fix printing of high HDMI sample rates</title>
<updated>2012-04-02T17:31:55Z</updated>
<author>
<name>Anssi Hannula</name>
<email>anssi.hannula@iki.fi</email>
</author>
<published>2012-03-13T15:43:02Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=0a86aeed32e7badc3a8e695c797c67d2c396cb62'/>
<id>urn:sha1:0a86aeed32e7badc3a8e695c797c67d2c396cb62</id>
<content type='text'>
commit 25dc16f69892182192b1234594fd3cf342ad4195 upstream.

A previous commit af65cbf296 (ALSA: hdmi: fix printout of SAD sampling
rates) fixed the sample rates shown in /proc/asound/cardX/eldY and
kernel log to not be entirely wrong. However, a missing rate from the
array added in the patch causes HDMI rates 88.2 kHz, 96 kHz, 176.4 kHz,
and 192 kHz to be shown as 96 kHz, 176.4 kHz, 192 kHz, and 384 kHz,
respectively.

Fix the reporting by adding the ALSA rate 64 kHz into the conversion
array between 48 kHz and 88.2 kHz.

Signed-off-by: Anssi Hannula &lt;anssi.hannula@iki.fi&gt;
Cc: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ASoC: pxa-ssp: atomically set stream active masks</title>
<updated>2012-04-02T17:31:54Z</updated>
<author>
<name>Daniel Mack</name>
<email>zonque@gmail.com</email>
</author>
<published>2012-03-19T08:12:53Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=67e5208f9495212fdcffcce64fbc9a970faade6b'/>
<id>urn:sha1:67e5208f9495212fdcffcce64fbc9a970faade6b</id>
<content type='text'>
commit 273b72c8ce6b28df6b49423d775c3e59072c73c5 upstream.

PXA's SSP engine fails to take its current channel phase into account
when enabling a stream while the engine is already running. This
results in randomly swapped left/right channels on either the record
or the playback side, depending on which one was enabled first.

The following patch fixes this by factoring out the bit field
modifications in question to a separate function that pauses the
engine temporarily, modifies the bits and kicks it off again
afterwards. Appearantly, a transition of SSCR0_SSE syncs both
directions properly.

The patch has been rolled out to quite a number of devices over the
last weeks and seems to fix the issue reliably.

Signed-off-by: Daniel Mack &lt;zonque@gmail.com&gt;
Reported-and-tested-by: Sven Neumann &lt;s.neumann@raumfeld.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master</title>
<updated>2012-04-02T17:31:54Z</updated>
<author>
<name>Timur Tabi</name>
<email>timur@freescale.com</email>
</author>
<published>2012-03-16T21:32:52Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=e7e9f000debc0128d1bc91b3a08e3f9013187d7a'/>
<id>urn:sha1:e7e9f000debc0128d1bc91b3a08e3f9013187d7a</id>
<content type='text'>
commit 70ac07bb633dee75ac554195b9a4d69adfa7803c upstream.

The WM8776 codec driver requires the machine driver to set one of the
SND_SOC_DAIFMT_CBx_xxx values.  The P1022DS machine driver should be setting
SND_SOC_DAIFMT_CBM_CFM, but since that value was zero, no one noticed.

Commit 75d9ac46 ("ASoC: Allow DAI formats to be specified in the
dai_link"), however, changed the value of SND_SOC_DAIFMT_CBM_CFM from zero
to a non-zero value, which means that it now needs to be specifically set
by the machine driver.

We also set SND_SOC_DAIFMT_NB_NF, for the same reason.

Signed-off-by: Timur Tabi &lt;timur@freescale.com&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ALSA: hdspm - Provide ioctl_compat</title>
<updated>2012-03-08T14:58:58Z</updated>
<author>
<name>Adrian Knoth</name>
<email>adi@drcomp.erfurt.thur.de</email>
</author>
<published>2012-03-08T14:38:04Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=8de5d6f19bbe7c77676a62ab52be901aa10d6b54'/>
<id>urn:sha1:8de5d6f19bbe7c77676a62ab52be901aa10d6b54</id>
<content type='text'>
snd_hdspm uses its own ioctls to acquire config- and status information.
Expose the corresponding ioctl handler via ioctl_compat, so that 32bit
applications can use it on 64bit kernels.

Signed-off-by: Adrian Knoth &lt;adi@drcomp.erfurt.thur.de&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>ALSA: hda/realtek - Apply the coef-setup only to ALC269VB</title>
<updated>2012-03-07T08:03:29Z</updated>
<author>
<name>Kailang Yang</name>
<email>kailang@realtek.com</email>
</author>
<published>2012-03-07T07:25:20Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=526af6eb4dc71302f59806e2ccac7793963a7fe0'/>
<id>urn:sha1:526af6eb4dc71302f59806e2ccac7793963a7fe0</id>
<content type='text'>
The coef setup in alc269_fill_coef() was designed only for ALC269VB
model, and this has some bad effects for other ALC269 variants, such
as turning off the external mic input.  Apply it only to ALC269VB.

Signed-off-by: Kailang Yang &lt;kailang@realtek.com&gt;
Cc: &lt;stable@kernel.org&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>ALSA: hda - add quirk to detect CD input on Gigabyte EP45-DS3</title>
<updated>2012-03-06T09:44:01Z</updated>
<author>
<name>Marton Balint</name>
<email>cus@passwd.hu</email>
</author>
<published>2012-03-05T20:33:23Z</published>
<link rel='alternate' type='text/html' href='https://git.amat.us/linux/commit/?id=8f2392142346f2754c8292a94cc62a157ed1e093'/>
<id>urn:sha1:8f2392142346f2754c8292a94cc62a157ed1e093</id>
<content type='text'>
My CD input got lost in commit 68ef0561efe494143516df38c03a16b837b8e79c.
Raymond helped me to add the necessary pin fixup to make it appear again. In
fact, this is basically his patch. It fixes alsa bug #5541.

Signed-off-by: Marton Balint &lt;cus@passwd.hu&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
</feed>
