Log - 804e0822fca2bd8d017cadb45a8ca394a7be3206 - platform/external/webrtc

804e082 CQ: Add android_compile_mips_dbg to default trybots by kjellander@webrtc.org · 9 years ago
250fc65 Lazily allocate output buffer for AsyncTCPSocket. by jbauch · 9 years ago
2b31a90 Roll chromium_revision a5a1e78..ee31124 (378154:378158) by kjellander · 9 years ago
3379edc Roll chromium_revision ba93bd2..a5a1e78 (378149:378154) by kjellander · 9 years ago
73eb679 Roll chromium_revision fd4c78b..ba93bd2 (378132:378149) by kjellander · 9 years ago
215f228 Roll chromium_revision b815f03..fd4c78b (378096:378132) by kjellander · 9 years ago
7b9601e Roll chromium_revision 968c1e7..b815f03 (377935:378096) by kjellander · 9 years ago
0a00759 Fix the stereo support in IntelligibilityEnhancer by aluebs · 9 years ago
7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
f0fcbf3 Roll chromium_revision 405f35f..968c1e7 (377868:377935) by kjellander · 9 years ago
cedddbd Android MediaCodecVideoDecoder: Limit measured decode time to 200ms by magjed · 9 years ago
3c16576 Don't allocate buffers for listening sockets. by jbauch · 9 years ago
9e69dfd Java SurfaceTextureHelper: Remove support for external thread by magjed · 9 years ago
54ebfca Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ ) by kjellander · 9 years ago
f8136ba Remove add/removal of RTP modules in PacketRouter. by Peter Boström · 9 years ago
8b79b07 Move RTP module activation into PayloadRouter. by Peter Boström · 9 years ago
9c01725 Simplify registration of RTP-header extensions. by Peter Boström · 9 years ago
ff474da Roll chromium_revision 38664e7..405f35f (377790:377868) by kjellander · 9 years ago
d6f6743 CQ: Add linux_ubsan to default trybots. by kjellander@webrtc.org · 9 years ago
2080196 Cleanup of webrtc::VideoFrame. by nisse · 9 years ago
c63f79a Fix ubsan warning in byteio_unittest by sprang · 9 years ago
e31dc95 Make pbos owner of additional video files. by nisse · 9 years ago
10cd6ff Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac by kjellander · 9 years ago
686a8ef Replace scoped_ptr with unique_ptr in webrtc/media/ by kwiberg · 9 years ago
029e220 Removes use of DeRegister Rtp Header Extension for video by Danil Chapovalov · 9 years ago
74622e0 Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ ) by perkj · 9 years ago
8067068 iSAC entropy coder: Avoid signed integer overflow by kwiberg · 9 years ago
db25d2e Make VideoTrack and VideoTrackRenderers implement rtc::VideoSourceInterface. by nisse · 9 years ago
fc59c44 Fix lowPowerModeEnabled crash on iOS8 by tkchin · 9 years ago
e9c0cdf Removed unused cricket::VideoCapturer methods: by perkj · 9 years ago
0e40f7c Remove incorrect reinterpret_cast from const. by Peter Boström · 9 years ago
6b03995 Compile rtc_api_objc on Mac. by hjon · 9 years ago
103c534 Roll chromium_revision 1008ac1..7542f07 (377550:377632) by kjellander · 9 years ago
7324eb9 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago
3dd5d1d Remove PacketRouter sender distinction. by Peter Boström · 9 years ago
13041cf Add CopyOnWriteBuffer class by jbauch · 9 years ago
99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
a5d8e4e Build SharedExclusiveLock in Chromium. by Peter Boström · 9 years ago
a2644c0 Disable tests failing under UBSan to enable deployment to main waterfall. by kjellander@webrtc.org · 9 years ago
75869a1 Roll chromium_revision a5f2cf0..1008ac1 (377481:377550) by kjellander · 9 years ago
a26ac92 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ ) by pbos · 9 years ago
da33a8a Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ ) by torbjorng · 9 years ago
91c5b56 Remove DCHECK on duplicate packets in RemoteEstimatorProxy. by Stefan Holmer · 9 years ago
bf66ae6 Remove thread check from PacketRouter::SendFeedback. by Stefan Holmer · 9 years ago
9ccedc3 Reland: Prevent data race in MessageQueue. by jbauch · 9 years ago
0c74ae1 MB: Fix typo in device mixin. by kjellander@webrtc.org · 9 years ago
b210506 Roll chromium_revision b97dbeb..a5f2cf0 (376966:377481) by kjellander · 9 years ago
f99af6b Fix the gain calculation in IntelligibilityEnhancer by aluebs · 9 years ago
6140fcc Move RTCFileLogger to webrtc/base/objc. by Jon Hjelle · 9 years ago
65c8fd7 Remove the 'audioDebugRecording' media constraint and the aec_dump AudioOptions flag. by solenberg · 9 years ago
861dcb7 MB: Add initial mb_config.pyl configuration file. by kjellander@webrtc.org · 9 years ago
4cc9f98 Fix bug 574524: DtlsTransportChannel crashes after SSL closes remotely by guoweis · 9 years ago
615fabb Add looping sound button to AppRTCDemo by Zeke Chin · 9 years ago
f6ff971 Fix division by zero in FindTMMBRBoundingSet by danilchap · 9 years ago
07fb9be Move RTCP histograms from vie_channel to video channel stats proxies. by sprang · 9 years ago
f14c47a Remove ignored return code from modules. by Peter Boström · 9 years ago
985177c Keep disabled RtpRtcp modules registered. by Peter Boström · 9 years ago
c379fcb Break out pacer thread from CongestionController to increase testability. by Stefan Holmer · 9 years ago
23353ab Increase encoder-overuse thresholds for HW. by Peter Boström · 9 years ago
3e60bf0 Adds low complexity audio mode for single core CPUs by henrika · 9 years ago
c2b785d Replace scoped_ptr with unique_ptr in webrtc/common_audio/ by kwiberg · 9 years ago
837b39e Fix ubsan warnings in BWE tests. by Stefan Holmer · 9 years ago
f01633e Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/ by kwiberg · 9 years ago
58e08cb Reset indexer upon initialization in AudioLoop. by minyue · 9 years ago
2346c5a Add tools/mb to .gitignore by Henrik Kjellander · 9 years ago
0665f05 Fix OOB read in pacing test. by stefan · 9 years ago
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