Log - 66a99283be575f967bd92144e5161ff42fa948b8 - platform/external/webrtc

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
3747838 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/test/ by kwiberg · 9 years ago
1d9ada7 Roll chromium_revision 99c33e8..8d1f312 (375382:375401) by kjellander · 9 years ago
2d0c332 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/ by kwiberg · 9 years ago
04af839 Move refcount.h and scoped_ref_ptr.h to rtc_base_approved. BUG= by tommi · 9 years ago
290ab41 Roll chromium_revision ee3223b..99c33e8 (375377:375382) by kjellander · 9 years ago
91d9756 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/ by kwiberg · 9 years ago
3f5f5aa Roll chromium_revision e1bc525..ee3223b (375361:375377) by kjellander · 9 years ago
be61562 Moved the GainControlForNewAGC class to be a separate file. by peah · 9 years ago
46e2cb8 Roll chromium_revision 96c72eb..e1bc525 (375334:375361) by kjellander · 9 years ago
88ec91d Roll chromium_revision c9db86b..96c72eb (374913:375334) by kjellander · 9 years ago
88b0a22 Add VP9 to full stack tests. by asapersson · 9 years ago
29ffdc1 Revert of Don't send FEC for H.264 with NACK enabled. (patchset #5 id:80001 of https://codereview.webrtc.org/1687303002/ ) by deadbeef · 9 years ago
5e7834e Android: Make VideoCapturer an interface for all VideoCapturers to implement by Magnus Jedvert · 9 years ago
e78765b Removes Nexus 5 from AEC and NS blacklists by henrika · 9 years ago
579e832 Fix race on VCM protection callback. by Peter Boström · 9 years ago
b72dada Remove Reset from conditionally-compiled decoders. by Peter Boström · 9 years ago
c90e9b6 Remove no-op VideoDecoder::Reset implementation. by Peter Boström · 9 years ago
25558ad Don't send FEC for H.264 with NACK enabled. by Peter Boström · 9 years ago
efce73e Rename libjingle_media_unittest -> rtc_media_unittests by kjellander@webrtc.org · 9 years ago
e9de296 Add OWNERS file in webrtc/pc by kjellander@webrtc.org · 9 years ago
040f68e Fix VideoCapturer::OnMessage override by Per · 9 years ago
a509241 This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting. by Per · 9 years ago
59013bc Remove spammy GetRTPStatistics() log. by Peter Boström · 9 years ago
51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago
d73c99c Initial cleanup of cricket::VideoFrame. by nisse · 9 years ago
65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
5ad1297 Rename webrtc/media/webrtc -> webrtc/media/engine by kjellander@webrtc.org · 9 years ago
f396f60 Update API for Objective-C RTCPeerConnection. by hjon · 9 years ago
404686a Use std::vector in the PayloadRouter interface. by Peter Boström · 9 years ago
8fb3557 rtc::Buffer: Replace an internal rtc::scoped_ptr with std::unique_ptr by kwiberg · 9 years ago
9b7a289 Roll chromium_revision 8974515..c9db86b (374881:374913) by kjellander · 9 years ago
af71655 Revert of Android: Remove VideoCapturer (patchset #2 id:20001 of https://codereview.webrtc.org/1684403002/ ) by kjellander · 9 years ago
73e2373 Fix negative shift exponent in WPDTree by aluebs · 9 years ago
66d2481 Fix division by zero errors in IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
6ecee07 Fixing bug in MediaStream.java that caused double disposal of track. by deadbeef · 9 years ago
09eab31 Android: Remove VideoCapturer by Magnus Jedvert · 9 years ago
9f35d55 Added accessor and Parse function. Create function merged into one. by danilchap · 9 years ago
ca83525 Move the decoder thread into VideoReceiveStream. by Peter Boström · 9 years ago
8e16e61 Added empty files for VideoBroadcaster so adding them don't break Chrome. by Per · 9 years ago
d3d8b67 iOS: Remove MB and Goma for GYP builders. by kjellander · 9 years ago
67ca0e1 Revert of Add tools/mb to setup_links.py (patchset #1 id:1 of https://codereview.webrtc.org/1692543002/ ) by kjellander · 9 years ago
1f7d77f Extract send-side ViEReceiver calls. by Peter Boström · 9 years ago
68da769 Add tools/mb to setup_links.py by kjellander · 9 years ago
4af8061 Roll chromium_revision 825c18d..8974515 (374845:374881) by kjellander · 9 years ago