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Author SHA1 Message Date
henrik.lundin@webrtc.org
5b111b06fa Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM" 
The change was reverted since it was thought to cause a flaky test.
But the test kept flaking after the change was reverted.

This effectively reverts r6394, relanding r6377.

BUG=3496
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 14:37:21 +00:00
phoglund@webrtc.org
8454ad1b3e Reland: Making WebRTC able to play and record audio to files for tests. 
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 14:12:04 +00:00
henrik.lundin@webrtc.org
ab85187e63 Remove unused resource 
The file resources/audio_coding/neteq_universal.rtp is no longer
used in any test. Removing the hash file neteq_universal.rtp.sha1.

BUG=2996
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:59:44 +00:00
pbos@webrtc.org
9e65a3b013 Re-land webrtcmediaengine.cc part of r6397. 
webrtcvideoengine.cc un-reverted by a bot roll in r6399 so half of r6397
is still applied. The applied fix (diff between r6397) is to put
WebRtcVideoEngine2 in ifdefs and only build for WEBRTC_CHROMIUM_BUILDs
corresponding to webrtcmediaengine.h.

BUG=
R=minyue@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19719005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:42:37 +00:00
stefan@webrtc.org
fbb567dacd Add APIs to enable padding with redundant payloads. 
Also makes a small change to the tests to remove flakiness. We can't do
BWE only based on rtp timestamps if we preemptively resend packets instead
of sending padding packets.

BUG=1812,2992
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:41:36 +00:00
buildbot@webrtc.org
5d223a7d2d (Auto)update libjingle 68982444-> 68983526 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:05:05 +00:00
minyue@webrtc.org
6604c6df26 Revert 6397 "(Auto)update libjingle 68949184-> 68982444" 
> (Auto)update libjingle 68949184-> 68982444

TBR=buildbot@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:02:36 +00:00
buildbot@webrtc.org
af214d804f (Auto)update libjingle 68949184-> 68982444 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 12:46:49 +00:00
minyue@webrtc.org
e08a11c4a1 Revert 6395 "Making WebRTC able to play and record audio to file..." 
> Making WebRTC able to play and record audio to files for tests.
> 
> By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
> WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
> play out audio to a file and feed audio in from a file. We want to do
> so we can better test WebRTC-using applications by recording what the
> audio stack outputs and feeding known audio in for quality tests.
> 
> R=henrika@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/20609004

TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 10:40:30 +00:00
phoglund@webrtc.org
fa042ca15d Making WebRTC able to play and record audio to files for tests. 
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 09:57:23 +00:00
henrik.lundin@webrtc.org
c726b1fc33 Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" 
BUG=3469
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 08:35:53 +00:00
bjornv@webrtc.org
18026abd82 common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 
This macro is only used at a few places and implies a cast to uint16_t before right shifting. All places already use uint16_t. Further, the amount of shifts applied in the macro has no sanity check for negativity, makes the macro dangerous to use.

BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 06:53:20 +00:00
bjornv@webrtc.org
782978cfcb common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix 
This macro is only used by the fixed point version of iSAC. Replacing the (five) locations in arith_routines_logist.c, where it is used, with the actual operation.

BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 06:39:03 +00:00
bjornv@webrtc.org
3f83072c26 modules/audio_processing: Adds a config for reported delays 
There are platforms and devices where the reported delays are untrusted and we currently solve that with an extended filter length and a slightly more conservative delay handling.
With this change we give the user the possibility to turn off reported system delay values completely.

- Includes new unit tests.

TESTED=trybots and manual testing
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 04:48:11 +00:00
jiayl@webrtc.org
e61b8e32d8 Adds end to end DataChannel tests. 
BUG=2626
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 23:54:13 +00:00
glaznev@webrtc.org
a40210aee2 Add support for NVidia VP8 HW encoder. 
- Some changes in HW VP8 encoder search logic to detect HW codec
with supported color space format.
- Support yuv420 and nv12 formants for encoder input.
- Add some extra logging and encoder frame drop statistics.

BUG=3176
R=fischman@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 23:48:29 +00:00
henrik.lundin@webrtc.org
fd59c39caa Delete last file in neteq4 folder 
The .isolate file can now be safely removed, since issue 3462 is
resolved.

BUG=2996
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 20:26:27 +00:00
andrew@webrtc.org
919914d71b MIPS optimizations for ISAC (patch #1) 
Implemented functions:
    - WebRtcIsacfix_AutocorrMIPS
    - WebRtcIsacfix_FilterArLoop
    - WebRtcIsacfix_FilterMaLoopMIPS
    - WebRtcIsacfix_AllpassFilter2FixDec16MIPS (only MIPS DSP)
    - WebRtcIsacfix_PitchFilterCore (only MIPS DSPR2)

Gain achieved: from aprox. 15% (MIPS32) up to aprox. 40% (MIPS DSPR2)

R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17559005

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 18:13:15 +00:00
mflodman@webrtc.org
0d7ab0a634 Adding the new video folder and pacer to the wathclist. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 13:59:37 +00:00
kwiberg@webrtc.org
12cd443752 Noise suppression: Change signature to work on floats instead of ints 
Internally, it already worked on floats. This patch just changes the
signature of a bunch of functions so that floats can be passed
directly from the new and improved AudioBuffer without converting the
data to int and back again first.

(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the noise suppressor comes immediately after
the echo canceller, which also works on floats. If I truncate to
integers between the two steps, ApmTest.Process doesn't complain, but
of course that's exactly the sort of thing the float conversion is
supposed to let us avoid...)

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 11:13:09 +00:00
kjellander@webrtc.org
1014101470 Revert 6380 "Replace libjingle_root with talk_root variable." 
It turns out this doesn't fix the problem we're trying to solve...

> Replace libjingle_root with talk_root variable.
> 
> This CL is similar to https://review.webrtc.org/9019004/
> It is needed in order to be able to build with different
> copies of libjingle. Having the libjingle_root variable didn't
> make this possible, since relative paths in the .isolate files
> ended up at the wrong directory level and .isolate files doesn't
> support all the normal GYP variables like <(DEPTH).
> 
> BUG=chromium:343106
> TEST=trybots passing compile step with clobber.
> R=tommi@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15709004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 10:13:38 +00:00
buildbot@webrtc.org
3eb2c2f4c3 (Auto)update libjingle 68891947-> 68893961 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 09:39:06 +00:00
pbos@webrtc.org
86f613d6b8 Move WebRtcVideoEngine2 fakes to unittest header. 
BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 08:53:05 +00:00
asapersson@webrtc.org
734a532723 Add additional metric (relative standard deviation of encode time) for overuse detection. 
This code is currently only for testing.

BUG=1577
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 06:35:22 +00:00
kjellander@webrtc.org
0238682984 Replace libjingle_root with talk_root variable. 
This CL is similar to https://review.webrtc.org/9019004/
It is needed in order to be able to build with different
copies of libjingle. Having the libjingle_root variable didn't
make this possible, since relative paths in the .isolate files
ended up at the wrong directory level and .isolate files doesn't
support all the normal GYP variables like <(DEPTH).

BUG=chromium:343106
TEST=trybots passing compile step with clobber.
R=tommi@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:46:31 +00:00
kjellander@webrtc.org
7b82c18979 Add kjellander@webrtc.org as OWNER for *.isolate 
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.

BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:42:53 +00:00
henrik.lundin@webrtc.org
620048172c Create a joint encoder/decoder wrapper for iSAC in ACM 
This CL extends the ACMISAC wrapper class to inherit from AudioDecoder
as well (the type of object that NetEq uses). The class has it's own
lock protecting the iSAC instance. This way, we can remove the
neteq_decode_lock_ (a.k.a. decoder_lock_) in a later CL.

The old AcmAudioDecoderIsac class is deleted.

R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 18:39:00 +00:00
henrik.lundin@webrtc.org
a90abdef62 Add thread annotations to AcmReceiver 
This change adds thread annotations to AcmReceiver. These are the
annotations that could be added without changing acquiring the locks in
more locations, or changing the lock structure.

BUG=3401
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6376 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 18:35:11 +00:00
henrik.lundin@webrtc.org
190a32fd55 Make some methods in Clock class const declared 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 17:40:49 +00:00
kjellander@webrtc.org
6b6e58d632 Remove unused test_env.py from isolate files + fix nss path. 
This is not necessary for executing tests for WebRTC.
It probably appeared in our .isolate files because of code
copied from Chromium.

BUG=
TEST=All non-baremetal trybots passing.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 14:35:09 +00:00
stefan@webrtc.org
85d2794e5b Adds support for the "apt" format parameter and turns on the RTX feature. 
BUG=1811,1095
R=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 12:51:39 +00:00
bjornv@webrtc.org
ed7edb8e89 Enables DelayCorrection tests 
The fix has been done elsewhere and the test pass.

BUG=3445
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15679007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 10:02:05 +00:00
phoglund@webrtc.org
582367f251 Updated conformance tests and w3c-ified them. 
I intend here to put these up for review on W3C. This moves the tests
to use the W3C-style vendor prefix handling and updates the tests to
the latest drafts.

This yields 44 Pass 24 Fail and 13 pass 54 fail 1 timeout on Firefox.
As far I can tell all failures are correct; in particular FF media
media stream tracks do not adhere to the standard.

Also I can't get FF to get a remote video up in the peerconnection
test, just the local one.

BUG=webrtc:3455
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 09:47:44 +00:00
henrik.lundin@webrtc.org
a1a2c0c190 Multi-threaded unit test for Audio Coding Module using iSAC 
This test extends AudioCodingModuleTest and AudioCodingModuleMtTest
to using iSAC as codec.

R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 09:37:17 +00:00
bjornv@webrtc.org
cb0ea43e57 audio_processing: Forces extended filter to be used in splitting filter test. 
The behavior differ between "normal" and "extended" modes when using AEC. In the extended filter mode nothing is processed until we have received a farend frame. This is exactly what is needed in this part of the splitting filter test.
On Android, we do not use the normal mode, which made the test to fail.

BUG=3445
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 08:21:52 +00:00
henrik.lundin@webrtc.org
9c55f0f957 Rename neteq4 folder to neteq 
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.

This CL effectively reverts r6257 "Rename neteq4 folder to neteq".

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 08:10:28 +00:00
kjellander@webrtc.org
31f967c611 Fix Dr Memory download 
In http://crrev.com/275232 the drmemory.DEPS directory was removed
since the Chromium bots have moved over to download from Google
Storage (http://crrev.com/275048).
This CL changes WebRTC to use the same approach.

Ideally the revision for the Dr Memory DEPS entry should use the
chromium_revision variable, but when I tried to roll to that revision
in https://review.webrtc.org/19679004/ I ran into errors with leaks
being detected in the compile step on the Linux ASan bot.
This CL allows our Dr Memory bots to go green while investigating this.

BUG=chromium:381366
TEST=Passing Win Dr Memory trybots.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 07:30:37 +00:00
henrik.lundin@webrtc.org
9221ab420d Re-enable AudioCodingModuleMtTest again 
Increase timeout and decrease test length.

BUG=3426
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15679006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-08 21:43:45 +00:00
kjellander@webrtc.org
9359edaf78 PRESUBMIT: Add Android ARM64 and remove Linux TSan 
Update the default trybots due to recent changes in the
trybots available.

TBR=tommi@webrtc.org
BUG=chromium:354539

Review URL: https://webrtc-codereview.appspot.com/21619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-08 17:55:51 +00:00
jiayl@webrtc.org
e3cdd9959e Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio." 
This reverts commit 56631a14bdae24aa0bfaceeb2b57df729fee1227.

TBR=henrike@webrtc.org
BUG=3235

Review URL: https://webrtc-codereview.appspot.com/19669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 22:32:57 +00:00
tkchin@webrtc.org
013bdf802a APPRTCDemo(objc): Remove regex parsing in favor of JSON struct. 
Also some cleanup/refactoring of APPRTCAppClient.

R=fischman@webrtc.org, glaznev@webrtc.org
BUG=3407

Review URL: https://webrtc-codereview.appspot.com/18499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 22:29:10 +00:00
fischman@webrtc.org
24c1778651 Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera." 
Makes stopping flakier for some reason :/

BUG=
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 22:24:40 +00:00
glaznev@webrtc.org
c3288c130d Add OpenGL Android video renderer which can display multiple 
yuv420 images in a single GLSurfaceView.
Start using new video renderer in AppRTC demo app.

BUG=
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 21:57:46 +00:00
jiayl@webrtc.org
b8f582591f Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed. 
BUG=crbug/374457
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 21:42:00 +00:00
fischman@webrtc.org
171d94177b AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera. 
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 21:22:37 +00:00
fischman@webrtc.org
b464618c84 Unbreak NDEBUG compile by RTC_UNUSED()ing an assert()d variable. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 20:13:49 +00:00
jiayl@webrtc.org
745a39cced Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio. 
BUG=3235
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 19:24:02 +00:00
fischman@webrtc.org
b273b60154 ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. 
Sure would be nice if the try fleet used both gcc _and_ clang...

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:59:30 +00:00
fischman@webrtc.org
9512719569 AppRTCDemo(android): support app (UI) & capture rotation. 
Now app UI rotates as the device orientation changes, and the captured stream
tries to maintain real-world-up, matching Chrome/Android and Hangouts/Android
behavior.

BUG=2432
R=glaznev@webrtc.org, henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:40:44 +00:00
fischman@webrtc.org
42694c5937 VideoCaptureImpl::IncomingFrame(): avoid deadlock by acquiring _apiCs. 
Since VCI::IF() fires a callback it risks a call back into VCI on the same
stack.  Failing to acquire _apiCs before _callbackCs means this is a lock
inversion and deadlock results.  By acquiring _apiCs first no lock inversion
occurs and the deadlock is removed.

BUG=3434
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:28:28 +00:00