Log - 80e12072cf9667ca25d9edf3b9e8d926dbe2f50c - platform/external/webrtc

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
62a5ccd Update bitrate only when we have incoming packet. by Stefan Holmer · 9 years ago
58cf5f1 Changed order of events when synthesizing a call. by peah · 9 years ago
0453ef8 Prevent busy-looping PacedSender on small packets. by Peter Boström · 9 years ago
1794b26 Extract ViESyncModule outside ViEChannel. by Peter Boström · 9 years ago
a3dc79e Move SSLIdentity Generate() implementations from .h to .cc file. by Torbjorn Granlund · 9 years ago
71e92dc Avoid overflow in WebRtcSpl_Sqrt by henrik.lundin · 9 years ago
092c951 Roll chromium_revision aefd358..1d144ca (375443:375480) by kjellander · 9 years ago
e8dc081 Implement certificate lifetime parameter as required by WebRTC RFC. by torbjorng · 9 years ago
b1ae3a4 Stop decoders in VideoReceiveStream destructor. by Peter Boström · 9 years ago
461121c Replaced eglbase_jni with with holding a EglBase in PeerConnectionFactory. by perkj · 9 years ago
8259c2d Roll chromium_revision 8d1f312..aefd358 (375401:375443) by kjellander · 9 years ago
8110482 Rename gtest_exclude for rtc_pc_unittests by kjellander@webrtc.org · 9 years ago
56e6269 Rename gtest_exclude for rtc_media_unittests. by Peter Boström · 9 years ago
88c52a7 Disable VerifyHistogramStatsWithRed on DrMemory. by Peter Boström · 9 years ago
16c5a96 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/ by kwiberg · 9 years ago
3959397 Reland of Don't send FEC for H.264 with NACK enabled. (patchset #1 id:1 of https://codereview.webrtc.org/1692783005/ ) by Peter Boström · 9 years ago
cde5d6b removed five redundant avsync tests to make webrtc_perf_test faster by Danil Chapovalov · 9 years ago
e829f58 Rename libjingle_p2p_unittest -> rtc_pc_unittests by kjellander@webrtc.org · 9 years ago