<feed xmlns='http://www.w3.org/2005/Atom'>
<title>user/sven/linux.git/sound, branch v3.18.74</title>
<subtitle>Linux Kernel
</subtitle>
<id>https://git.stealer.net/cgit.cgi/user/sven/linux.git/atom?h=v3.18.74</id>
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<updated>2017-10-08T08:11:22Z</updated>
<entry>
<title>ALSA: au88x0: avoid theoretical uninitialized access</title>
<updated>2017-10-08T08:11:22Z</updated>
<author>
<name>Arnd Bergmann</name>
<email>arnd@arndb.de</email>
</author>
<published>2017-03-23T15:15:55Z</published>
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<id>urn:sha1:0cd9a4647a23243db0d1c45ebd4df4242006b8f6</id>
<content type='text'>
commit 13f99ebdd602ebdafb909e15ec6ffb1e34690167 upstream.

The latest gcc-7.0.1 snapshot points out that we if nr_ch is zero, we never
initialize some variables:

sound/pci/au88x0/au88x0_core.c: In function 'vortex_adb_allocroute':
sound/pci/au88x0/au88x0_core.c:2304:68: error: 'mix[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized]
sound/pci/au88x0/au88x0_core.c:2305:58: error: 'src[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized]

I assume this can never happen in practice, but adding a check here doesn't
hurt either and avoids the warning. The code has been unchanged since
the start of git history.

Signed-off-by: Arnd Bergmann &lt;arnd@arndb.de&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>ALSA: msnd: Optimize / harden DSP and MIDI loops</title>
<updated>2017-09-13T21:03:47Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2017-07-06T10:34:40Z</published>
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<id>urn:sha1:bceac1033cd99ff5d2aaa69c700367f866bf6f04</id>
<content type='text'>
commit 20e2b791796bd68816fa115f12be5320de2b8021 upstream.

The ISA msnd drivers have loops fetching the ring-buffer head, tail
and size values inside the loops.  Such codes are inefficient and
fragile.

This patch optimizes it, and also adds the sanity check to avoid the
endless loops.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196131
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196133
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: grygorii tertychnyi &lt;gtertych@cisco.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ALSA: au88x0: Fix zero clear of stream-&gt;resources</title>
<updated>2017-09-02T05:05:45Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2016-05-12T16:04:16Z</published>
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<id>urn:sha1:687417919d03454cc417cf6c90fc5a59d118d96e</id>
<content type='text'>
commit 639db596165746ca87bbcb56559b094fd9042890 upstream.

There are a few calls of memset() to stream-&gt;resources, but they all
are called in a wrong size, sizeof(unsigned char) * VORTEX_RESOURCE_LAST,
while this field is a u32 array.  This may leave the memories not
zero-cleared.

Fix it by replacing them with a simpler sizeof(stream-&gt;resources)
instead.

Reported-by: David Binderman &lt;dcb314@hotmail.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>ALSA: hda - Add stereo mic quirk for Lenovo G50-70 (17aa:3978)</title>
<updated>2017-08-30T08:16:55Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2017-08-23T07:30:17Z</published>
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<id>urn:sha1:e3d62b2d359c52c9af8e18cec74080c22993b072</id>
<content type='text'>
commit bbba6f9d3da357bbabc6fda81e99ff5584500e76 upstream.

Lenovo G50-70 (17aa:3978) with Conexant codec chip requires the
similar workaround for the inverted stereo dmic like other Lenovo
models.

Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1020657
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: core: Fix unexpected error at replacing user TLV</title>
<updated>2017-08-30T08:16:55Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2017-08-22T06:15:13Z</published>
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<id>urn:sha1:639ea02e79ec2aab07f6bf9e78993e2066228ebc</id>
<content type='text'>
commit 88c54cdf61f508ebcf8da2d819f5dfc03e954d1d upstream.

When user tries to replace the user-defined control TLV, the kernel
checks the change of its content via memcmp().  The problem is that
the kernel passes the return value from memcmp() as is.  memcmp()
gives a non-zero negative value depending on the comparison result,
and this shall be recognized as an error code.

The patch covers that corner-case, return 1 properly for the changed
TLV.

Fixes: 8aa9b586e420 ("[ALSA] Control API - more robust TLV implementation")
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: usb-audio: Apply sample rate quirk to Sennheiser headset</title>
<updated>2017-08-25T00:00:10Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2017-08-14T12:35:50Z</published>
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<id>urn:sha1:f1bfa63f2e683a0d18c8508694d5aff81afe8298</id>
<content type='text'>
commit a8e800fe0f68bc28ce309914f47e432742b865ed upstream.

A Senheisser headset requires the typical sample-rate quirk for
avoiding spurious errors from inquiring the current sample rate like:
 usb 1-1: 2:1: cannot get freq at ep 0x4
 usb 1-1: 3:1: cannot get freq at ep 0x83

The USB ID 1395:740a has to be added to the entries in
snd_usb_get_sample_rate_quirk().

Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1052580
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: do not close shared backend dailink</title>
<updated>2017-08-11T16:30:13Z</updated>
<author>
<name>Banajit Goswami</name>
<email>bgoswami@codeaurora.org</email>
</author>
<published>2017-07-15T06:15:05Z</published>
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<id>urn:sha1:7de27d201336252bc0034e5d217047e6deb899ba</id>
<content type='text'>
commit b1cd2e34c69a2f3988786af451b6e17967c293a0 upstream.

Multiple frontend dailinks may be connected to a backend
dailink at the same time. When one of frontend dailinks is
closed, the associated backend dailink should not be closed
if it is connected to other active frontend dailinks. Change
ensures that backend dailink is closed only after all
connected frontend dailinks are closed.

Signed-off-by: Gopikrishnaiah Anandan &lt;agopik@codeaurora.org&gt;
Signed-off-by: Banajit Goswami &lt;bgoswami@codeaurora.org&gt;
Signed-off-by: Patrick Lai &lt;plai@codeaurora.org&gt;
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ALSA: hda - Fix speaker output from VAIO VPCL14M1R</title>
<updated>2017-08-11T16:30:13Z</updated>
<author>
<name>Sergei A. Trusov</name>
<email>sergei.a.trusov@ya.ru</email>
</author>
<published>2017-08-02T10:23:48Z</published>
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<id>urn:sha1:4b2bc1d63404af650cf6b87774bccf5f65152788</id>
<content type='text'>
commit 3f3c371421e601fa93b6cb7fb52da9ad59ec90b4 upstream.

Sony VAIO VPCL14M1R needs the quirk to make the speaker working properly.

Tested-by: Dmitriy &lt;mexx400@yandex.ru&gt;
Signed-off-by: Sergei A. Trusov &lt;sergei.a.trusov@ya.ru&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: tlv320aic3x: Mark the RESET register as volatile</title>
<updated>2017-08-11T16:30:12Z</updated>
<author>
<name>Peter Ujfalusi</name>
<email>peter.ujfalusi@ti.com</email>
</author>
<published>2016-12-23T09:21:10Z</published>
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<id>urn:sha1:72fafee16367d3c4782dd3f6869b297a91d7cee0</id>
<content type='text'>
[ Upstream commit 63c3194b82530bd71fd49db84eb7ab656b8d404a ]

The RESET register only have one self clearing bit and it should not be
cached. If it is cached, when we sync the registers back to the chip we
will initiate a software reset as well, which is not desirable.

Signed-off-by: Peter Ujfalusi &lt;peter.ujfalusi@ti.com&gt;
Reviewed-by: Jarkko Nikula &lt;jarkko.nikula@bitmer.com&gt;
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;alexander.levin@verizon.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</content>
</entry>
<entry>
<title>ASoC: compress: Derive substream from stream based on direction</title>
<updated>2017-07-27T22:03:25Z</updated>
<author>
<name>Satish Babu Patakokila</name>
<email>sbpata@codeaurora.org</email>
</author>
<published>2017-06-17T00:33:40Z</published>
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<id>urn:sha1:23fa7dc7efe4d11dea7f25430bee838cf24ef16a</id>
<content type='text'>
commit 01b8cedfd0422326caae308641dcadaa85e0ca72 upstream.

Currently compress driver hardcodes direction as playback to get
substream from the stream. This results in getting the incorrect
substream for compressed capture usecase.
To fix this, remove the hardcoding and derive substream based on
the stream direction.

Signed-off-by: Satish Babu Patakokila &lt;sbpata@codeaurora.org&gt;
Signed-off-by: Banajit Goswami &lt;bgoswami@codeaurora.org&gt;
Acked-By: Vinod Koul &lt;vinod.koul@intel.com&gt;
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
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