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Author SHA1 Message Date
stefan@webrtc.org
4ef438e2de Remove the send-side cname getter APIs from voice and video engine. 
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 09:55:30 +00:00
henrikg@webrtc.org
0f426685e1 Roll chromium_revision 280876:282462 
No significant DEPS changes in this roll, only some changes in how clang_format is downloaded.

clang_format changes based on https://webrtc-codereview.appspot.com/20829004 which was reverted.

R=henrika@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 08:10:19 +00:00
fbarchard@google.com
cb973686e8 roll libyuv to r1033 for clang-cl support on windows. 
BUG=chromium:391927
TESTED=manual testing libyuv compiles with clang-cl
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 23:40:15 +00:00
henrike@webrtc.org
b614d0626f Rebase webrtc/base with r6655 version of talk/base: 
cls to port: r6633,r6639 (there is no cl in between that affects base and all other talk/base cls took care of webrtc/base as well (see r6569, r6624)):
svn diff -r 6632:6639 http://webrtc.googlecode.com/svn/trunk/talk/base > 6655.diff
sed -i.bak "s/talk_base/rtc/g" 6655.diff
patch -p0 -i 6555.diff

BUG=3379
TBR=tommi@webrtc.org,jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 22:47:02 +00:00
pbos@webrtc.org
72491b9a90 Count total bytes sent in RTPSender::Bytes(). 
Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 16:24:54 +00:00
pbos@webrtc.org
0422100818 Fix data race in VCMTiming::ResetDecodeTime. 
Also thread annotating class.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 15:25:37 +00:00
pbos@webrtc.org
bd9c0920ec Skip encoding in fake VP8 encoder. 
Broke memcheck, FakeEncoder::Encode doesn't produce valid VP8 frames.

BUG=3424
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 13:21:40 +00:00
andresp@webrtc.org
7ae9108b60 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 10:35:12 +00:00
pbos@webrtc.org
91f1752f2d Support VP8 encoder settings in VideoSendStream. 
Stop-gap solution to support VP8 codec settings in the new API until
encoder settings can be passed on to the VideoEncoder without requiring
explicit support for the codec.

BUG=3424
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 10:13:37 +00:00
andresp@webrtc.org
8f1512140e Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 09:39:23 +00:00
bjornv@webrtc.org
5bde66e913 audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h 
The change of definitions moved to aec_common.h was done in CL17839005.

BUG=3131
TBR=kwiberg@webrtc.org
TESTED=builds locally

Review URL: https://webrtc-codereview.appspot.com/16859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 08:09:50 +00:00
bjornv@webrtc.org
555fc78f27 Neon version of SubbandCoherence() 
The performance gain on a Nexus 7 reported by audioproc is ~1.4%

The output is NOT bit exact.  Any difference seen is +-1.

BUG=3131
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17839005

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 08:03:11 +00:00
bjornv@webrtc.org
ac800c8004 Neon version of rftbsub_128() 
The performance gain on a Nexus 7 reported by audioproc is ~4.5%

The output is bit exact.

BUG=3131
TESTED=trybots and manually
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19919005

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 07:53:13 +00:00
andresp@webrtc.org
5ac876bae0 Revert "Remove remains of WEBRTC_NO_STL." (rev 6641). 
Reason breaks linux_memcheck.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 07:41:59 +00:00
henrikg@webrtc.org
e91ba268e3 Revert 6643 "Revert 6637 "Revert 6636 "Roll chromium_revision 28..." 
Ha. Of course if won't work since a newer revision is required that pulls in the GN fix.

> Revert 6637 "Revert 6636 "Roll chromium_revision 280876:281479""
> 
> GN issue should be fixed in http://crrev.com/282138.
> 
> > Revert 6636 "Roll chromium_revision 280876:281479"
> > 
> > Still breaks GN bot.
> > 
> > > Roll chromium_revision 280876:281479
> > > 
> > > No significant DEPS changes in this roll, only some changes in how clang_format is downloaded. clang_format changes based on https://webrtc-codereview.appspot.com/20829004 which was reverted.
> > > 
> > > Review URL: https://webrtc-codereview.appspot.com/19929004
> > 
> > TBR=henrikg@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/14909004
> 
> TBR=henrikg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/20899004

TBR=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6644 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 07:08:32 +00:00
henrikg@webrtc.org
02dce515d3 Revert 6637 "Revert 6636 "Roll chromium_revision 280876:281479"" 
GN issue should be fixed in http://crrev.com/282138.

> Revert 6636 "Roll chromium_revision 280876:281479"
> 
> Still breaks GN bot.
> 
> > Roll chromium_revision 280876:281479
> > 
> > No significant DEPS changes in this roll, only some changes in how clang_format is downloaded. clang_format changes based on https://webrtc-codereview.appspot.com/20829004 which was reverted.
> > 
> > Review URL: https://webrtc-codereview.appspot.com/19929004
> 
> TBR=henrikg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/14909004

TBR=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 06:56:45 +00:00
buildbot@webrtc.org
72670206db (Auto)update libjingle 70813271-> 70818369 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6642 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 20:40:58 +00:00
andresp@webrtc.org
47d1c98a4e Remove remains of WEBRTC_NO_STL. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 20:18:28 +00:00
jiayl@webrtc.org
10ef8fe611 Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault. 
BUG=crbug/385294
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 19:41:32 +00:00
jiayl@webrtc.org
4b1f330b4f Fix a bug in SocketAddress where "a.b.c.d:1" and "b.b.c.d:1" are incorrectly considered equal. 
BUG=3558
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 19:14:24 +00:00
stefan@webrtc.org
7af12be781 Thread annotations for vie_encoder.cc/.h 
Review URL: https://webrtc-codereview.appspot.com/8739005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 14:46:31 +00:00
henrikg@webrtc.org
e7771d07c8 Revert 6636 "Roll chromium_revision 280876:281479" 
Still breaks GN bot.

> Roll chromium_revision 280876:281479
> 
> No significant DEPS changes in this roll, only some changes in how clang_format is downloaded. clang_format changes based on https://webrtc-codereview.appspot.com/20829004 which was reverted.
> 
> Review URL: https://webrtc-codereview.appspot.com/19929004

TBR=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6637 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 13:15:50 +00:00
henrikg@webrtc.org
543da997f2 Roll chromium_revision 280876:281479 
No significant DEPS changes in this roll, only some changes in how clang_format is downloaded. clang_format changes based on https://webrtc-codereview.appspot.com/20829004 which was reverted.

Review URL: https://webrtc-codereview.appspot.com/19929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 13:03:39 +00:00
andresp@webrtc.org
045a9b17da Remove unnecessary race suppressions copied from chromium. 
And added suppressions to allow to run tests with gtest_parallel in which case some new races were showing up.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 11:44:34 +00:00
stefan@webrtc.org
b8e9e44eac Add full stack test cases with a fake network pipe. 
R=pbos@webrtc.org
BUG=1872

Review URL: https://webrtc-codereview.appspot.com/20889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 11:29:06 +00:00
tommi@webrtc.org
e9cefdef68 Improve libjingle's ASSERT and VERIFY macros on Windows. 
This change has the effect that when using a debugger, a failing ASSERT/VERIFY will break exactly where the failing expression is and not two callstacks up.
Minidumps (for debug builds) will also have the failing expression at the top of the call stack.

R=xians@webrtc.org, xians

Review URL: https://webrtc-codereview.appspot.com/12929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 08:04:12 +00:00
xians@webrtc.org
01bda2068b Fixed the stats problem when new track is using the same ssrc as the previous track. 
Before this patch, when switching from voice mode to stereo mode, the stats won't be updated because StatsCollector binded the ssrc report with the old track, so the report can't be updated by the new track.
This patch fixes the porblem by changing the ssrc report track id to use the new track id.

TEST=libjingle_peerconnection_unittest --gtest_filter="*StatsCollectorTest*"
R=hta@chromium.org

Review URL: https://webrtc-codereview.appspot.com/17859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 07:38:38 +00:00
bjornv@webrtc.org
b753762ce6 delay_estimator: Increases test coverage and makes input spectrum const 
Noticed lack in tests verifying initial state is not left if we have zero input spectra. This CL adds such a test and change input spectra to const at affected places.

BUG=N/A
TESTED=trybots and manually
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 06:40:09 +00:00
jiayl@webrtc.org
12b4efefdd Implement a work around for Chrome full-screen tab switch on Mac. 
Chrome creates a new window in full-screen and minimizes the old window when a tab is switched to full-screen.
We try to find the new window to continue capturing for window sharing.

BUG=crbug/385294
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/19839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 22:05:24 +00:00
bjornv@webrtc.org
e55641d4f7 Neon version of rftfsub_128() 
The performance gain on a Nexus 7 reported by audioproc is ~3.3%

The output is bit exact.

BUG=3131
TESTED=trybots and manually on N7
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14819004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 21:12:23 +00:00
buildbot@webrtc.org
55535d4e58 (Auto)update libjingle 70711261-> 70733822 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6627 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 18:18:55 +00:00
andresp@webrtc.org
d11bec40b2 Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 14:32:58 +00:00
stefan@webrtc.org
3d7da88e06 Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. 
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 13:59:46 +00:00
tommi@webrtc.org
ecb8723402 Change Timing::WallTimeNow to be static. 
There's no need to construct a Timing object to call this method.
On Windows we were unnecessarily calling CreateWaitableTimer + CloseHandle but never actually using that waitable timer.

There's otherwise no change in functionality.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:48:29 +00:00
pbos@webrtc.org
62bafae661 Some refactoring inside rtp_rtcp/. 
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:10:51 +00:00
phoglund@webrtc.org
241a9b0b65 Fixing compile error. 
Made a mistake in https://webrtc-codereview.appspot.com/13849004/,
fixing that here.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6622 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 11:48:37 +00:00
phoglund@webrtc.org
22292df53b Adding explicit check for using dummy file devices. 
Calling into the file device factory without being compiled with file
devices makes no sense and would cause hard-to-debug errors. Therefore
I'm adding an explicit check so this isn't allowed.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 11:39:19 +00:00
andresp@webrtc.org
33d110d8ea Tight data race suppressions around thread_posix. 
BUG=3372,3549
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 10:36:39 +00:00
pbos@webrtc.org
af38f4e511 Extract RTP-header SSRC inline in Call. 
Prevents unknown-RTP-header-extension warnings to be flooding from the
RTP-header parsing as there's no way to register RTP extensions for the
parser in Call as they're allowed to differ between RTP streams.

RTP-header parsing should instead be done separately in every
VideoReceiveStream.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 07:38:12 +00:00
mallinath@webrtc.org
a70be68f65 Disabling shared socket mode for TURN ports. This is done as currently when 
TURN server also used as STUN server, binding responses will be handed over
to TURN port, which simply discard these messages, as requests are originated
from StunPort.

Until we find the right solution for this problem, it's better we disable this
feature.

BUG=https://code.google.com/p/webrtc/issues/detail?id=3537
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6618 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 20:47:24 +00:00
andresp@webrtc.org
3c637cdaa5 Clean data races from system_wrappers_unittests. 
- Remove unittest_utilities that are not used.
 - Remove SetLevelFilter that does not seems necessary and anyhow was racy.

BUG=3549
R=henrike@webrtc.org, henrike

Review URL: https://webrtc-codereview.appspot.com/16819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 20:37:39 +00:00
andresp@webrtc.org
285e9bc84d Fix potential deadlock in webrtc/system_wrappers/source/logging_unittest.cc. 
crit_ should not be held while calling Trace.

BUG=3003
R=henrike@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 20:27:33 +00:00
henrike@webrtc.org
5f2c81c17f webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition. 
BUG=3379
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 17:42:45 +00:00
henrike@webrtc.org
ba93f9a986 drmemory flaky: EndToEndTest.RestartingSendStreamPreservesRtpState[WithRtx] suppressed on drMemory. 
BUG=3552
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 16:52:19 +00:00
pbos@webrtc.org
161f808500 Add test for VideoEncoder setup/teardown. 
Verifies that InitEncode and RegisterEncodeCompleteCallback gets
called before Encode is called. Also verifies that teardown is correctly
done during DestroyVideoSendStream().

BUG=2339
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 14:22:35 +00:00
pbos@webrtc.org
2bb1bdab8d Preserve RTP states for restarted VideoSendStreams. 
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.

Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 13:06:48 +00:00
stefan@webrtc.org
73823cafa4 Add initial gn build files for video_coding and video_processing. 
BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 11:46:43 +00:00
pbos@webrtc.org
03c817e405 Fix pacer to accept duplicate sequence numbers on different SSRCs. 
BUG=3550
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 10:20:35 +00:00
andresp@webrtc.org
b941fe8098 Fix data races related with traces in bitrate estimator test. 
BUG=3549
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 08:50:48 +00:00
pbos@webrtc.org
bd249bc711 Remove GetDefaultConfigs() from Call. 
Defaults for configs are instead placed in the Config constructors.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 04:45:15 +00:00