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gerrit-public.fairphone.software
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platform
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external
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webrtc
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47fadba7502f852629cb635426047efb797c1e31
47fadba
Add include stdlib.h to files using abs.
by stefan@webrtc.org
· 11 years ago
4ab4fc0
Add test for automatically disabling padding when no video is being captured.
by stefan@webrtc.org
· 11 years ago
b5bc098
Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error.
by fbarchard@google.com
· 11 years ago
aa74b5d
Add success/error callback to set sdp calls.
by wu@webrtc.org
· 11 years ago
5272eb8
Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest.
by turaj@webrtc.org
· 11 years ago
e839da0
Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin.
by sergeyu@chromium.org
· 11 years ago
78b41a0
Fix issues with sequence number wrap-around in jitter statistics.
by turaj@webrtc.org
· 11 years ago
832bd74
libyuv r874 for build improvements on ios/android, and improved YUV scale performance.
by fbarchard@google.com
· 11 years ago
b43202d
Disable PeerConnectionEndToEndTest for tsanv2 build.
by wu@webrtc.org
· 11 years ago
1e8c93c
Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out.
by turaj@webrtc.org
· 11 years ago
2ffb149
Replace VideoFrameI420 with I420VideoFrame.
by pbos@webrtc.org
· 11 years ago
b0ed8f8
Don't reset the AEC filter in extended mode.
by andrew@webrtc.org
· 11 years ago
9e85c01
Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release.
by dwkang@webrtc.org
· 11 years ago
9fe3603
Renaming ViEEncoderObserver::VideoSuspended
by henrik.lundin@webrtc.org
· 11 years ago
484ee96
Protect reads of ViEEncoder::video_suspended_.
by pbos@webrtc.org
· 11 years ago
1977960
AppRTCDemo(ios): remove codesigning hack now that gyp signs by default.
by fischman@webrtc.org
· 11 years ago
ef2d554
Increase size of pacer window to 500 ms as that better matches the encoder.
by stefan@webrtc.org
· 11 years ago
331d440
Connect pacer/padding to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
ffe1b17
Lock access to ModuleRtpRtcpImpl::simulcast_.
by pbos@webrtc.org
· 11 years ago
2c46f8d
Rename DestroyStream methods to include Video.
by pbos@webrtc.org
· 11 years ago
6f6ba6e
Fix issues with sequence number wrap-around in jitter statistics
by henrik.lundin@webrtc.org
· 11 years ago
b3cc78d
Add -Wnon-virtual-dtor warning for C++ code.
by pbos@webrtc.org
· 11 years ago
72964bd
Make interface destructor virtual
by sprang@webrtc.org
· 11 years ago
8d02f5d
Added API for enabling/disabling RTCP Receiver Reference Time extension.
by asapersson@webrtc.org
· 11 years ago
54a0551
Increase run-time for full stack test for the rtt to be added reliably to the delay measurement.
by asapersson@webrtc.org
· 11 years ago
425e1d0
Typo in vie_autotest_win.cc
by braveyao@webrtc.org
· 11 years ago
a750044
Fixes a crash in VoE when unregistering JNI hooks.
by henrike@webrtc.org
· 11 years ago
364f204
Update talk to 56698267.
by wu@webrtc.org
· 11 years ago
dc50aae
Interface changes to old api, for use by new api transition.
by sprang@webrtc.org
· 11 years ago
b24d335
Added ViE API for getting overuse measure.
by asapersson@webrtc.org
· 11 years ago
d29d4e9
Deliver I420VideoFrames from VideoRender module.
by pbos@webrtc.org
· 11 years ago
1ae1d0c
Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).
by asapersson@webrtc.org
· 11 years ago
27326b6
Rename newapi::Transport::SendRTP()->SendRtp().
by pbos@webrtc.org
· 11 years ago
ce90eff
Rename RTP-extension constants.
by pbos@webrtc.org
· 11 years ago
53c8573
Rename video streams' start/stop methods.
by pbos@webrtc.org
· 11 years ago
5a63655
Rename Call::Create{Receive,Send}Stream().
by pbos@webrtc.org
· 11 years ago
0b72f58
Add experimental noise suppression dummy API.
by aluebs@webrtc.org
· 11 years ago
5d85819
Fix DesktopAndCursorComposer to restore frames to the original state.
by sergeyu@chromium.org
· 11 years ago
7a05ae5
Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded.
by turaj@webrtc.org
· 11 years ago
9c5fb76
Exclude AV-sync test from Valgrind platforms.
by pbos@webrtc.org
· 11 years ago
ce8e093
Rename AutoMute to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
28bf50f
Fix test broken with r5128.
by stefan@webrtc.org
· 11 years ago
b082ade
Hook up audio/video sync to Call.
by stefan@webrtc.org
· 11 years ago
4cfa605
Fix breakage after introducing new test.
by stefan@webrtc.org
· 11 years ago
69969e2
Improve Call tests for RTX.
by stefan@webrtc.org
· 11 years ago
6e95d7a
Increment RTP timestamps for padding packets
by henrik.lundin@webrtc.org
· 11 years ago
6488761
Implement VideoSendStream::SetCodec().
by pbos@webrtc.org
· 11 years ago
183c727
Disable datachannel_unittest.cc
by sergeyu@chromium.org
· 11 years ago
a23f0ca
Update talk to 56619788
by sergeyu@chromium.org
· 11 years ago
e872285
Disable all vie_auto_tests on Linux for now (take 2)
by kjellander@webrtc.org
· 11 years ago
c848985
Disable all automated vie_auto_tests on Linux for now
by kjellander@webrtc.org
· 11 years ago
9b82f5a
Fix for RTX in combination with pacing.
by stefan@webrtc.org
· 11 years ago
03f3370
Inject config when creating channels to override the existing one.
by turaj@webrtc.org
· 11 years ago
e8433eb
Reimplementing NetEq4's AudioVector
by henrik.lundin@webrtc.org
· 11 years ago
3859951
Parse next RTCP XR report block after an unsupported block type.
by asapersson@webrtc.org
· 11 years ago
3e42726
Reducing opus_test runtime to pass Android test
by minyue@webrtc.org
· 11 years ago
e03cafa
MIPS optimizations for AECM audio processing module
by andrew@webrtc.org
· 11 years ago
b073010
Move audio_processing dependencies to a variable.
by andrew@webrtc.org
· 11 years ago
57eb858
Remove ".." from include_dirs in build/common.
by pbos@webrtc.org
· 11 years ago
6e908b3
Remove unnecessary include_dirs from audio_processing.
by andrew@webrtc.org
· 11 years ago
00ed170
Roll libvpx 225010:232686.
by marpan@webrtc.org
· 11 years ago
5973f3a
Remove unneeded includes from trace_posix.cc.
by andrew@webrtc.org
· 11 years ago
48df381
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
by stefan@webrtc.org
· 11 years ago
bff9620
Fix log build error for Chromium builds.
by henrikg@webrtc.org
· 11 years ago
4c828e1
Remove update_resources.py as it's no longer used.
by kjellander@webrtc.org
· 11 years ago
f1a4817
Replace disabled logging with a restricted logging mode.
by andrew@webrtc.org
· 11 years ago
5adc897
Updated WebRTC version to 3.46
by elham@webrtc.org
· 11 years ago
a7855a8
Fix for xgetbv on Visual Studio 2010.
by fbarchard@google.com
· 11 years ago
bde3056
Fix for video_processor_intergration_tests to run in parallel.
by marpan@webrtc.org
· 11 years ago
c4225b6
Update getUserMedia W3C conformance tests.
by kjellander@webrtc.org
· 11 years ago
8bad50e
Sending status fix for module.
by asapersson@webrtc.org
· 11 years ago
16d6254
Update talk to 56183333.
by wu@webrtc.org
· 11 years ago
d16d307
Fix bad Google Storage uploads of resource files.
by kjellander@webrtc.org
· 11 years ago
0e03360
Add OWNERS for resources/
by kjellander@webrtc.org
· 11 years ago
7a36cb4
Add missing dependencies to .isolate files
by kjellander@webrtc.org
· 11 years ago
1e8b671
Roll chromium_revision 231713:232627
by kjellander@webrtc.org
· 11 years ago
da7f658
Add svn:ignore to avoid re-download of resources
by kjellander@webrtc.org
· 11 years ago
b8cb85b
Fix broken build on x86 Android
by fischman@webrtc.org
· 11 years ago
7b273a5
PeerConnection iOS: update README instructions
by fischman@webrtc.org
· 11 years ago
07a6fbe
Update talk to 56092586.
by wu@webrtc.org
· 11 years ago
3779c1c
Fix invalid .sha1 files for audio_coding
by kjellander@webrtc.org
· 11 years ago
8017458
Replace old resources download script with depot_tools
by kjellander@webrtc.org
· 11 years ago
a452fc2
Remove resources/ svn:ignore to prepare for updated resource handling
by kjellander@webrtc.org
· 11 years ago
58bcdee
Roll chromium_revision 229708:231713
by kjellander@webrtc.org
· 11 years ago
766154a
Removed unused code.
by asapersson@webrtc.org
· 11 years ago
e2df8b7
Make video quality analysis unittests print to log instead of stdout.
by kjellander@webrtc.org
· 11 years ago
5dd2ecb
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
by sheu@chromium.org
· 11 years ago
74e6e84
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
by sheu@chromium.org
· 11 years ago
d705649
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
by sheu@chromium.org
· 11 years ago
1a4ed0d
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
by sheu@chromium.org
· 11 years ago
de30501
Update talk to 55906045.
by wu@webrtc.org
· 11 years ago
58cd316
Address Clag Analyzer issues.
by turaj@webrtc.org
· 11 years ago
7d6bd22
Propagate estimated RTT from receivers to rtt observer.
by asapersson@webrtc.org
· 11 years ago
da2c37b
Video bandwidth not reported correctly
by sprang@webrtc.org
· 11 years ago
773e727
Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc
by sergeyu@chromium.org
· 11 years ago
de748c8
Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build.
by wu@webrtc.org
· 11 years ago
dce70cc
Add delay limit to ChokeFilter.
by solenberg@webrtc.org
· 11 years ago
f424cb8
Update talk to 55863981.
by wu@webrtc.org
· 11 years ago
d6e4663
Logging for BWE test framework.
by solenberg@webrtc.org
· 11 years ago
cecfd18
Update talk to 55821645.
by wu@webrtc.org
· 11 years ago
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