<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux-stable.git/sound, branch v3.2.16</title>
<subtitle>Linux kernel stable tree</subtitle>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/'/>
<entry>
<title>ASoC: ak4642: fixup: mute needs +1 step</title>
<updated>2012-04-13T15:33:50+00:00</updated>
<author>
<name>Kuninori Morimoto</name>
<email>kuninori.morimoto.gx@renesas.com</email>
</author>
<published>2012-04-05T06:28:01+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=b0a4205ff563f69026c852834a78a1a5952421b2'/>
<id>b0a4205ff563f69026c852834a78a1a5952421b2</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>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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;

</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: wm8994: Update WM8994 DCS calibration</title>
<updated>2012-04-13T15:33:45+00:00</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2012-03-21T13:22:40+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=97ff9284d7872dad9785ebb4b78d030b7c3cde5c'/>
<id>97ff9284d7872dad9785ebb4b78d030b7c3cde5c</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>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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;

</pre>
</div>
</content>
</entry>
<entry>
<title>ALSA: hda - fix printing of high HDMI sample rates</title>
<updated>2012-04-02T16:52:45+00:00</updated>
<author>
<name>Anssi Hannula</name>
<email>anssi.hannula@iki.fi</email>
</author>
<published>2012-03-13T15:43:02+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=d8b8c61ff79f4edc8ef979a9fcb94b79beb77cdf'/>
<id>d8b8c61ff79f4edc8ef979a9fcb94b79beb77cdf</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>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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;

</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: pxa-ssp: atomically set stream active masks</title>
<updated>2012-04-02T16:52:44+00:00</updated>
<author>
<name>Daniel Mack</name>
<email>zonque@gmail.com</email>
</author>
<published>2012-03-19T08:12:53+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=0f813454b313d17992a2c8ec7483c733ce5107b3'/>
<id>0f813454b313d17992a2c8ec7483c733ce5107b3</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>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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;

</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master</title>
<updated>2012-04-02T16:52:44+00:00</updated>
<author>
<name>Timur Tabi</name>
<email>timur@freescale.com</email>
</author>
<published>2012-03-16T21:32:52+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=34ef160b39875c7f823f29b8925e1331d940148c'/>
<id>34ef160b39875c7f823f29b8925e1331d940148c</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>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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;

</pre>
</div>
</content>
</entry>
<entry>
<title>ALSA: hda/realtek - Apply the coef-setup only to ALC269VB</title>
<updated>2012-03-19T16:02:18+00:00</updated>
<author>
<name>Kailang Yang</name>
<email>kailang@realtek.com</email>
</author>
<published>2012-03-07T07:25:20+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=46bf2e14af13fece62ff6f4f294f6658578f1600'/>
<id>46bf2e14af13fece62ff6f4f294f6658578f1600</id>
<content type='text'>
commit 526af6eb4dc71302f59806e2ccac7793963a7fe0 upstream.

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

</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
commit 526af6eb4dc71302f59806e2ccac7793963a7fe0 upstream.

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

</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: neo1973: fix neo1973 wm8753 initialization</title>
<updated>2012-03-19T16:02:18+00:00</updated>
<author>
<name>Denis 'GNUtoo' Carikli</name>
<email>GNUtoo@no-log.org</email>
</author>
<published>2012-02-26T18:21:54+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=4bfcbebdbe00810c14579ddcf73f33805676b129'/>
<id>4bfcbebdbe00810c14579ddcf73f33805676b129</id>
<content type='text'>
commit b2ccf065f7b23147ed135a41b01d05a332ca6b7e upstream.

The neo1973 driver had wrong codec name which prevented the "sound card"
from appearing.

Signed-off-by: Denis 'GNUtoo' Carikli &lt;GNUtoo@no-log.org&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
commit b2ccf065f7b23147ed135a41b01d05a332ca6b7e upstream.

The neo1973 driver had wrong codec name which prevented the "sound card"
from appearing.

Signed-off-by: Denis 'GNUtoo' Carikli &lt;GNUtoo@no-log.org&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: i.MX SSI: Fix DSP_A format.</title>
<updated>2012-03-12T19:31:36+00:00</updated>
<author>
<name>Javier Martin</name>
<email>javier.martin@vista-silicon.com</email>
</author>
<published>2012-02-23T14:43:18+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=b3b7b02653b8b9f8f2c6fabe97801600e4716d74'/>
<id>b3b7b02653b8b9f8f2c6fabe97801600e4716d74</id>
<content type='text'>
commit 5ed80a75b248bfaf840ea6b38f941edcf6ee7dc7 upstream.

According to i.MX27 Reference Manual (p 1593) TXBIT0 bit selects
whether the most significant or the less significant part of the
data word written to the FIFO is transmitted.

As DSP_A is the same as DSP_B with a data offset of 1 bit, it
doesn't make any sense to remove TXBIT0 bit here.

Signed-off-by: Javier Martin &lt;javier.martin@vista-silicon.com&gt;
Acked-by: Sascha Hauer &lt;s.hauer@pengutronix.de&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
commit 5ed80a75b248bfaf840ea6b38f941edcf6ee7dc7 upstream.

According to i.MX27 Reference Manual (p 1593) TXBIT0 bit selects
whether the most significant or the less significant part of the
data word written to the FIFO is transmitted.

As DSP_A is the same as DSP_B with a data offset of 1 bit, it
doesn't make any sense to remove TXBIT0 bit here.

Signed-off-by: Javier Martin &lt;javier.martin@vista-silicon.com&gt;
Acked-by: Sascha Hauer &lt;s.hauer@pengutronix.de&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: dapm: Check for bias level when powering down</title>
<updated>2012-03-12T19:31:36+00:00</updated>
<author>
<name>Mark Brown</name>
<email>broonie@opensource.wolfsonmicro.com</email>
</author>
<published>2012-02-22T15:52:56+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=aee50f92f5b1b34650f76b164537691be95bfa24'/>
<id>aee50f92f5b1b34650f76b164537691be95bfa24</id>
<content type='text'>
commit 7679e42ec833ed70aa34790a5f39dcb7e5bda4fe upstream.

Recent enhancements in the bias management means that we might not be
in standby when the CODEC is idle and can have active widgets without
being in full power mode but the shutdown functionality assumes these
things. Add checks for the bias level at each stage so that we don't
do transitions other than the ON-&gt;PREPARE-&gt;STANDBY-&gt;OFF ones that the
drivers are expecting.

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

</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
commit 7679e42ec833ed70aa34790a5f39dcb7e5bda4fe upstream.

Recent enhancements in the bias management means that we might not be
in standby when the CODEC is idle and can have active widgets without
being in full power mode but the shutdown functionality assumes these
things. Add checks for the bias level at each stage so that we don't
do transitions other than the ON-&gt;PREPARE-&gt;STANDBY-&gt;OFF ones that the
drivers are expecting.

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

</pre>
</div>
</content>
</entry>
<entry>
<title>ALSA: hda - Always set HP pin in unsol handler for STAC/IDT codecs</title>
<updated>2012-03-12T19:31:24+00:00</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2012-02-29T08:41:17+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=b1748825081ec6b89c1da25a996a7591087ca39b'/>
<id>b1748825081ec6b89c1da25a996a7591087ca39b</id>
<content type='text'>
commit 7bff172a352a2fbe9856bba517d71a2072aab041 upstream.

A bug report with an old Sony laptop showed that we can't rely on BIOS
setting the pins of headphones but the driver should set always by
itself.

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

</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
commit 7bff172a352a2fbe9856bba517d71a2072aab041 upstream.

A bug report with an old Sony laptop showed that we can't rely on BIOS
setting the pins of headphones but the driver should set always by
itself.

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

</pre>
</div>
</content>
</entry>
</feed>
