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Author SHA1 Message Date
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
buildbot@webrtc.org
91c910469f (Auto)update libjingle 68701339-> 68703656 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 16:29:00 +00:00
pbos@webrtc.org
910473b31a Fix C++11 -Wnarrowing in channel_unittest.cc. 
Implicit conversion from int to unsigned char inside {} initializers is
ill-formed C++11 and triggers a warning in clang when building it as
such.

BUG=
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 15:44:00 +00:00
buildbot@webrtc.org
7b6cbb3aa0 (Auto)update libjingle 68689052-> 68689059 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 10:54:08 +00:00
pbos@webrtc.org
6ae48c6609 Make VideoSendStream/VideoReceiveStream configs const. 
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.

CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.

This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.

R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
BUG=3260

Review URL: https://webrtc-codereview.appspot.com/20409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 10:49:19 +00:00
buildbot@webrtc.org
4b83a471de (Auto)update libjingle 68646004-> 68648993 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 21:11:28 +00:00
henrike@webrtc.org
4e5f65a4c6 Rebase webrtc/base with r6345 version of talk/base: 
cd webrtc/base
svn diff -r 6249:6300 http://webrtc.googlecode.com/svn/trunk/talk/base >
6300.diff
patch -p0 -i 6300.diff
ls genericslot* | xargs rm
cp ../../talk/base/sigslottester* .
manual edits of sigslottester* to get rid of talk and talk_base.

BUG=3379
TBR=jiayang

Review URL: https://webrtc-codereview.appspot.com/19649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 20:40:11 +00:00
wu@webrtc.org
94454b71ad Fix the chain that propagates the audio frame's rtp and ntp timestamp including: 
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 20:34:08 +00:00
fischman@webrtc.org
130fa64d4c AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct. 
BUG=3407
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 20:31:41 +00:00
tina.legrand@webrtc.org
65d61c3924 Opus send rate overflows if over 65 kbps 
The member holding the send rate for Opus had too low resolution for rates above ~65 kbps.

I've added a test that checks if the average rate in a Opus test is in the right range. The test fails before my fix, and now passes.

BUG=3267
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 13:42:51 +00:00
bjornv@webrtc.org
b51d3ea593 Revert 6341 "Fixes and enables SystemDelayTests." 
> Fixes and enables SystemDelayTests.
> 
> The root cause for failure was that the delay handling of reported delays was bypassed on Android, whereas the tests assumes that part of AEC to be run.
> This CL checks if it is in use.
> 
> BUG=3445
> R=kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12689005

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 13:41:33 +00:00
kjellander@webrtc.org
681aaae71b Remove remaining samples (AppRTC) since moved to Github 
In r5871 the samples directory was removed since they've now
moved to GitHub at https://github.com/GoogleChrome/webrtc

AppRTC needed to be kept in here (restored in r5873) since
automated tests in Chromium pulled AppRTC.
Now that a Chromium mirror has been setup for the GitHub repo
and that the automated tests have been updated, we can remove
this once and for all.

BUG=chromium:362483
TEST=None, but the automated tests have been verified syncing
the new location.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6342 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 13:09:59 +00:00
bjornv@webrtc.org
1f971b5788 Fixes and enables SystemDelayTests. 
The root cause for failure was that the delay handling of reported delays was bypassed on Android, whereas the tests assumes that part of AEC to be run.
This CL checks if it is in use.

BUG=3445
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 10:58:55 +00:00
henrik.lundin@webrtc.org
2f816bbae7 NetEq: Add thread annotation to const scoped_ptrs 
Since the objects pointed to are not const, only the pointer to them,
they too must be accessed under lock.

Move the crit_sect to above the variables it is protecting.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12679006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 10:37:13 +00:00
mflodman@webrtc.org
eae7924836 Adding back platform specific renderer to video loopback test. 
BUG=3039
TEST=locally on Mac and Win, video_loopback test
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 09:32:51 +00:00
pbos@webrtc.org
0d523eea83 Remove static initializer from WebRtcVideoEngine2. 
BUG=
R=pliard@google.com, pthatcher@webrtc.org, pliard@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 09:10:55 +00:00
bjornv@webrtc.org
aafd7a88c5 The correct fix of workaround in r6261. 
The CL also includes same changes to filterbanks.c in iSAC fix and aecm_core_c.c

BUG=3370,3395,3439
TESTED=trybots
R=fdegans@chromium.org, glaznev@webrtc.org, kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:53:51 +00:00
bjornv@webrtc.org
edbe886a0b common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND 
This macro was only used at two places in fixed point iSAC, where it has been replaced with the operation.

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

Review URL: https://webrtc-codereview.appspot.com/15669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:42:53 +00:00
stefan@webrtc.org
ef92755780 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.

BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:25:29 +00:00
henrik.lundin@webrtc.org
c578962006 Disable a test in libjingle_peerconnection_unittest for DrMemory 
BUG=3453
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 07:38:31 +00:00
buildbot@webrtc.org
f1adbeedb4 (Auto)update libjingle 68562943-> 68571194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 21:57:16 +00:00
henrike@webrtc.org
e6e139159f Android: cleanup gtest_target_type conditions. 
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library

Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).

R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 20:46:50 +00:00
tkchin@webrtc.org
738df8913d Fix retain cycle in RTCEAGLVideoView. 
CADisplayLink increases its target's refcount. In order to break retain cycle, we wrap CADisplayLink in a new RTCDisplayLinkTimer class and use that instead.

R=fischman@webrtc.org, noahric@chromium.org
BUG=3391

Review URL: https://webrtc-codereview.appspot.com/16599006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 20:19:39 +00:00
solenberg@webrtc.org
c6db88b0cf Make it possible to build webrtc for arm64. 
- Bump revision of protobuf lib
- Remove -Wextra for arm64 gcc targets (warnings in stlport)
- Add MemoryBarrier implementation in single_rw_fifo.cc.
- [pending 15619004]: Bump revision of /deps/tools/android to get md5sum_bin for arm64.

BUG=chromium:354405,chromium:354539
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 17:15:42 +00:00
buildbot@webrtc.org
6f237769b3 (Auto)update libjingle 68507189-> 68543735 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 16:23:10 +00:00
buildbot@webrtc.org
40b45fc07a (Auto)update libjingle 68506654-> 68507189 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 14:48:33 +00:00
henrik.lundin@webrtc.org
d3dcebf6b4 Disable P2PTransportChannelMultihomedTest.TestFailover under Memcheck 
BUG=3447
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 13:23:07 +00:00
bjornv@webrtc.org
147f4fe3c0 Disables SystemDelayTest.CorrectDelayDuringDrift on Android 
Should have been part of https://webrtc-codereview.appspot.com/19629004/

BUG=3445
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 13:17:58 +00:00
bjornv@webrtc.org
b616e1211f Disables some modules_unittests on Android. 
BUG=3445
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 12:12:58 +00:00
andresp@webrtc.org
4436b4436a Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 09:05:30 +00:00
henrik.lundin@webrtc.org
2bdd3994e3 Suppress memcheck error in VideoProcessorIntegrationTest 
BUG=3446
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19629005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 08:47:29 +00:00
mflodman@webrtc.org
19fc09efba Adding missing break in media_file_utility.cc. 
There has been no reports of problems, but adding this to get it correct.

Review URL: https://webrtc-codereview.appspot.com/19599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 05:21:56 +00:00
buildbot@webrtc.org
0cdcd23a03 (Auto)update libjingle 68501302-> 68506654 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 01:31:14 +00:00