Log - 12f4cda0867b7264f0ad65b2b00e592de9a2a753 - platform/external/webrtc

12f4cda Histograms for H264EncoderImpl/H264DecoderImpl initialization and errors. by hbos · 9 years ago
0ab8e81 Move histograms for rtp receive counters to ReceiveStatisticsProxy by sprang · 9 years ago
b7261fd iSAC float: Check for end of input buffer while decoding by kwiberg · 9 years ago
b01c781 Added functional variants of Buffer::SetData and Buffer::AppendData. by ossu · 9 years ago
17849fc Reland of Add tools/mb to setup_links.py (patchset #1 id:1 of https://codereview.webrtc.org/1691723003/ ) by kjellander · 9 years ago
3e8043b Roll chromium_revision 4d6ba6e..b97dbeb (376909:376966) by kjellander · 9 years ago
1cfe940 Added the agc digital_agc.c file to the ubsan blacklist by peah · 9 years ago
f75d008 Bitrate controller for VideoToolbox encoder. by tkchin · 9 years ago
0ed85b2 Track pending ICE restarts independently for different media sections. by deadbeef · 9 years ago
8df5d4f Moved the AEC C code to be built using C++. by peah · 9 years ago
e80f9d0 Revert of Replace scoped_ptr with unique_ptr in webrtc/common_audio/ (patchset #4 id:60001 of https://codereview.webrtc.org/1712513002/ ) by kjellander · 9 years ago
9788534 Removing some redundant ostringstreams declarations. by Taylor Brandstetter · 9 years ago
71d9721 iOS: Fix JSON for tryserver configurations. by kjellander@webrtc.org · 9 years ago
fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
f4d8441 Disabled flaky tests by philipel · 9 years ago
77f3e0d Screen was flickering when the picker for desktop medias showed up in Windows platform. Keeping track of window size for each window so that BitBlt() instead of PrintWindow() will be called for windows with unchanged sizes. by gyzhou · 9 years ago
b1eaa8d Only average positive quality stats. by Peter Boström · 9 years ago
b68e02f Revert of CQ: Disable linux_baremetal pending installation fix. (patchset #1 id:1 of https://codereview.webrtc.org/1710363002/ ) by kjellander · 9 years ago
80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
ba3e25e Simple RTCP receiver fuzzer. by Peter Boström · 9 years ago
79d7a49 Replace scoped_ptr with unique_ptr in webrtc/common_audio/ by kwiberg · 9 years ago
0be9df4 Roll chromium_revision aa04eb9..4d6ba6e (376768:376909) by Henrik Kjellander · 9 years ago
dc0e381 Add more camera resolutions to camera scaling slider. by Alex Glaznev · 9 years ago
18fcbcf Use VAD to get a better speech power estimation in the IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
67b81f9 Tune QP thresholds for HW H.264 encoder. by Alex Glaznev · 9 years ago
18f9ddd Roll chromium_revision 14bbbf2..aa04eb9 (376710:376768) by kjellander · 9 years ago
a094fd1 RTT intermediate calculation use ntp time instead of milliseconds. by Danil Chapovalov · 9 years ago
723ead8 Move simple RtpRtcp calls to VideoSendStream. by Peter Boström · 9 years ago
2e67ae1 Roll chromium_revision fbc4ecf..14bbbf2 (376680:376710) by kjellander · 9 years ago
eee7d9e iOS: Promote iOS simulator testing to main waterfall. by kjellander@webrtc.org · 9 years ago
7ddc9de Reduce the scope of rtc::Event::Wait() locking. by Peter Boström · 9 years ago
d1f718b Changes in the wav_file implementation in order to by peah · 9 years ago
253d8fa Simplified the function for detecting whether capture data is modified. by peah · 9 years ago
ada8fe5 iOS: Don't run modules_unittests on iOS simulator by kjellander@webrtc.org · 9 years ago
0077272 Roll chromium_revision 4101b15..fbc4ecf (376664:376680) by kjellander · 9 years ago
da1d656 Roll chromium_revision 789f25d..4101b15 (376663:376664) by kjellander · 9 years ago
b9f943d Roll chromium_revision 1120bd3..789f25d (376660:376663) by kjellander · 9 years ago
a18f638 Include "sharedexclusivelock.cc" in Chromium GN build. by jbauch · 9 years ago
fa830dc Roll chromium_revision 9dc1788..1120bd3 (376655:376660) by kjellander · 9 years ago
bf81175 Roll chromium_revision 5618e25..9dc1788 (376642:376655) by kjellander · 9 years ago
330d3d8 Roll chromium_revision fa5d546..5618e25 (376142:376642) by kjellander · 9 years ago
b9dd7c5 Remove GetTransport() from TransportChannelImpl by mikescarlett · 9 years ago
9bf5cde Update build_ios_libs.sh script to build new Objective-C API and gather header files. by hjon · 9 years ago
91fe304 vp9: Adjust parameter for a test in videoprocessor_integrationtest.cc by Marco · 9 years ago
a9d0892 Add initial bitrate and frame resolution parameters to quality scaler. by Alex Glaznev · 9 years ago
0013dcc Simplify SSRC usage inside ViEEncoder. by Peter Boström · 9 years ago
7254890 Nuke SetSenderBufferingMode. by Peter Boström · 9 years ago
da9ae0c Revert of CQ: Change Android trybots to not run device tests. (patchset #1 id:1 of https://codereview.webrtc.org/1715643002/ ) by kjellander · 9 years ago
e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
45c44f0 Simplify EncoderStateFeedback. by Peter Boström · 9 years ago
9674d7c Revert of Prevent data race in MessageQueue. (patchset #3 id:40001 of https://codereview.webrtc.org/1675923002/ ) by jbauch · 9 years ago
fc968a2 Fix sequence-number replay race for padding. by Peter Boström · 9 years ago
88788ad Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/ by kwiberg · 9 years ago
df88460 Prevent data race in MessageQueue. by jbauch · 9 years ago
1e80ce4 webrtc::RtpPacket name freed for better RtpPacket by Danil Chapovalov · 9 years ago
c51d694 CQ: Disable linux_baremetal pending installation fix. by kjellander@webrtc.org · 9 years ago
728012e Changed the semantics of Buffer::Clear to not alter the capacity by ossu · 9 years ago
ecdeb4c CQ: Change Android trybots to not run device tests. by kjellander@webrtc.org · 9 years ago
c4e3ead Blacklist "build/c++11" cpplint filter. by jbauch · 9 years ago
4458d09 Drop support for playing output through aplay in intelligibility_proc by Alejandro Luebs · 9 years ago
b3fb71c Add RTCAudioSession proxy class. by Zeke Chin · 9 years ago
9ac4df1 iOS: Enable modules_unittests and common_audio_unittests by kjellander · 9 years ago
235aaa7 Fix Linux 32-bit compilation after sysroot switch. by Henrik Kjellander · 9 years ago
66a9928 Roll chromium_revision 1d144ca..fa5d546 (375480:376142) by kjellander@webrtc.org · 9 years ago
0e2e50c Always append the BYE packet type at the end by aleungbroadsoft · 9 years ago
452df1c Suppress UBSan errors in common_audio by henrik.lundin · 9 years ago
f45381e VideoCapturerAndroid: Report onFirstFrameAvailable() for textures as well by Magnus Jedvert · 9 years ago
5199c74 AndroidVideoCapturer getSupportedFormats(): Change interface from JSON string to List/vector by Magnus Jedvert · 9 years ago
347c0bb Android GLShader: Check return value of glCreateShader() by magjed · 9 years ago
3ee73a5 Make RemoteBitrateEstimator::GetStats() virtual. by Stefan Holmer · 9 years ago
fd22e6c Change PeerConnectionFactory.setVideoHwAccelerationOptions to be able to replace Egl context. by Per · 9 years ago
74db777 Revert of Remove GetTransport() from TransportChannelImpl (patchset #3 id:40001 of https://codereview.webrtc.org/1691673002/ ) by guidou · 9 years ago
59c634b Re-add RemoteBitrateEstimator::GetStats. by Stefan Holmer · 9 years ago
3234819 Fix and simplify the power estimation in the IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
ee18220 Remove GetTransport() from TransportChannelImpl by mikescarlett · 9 years ago
ee75c7a Compile rtc_base_objc for Mac. by tkchin · 9 years ago
e3c6c82 When doing continual gathering, remove the local ports when a corresponding network is dropped. by honghaiz · 9 years ago
a08bb0d Disabled the test EndToEndTest RestartingSendStreamPreservesRtpState due to the test being flaky. by peah · 9 years ago
b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
dabf07f Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/vad/ by kwiberg · 9 years ago
a293ef0 Apply VideoOptions per stream. by nisse · 9 years ago
789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
bad7804 Remove unused VideoSendStream TransportAdapter. by Peter Boström · 9 years ago
62eaacf Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/ by kwiberg · 9 years ago
28c99bc iOS: Include legacy objc API in all.gyp + fix H264 libyuv dependency by kjellander · 9 years ago
4b4dc86 Remove conference_mode flag from AudioOptions and VideoOptions. by nisse · 9 years ago
22785c7 Exclude legacy objc API tests properly. by kjellander · 9 years ago
69e59e6 [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer by danilchap · 9 years ago
67680c1 Ignore padding-only RTX packets in test. by Peter Boström · 9 years ago
a332e2d Added boilerplate code for being able to test the upcoming AEC functionality. by peah · 9 years ago
0206000 iOS: Add resource files for tests and implement OutputPath by kjellander · 9 years ago
85d8bb0 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/transient/ by kwiberg · 9 years ago
9d3584c Implementing unified plan encoding of msid. by deadbeef · 9 years ago
25d6a0f Adding TSan suppressions temporarily to fix some flaky unit tests. by deadbeef · 9 years ago
e1a0c94 Add network cost as part of the connection ranking. by honghaiz · 9 years ago
2c38c20 Fix out-of-buffer write in iLBC by henrik.lundin · 9 years ago
44c65e9 Enable adaptive threshold experiment by default. by Stefan Holmer · 9 years ago
9d0c432 Remove video-codec max bitrate from TMMBN. by Peter Boström · 9 years ago
d20327c Increase the allowed number of probe packets in test to please msan. by Stefan Holmer · 9 years ago
ee31f0a Fix out-of-buffer read in iLBC by henrik.lundin · 9 years ago