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<title>user/sven/linux.git/include/sound, branch v4.19.40</title>
<subtitle>Linux Kernel
</subtitle>
<id>https://git.stealer.net/cgit.cgi/user/sven/linux.git/atom?h=v4.19.40</id>
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<updated>2019-02-12T18:47:23Z</updated>
<entry>
<title>ALSA: compress: Fix stop handling on compressed capture streams</title>
<updated>2019-02-12T18:47:23Z</updated>
<author>
<name>Charles Keepax</name>
<email>ckeepax@opensource.cirrus.com</email>
</author>
<published>2019-02-05T16:29:40Z</published>
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<id>urn:sha1:93e7d51029bd366c0f4bd432edb9fbfa8430f543</id>
<content type='text'>
commit 4f2ab5e1d13d6aa77c55f4914659784efd776eb4 upstream.

It is normal user behaviour to start, stop, then start a stream
again without closing it. Currently this works for compressed
playback streams but not capture ones.

The states on a compressed capture stream go directly from OPEN to
PREPARED, unlike a playback stream which moves to SETUP and waits
for a write of data before moving to PREPARED. Currently however,
when a stop is sent the state is set to SETUP for both types of
streams. This leaves a capture stream in the situation where a new
start can't be sent as that requires the state to be PREPARED and
a new set_params can't be sent as that requires the state to be
OPEN. The only option being to close the stream, and then reopen.

Correct this issues by allowing snd_compr_drain_notify to set the
state depending on the stream direction, as we already do in
set_params.

Fixes: 49bb6402f1aa ("ALSA: compress_core: Add support for capture streams")
Signed-off-by: Charles Keepax &lt;ckeepax@opensource.cirrus.com&gt;
Cc: &lt;stable@vger.kernel.org&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: pcm: Fix interval evaluation with openmin/max</title>
<updated>2018-12-13T08:16:16Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2018-11-29T11:05:19Z</published>
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<id>urn:sha1:a2928e74981266381b5df13f150384a5c5120ff3</id>
<content type='text'>
commit 5363857b916c1f48027e9b96ee8be8376bf20811 upstream.

As addressed in alsa-lib (commit b420056604f0), we need to fix the
case where the evaluation of PCM interval "(x x+1]" leading to
-EINVAL.  After applying rules, such an interval may be translated as
"(x x+1)".

Fixes: ff2d6acdf6f1 ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max")
Cc: &lt;stable@vger.kernel.org&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>Merge tag 'asoc-v4.19-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus</title>
<updated>2018-09-17T16:59:21Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2018-09-17T16:59:21Z</published>
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<id>urn:sha1:196f4eeeb78f53e0a598db8f9408b6f8b270c355</id>
<content type='text'>
ASoC: Fixes for v4.19

This is the usual set of small fixes scatterd around various drivers,
plus one fix for DAPM and a UAPI build fix.  There's not a huge amount
that stands out here relative to anything else.
</content>
</entry>
<entry>
<title>sound: don't call skl_init_chip() to reset intel skl soc</title>
<updated>2018-09-12T10:22:09Z</updated>
<author>
<name>Yu Zhao</name>
<email>yuzhao@google.com</email>
</author>
<published>2018-09-11T21:15:16Z</published>
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<id>urn:sha1:75383f8d39d4c0fb96083dd460b7b139fbdac492</id>
<content type='text'>
Internally, skl_init_chip() calls snd_hdac_bus_init_chip() which
1) sets bus-&gt;chip_init to prevent multiple entrances before device
is stopped; 2) enables interrupt.

We shouldn't use it for the purpose of resetting device only because
1) when we really want to initialize device, we won't be able to do
so; 2) we are ready to handle interrupt yet, and kernel crashes when
interrupt comes in.

Rename azx_reset() to snd_hdac_bus_reset_link(), and use it to reset
device properly.

Fixes: 60767abcea3d ("ASoC: Intel: Skylake: Reset the controller in probe")
Reviewed-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Yu Zhao &lt;yuzhao@google.com&gt;
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: dapm: Fix NULL pointer deference on CODEC to CODEC DAIs</title>
<updated>2018-08-15T14:52:20Z</updated>
<author>
<name>Charles Keepax</name>
<email>ckeepax@opensource.cirrus.com</email>
</author>
<published>2018-08-15T12:11:35Z</published>
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<id>urn:sha1:249dc49576fc953a7378b916c6a6d47ea81e4da2</id>
<content type='text'>
Commit a655de808cbde ("ASoC: core: Allow topology to override
machine driver FE DAI link config.") caused soc_dai_hw_params to
be come dependent on the substream private_data being set with
a pointer to the snd_soc_pcm_runtime. Currently, CODEC to CODEC
links don't set this, which causes a NULL pointer dereference:

[&lt;4069de54&gt;] (soc_dai_hw_params) from
[&lt;40694b68&gt;] (snd_soc_dai_link_event+0x1a0/0x380)

Since the ASoC core in general assumes that the substream
private_data will be set to a pointer to the snd_soc_pcm_runtime,
update the CODEC to CODEC links to respect this.

Signed-off-by: Charles Keepax &lt;ckeepax@opensource.cirrus.com&gt;
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>Merge tag 'asoc-v4.19' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus</title>
<updated>2018-08-13T10:12:31Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2018-08-13T10:12:31Z</published>
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<id>urn:sha1:f5b6c1fcb42fe7d6f2f6eb2220512e2a5f875133</id>
<content type='text'>
ASoC: Updates for v4.19

A fairly big update, including quite a bit of core activity this time
around (which is good to see) along with a fairly large set of new
drivers.

 - A new snd_pcm_stop_xrun() helper which is now used in several
   drivers.
 - Support for providing name prefixes to generic component nodes.
 - Quite a few fixes for DPCM as it gains a bit wider use and more
   robust testing.
 - Generalization of the DIO2125 support to a simple amplifier driver.
 - Accessory detection support for the audio graph card.
 - DT support for PXA AC'97 devices.
 - Quirks for a number of new x86 systems.
 - Support for AM Logic Meson, Everest ES7154, Intel systems with
   RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707.
</content>
</entry>
<entry>
<title>ASoC: core: remove support for card rebind using component framework</title>
<updated>2018-08-02T16:19:46Z</updated>
<author>
<name>Srinivas Kandagatla</name>
<email>srinivas.kandagatla@linaro.org</email>
</author>
<published>2018-08-02T15:03:38Z</published>
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<id>urn:sha1:611cbc8799b672f41b6ba7afed758ad9efb959a7</id>
<content type='text'>
DRM based audio components get registered inside the component framework
bind callback. However component framework has a big mutex lock taken for
every call to component_add, component_del and bind, unbind callbacks.

This can lead to deadlock situation if we are trying to add new/remove
component within a bind/unbind callbacks. Which is what was happening
with bcm2837 rpi 3.

Revert this change till we sort out the mutex issue.

Reported-by: Guillaume Tucker &lt;guillaume.tucker@collabora.com&gt;
Reported-by: Stefan Wahren &lt;stefan.wahren@i2se.com&gt;
Signed-off-by: Srinivas Kandagatla &lt;srinivas.kandagatla@linaro.org&gt;
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: fsi: convert to SPDX identifiers</title>
<updated>2018-08-02T09:56:59Z</updated>
<author>
<name>Kuninori Morimoto</name>
<email>kuninori.morimoto.gx@renesas.com</email>
</author>
<published>2018-08-02T01:47:30Z</published>
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<id>urn:sha1:fdec79c18b08c68cfa079f2d3ee23c5a120a2eda</id>
<content type='text'>
Signed-off-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ALSA: seq: Minor cleanup of MIDI event parser helpers</title>
<updated>2018-08-01T20:54:35Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2018-08-01T12:38:18Z</published>
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<id>urn:sha1:ef965ad5a7697ff16e3be01954f5c57208e36c22</id>
<content type='text'>
snd_midi_event_encode_byte() can never fail, and it can return rather
true/false.  Change the return type to bool, adjust the argument to
receive a MIDI byte as unsigned char, and adjust the comment
accordingly.  This allows callers to drop error checks, which
simplifies the code.

Meanwhile, snd_midi_event_encode() helper is used only in seq_midi.c,
and it can be better folded into it.  This will reduce the total
amount of lines in the end.

Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>Merge branch 'topic/drm_audio_component' of https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-4.19</title>
<updated>2018-08-01T09:32:05Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@kernel.org</email>
</author>
<published>2018-08-01T09:32:05Z</published>
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