Log - 66a99283be575f967bd92144e5161ff42fa948b8 - platform/external/webrtc

a7584c9 iOS: Add mb_type config for GYP builders. by kjellander · 9 years ago
569cf94 PRESUBMIT: Add check for JSON parse errors + fix JSON by kjellander · 9 years ago
8c66a00 Initialize VideoSendStream members in constructor. by Peter Boström · 9 years ago
1e01660 Add support for rtx with h264. by stefan · 9 years ago
07c51e3 Fix two UBSan warnings in NetEq by henrik.lundin · 9 years ago
162c339 Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #14 id:300001 of https://codereview.webrtc.org/1655793003/ ) by perkj · 9 years ago
4d19c5b This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it. by Per · 9 years ago
f9b59c4 Revert of Initial cleanup of cricket::VideoFrame. (patchset #3 id:40001 of https://codereview.webrtc.org/1688643003/ ) by nisse · 9 years ago
4b2a5a8 Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #12 id:260001 of https://codereview.webrtc.org/1655793003/ ) by perkj · 9 years ago
4d575b0 Initial cleanup of cricket::VideoFrame. by nisse · 9 years ago
2f21789 This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it. by perkj · 9 years ago
43fcd57 Roll chromium_revision afdce76..825c18d (374697:374845) by kjellander · 9 years ago
42d8aa7 Fix IntelligibilityEnhancerTest.TestRenderUpdate test by Alejandro Luebs · 9 years ago
4ed324d Roll chromium_revision b17dd83..afdce76 (374637:374697) by kjellander · 9 years ago
c466bad Using the NS noise estimate for the IE by aluebs · 9 years ago
3123cbc Associate "score" attribute with SSIM so that future perf graphs know that an increase in SSIM should be considered an improvement. by tnakamura · 9 years ago
84d1f12 Remove VideoFormat from WebRtcVideoEngine2. by Peter Boström · 9 years ago
9c6a0c7 Added A/V sync tests with drifting clocks. by danilchap · 9 years ago
a75339c Set rtc_prefer_fixed_point to true on arm64 to match GYP by pkotwicz · 9 years ago
541f186 Cleanup temporary files created by tests. by jbauch · 9 years ago
448468d Experimental patch for adapting adaptation to CPU count on Mac. by torbjorng · 9 years ago
b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
e2812e7 Cleanup after talk/media move. by kjellander@webrtc.org · 9 years ago
1685295 Roll chromium_revision 8fff44b..b17dd83 (374588:374637) by kjellander · 9 years ago
abedbbd Suppress UBSan errors from iSAC by henrik.lundin · 9 years ago
c6344aa Remove java_home GYP variable from webrtc/build/common.gypi by kjellander · 9 years ago
1147b75 Moved buffering of farend into the EchoSubtraction method. by peah · 9 years ago
6608d9a NetEq: Fix a negative shift value by henrik.lundin · 9 years ago
15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago
f5368ab Roll chromium_revision a6aefb7..8fff44b (374428:374588) by kjellander · 9 years ago
cc9669c Cleanup shared memory handling in DesktopCapturer interface. by sergeyu · 9 years ago
c815c60 Roll chromium_revision 3a90ecf..a6aefb7 (374096:374428) by kjellander · 9 years ago
fa639f0 Surface the noise estimate of the NS to be used by other components by Alejandro Luebs · 9 years ago
78ddd73 Update path for audioproc_debug proto output. by kjellander · 9 years ago
4bba35f Switch third_party/gflags to use updated GitHub repo. by kjellander · 9 years ago
09fef9e [rtp_rtcp] Added Sender Report Request rtcp packet. by danilchap · 9 years ago
dfb769d Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete by perkj · 9 years ago
0715a83 Avoid OpenH264 encoder bug for #threads > 1 on Mac and Chromium+Sandbox. by hbos · 9 years ago
097d549 Added thread annotations to FifoBuffer. by jbauch · 9 years ago
e594213 Fix div-by-0 in NetEq's StatisticsCalculator by henrik.lundin · 9 years ago
fd2be27 Fuzzer tests for AudioDecoder's DecodeRedundant and IncomingPacket by henrik.lundin · 9 years ago
7ae5e52 Revert of Analyze support in gyp_webrtc (patchset #1 id:1 of https://codereview.webrtc.org/1369683004/ ) by kjellander · 9 years ago
d2a2296 Enable cpplint for webrtc/modules/pacing and fix all uncovered cpplint errors. by jbauch · 9 years ago
cd0e475 Create QuicSession by mikescarlett · 9 years ago
456801d Add perkj+magjed to webrtc/media/OWNERS by kjellander · 9 years ago
c0ae305 Fix null-pointer dereference in RTPSenderVideo. by Peter Boström · 9 years ago
58c664c Clean up of CongestionController. by Stefan Holmer · 9 years ago
d1d66ba Remove ViEChannel calls for VideoReceiveStream. by Peter Boström · 9 years ago
2945153 Roll chromium_revision 8da2495..3a90ecf (374076:374096) by kjellander · 9 years ago
7336eeb [rtp_rtcp] rtcp::Tmmbn cleaned and got Parse function by danilchap · 9 years ago
62756ee Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. by hbos · 9 years ago
47b6263 Remove Java PC support. This cl removes none Android Java support. by perkj · 9 years ago
f6b5509 Fix GYP and GN references that are invalid in Chromium builds. by kjellander · 9 years ago
1afca73 Change to WebRTC license in webrtc/media by kjellander · 9 years ago
66a1401 Roll chromium_revision 3a7cbe0..8da2495 (374049:374076) by kjellander · 9 years ago
a81f6a3 Revert of Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. (patchset #1 id:1 of https://codereview.webrtc.org/1674103002/ ) by hbos · 9 years ago
10b9dd7 Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. by hbos · 9 years ago
c37b59f Roll chromium_revision 9127267..3a7cbe0 (374043:374049) by kjellander · 9 years ago
f9f84b2 Roll chromium_revision 70700a1..9127267 (374041:374043) by kjellander · 9 years ago
39be561 Roll chromium_revision f0cfd18..70700a1 (374026:374041) by kjellander · 9 years ago
cdc4451 Roll chromium_revision 3c45587..f0cfd18 (373863:374026) by kjellander · 9 years ago
e796f96 check v4l frame rate capability with bitwise method. by Weiyong Yao · 9 years ago
fd6706a Log Android HW decoder delay time statistics. by glaznev · 9 years ago
1c24a6d Set use_gtk=0 as default for Chromium builds. by kjellander · 9 years ago
210cf01 Roll chromium_revision 6e376b8..3c45587 (373731:373863) by kjellander · 9 years ago
c09525a Revert of Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. (patchset #1 id:1 of https://codereview.webrtc.org/1660403004/ ) by hbos · 9 years ago
50fca62 Remove fake cricket::VideoCapturer devices. by Peter Boström · 9 years ago
7cd94f6 Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. by hbos · 9 years ago
900f975 H264: Improve FFmpeg decoder performance by using I420BufferPool. by hbos · 9 years ago
c6e16e3 Use a delayed encoder in GetStats test. by Peter Boström · 9 years ago
f751bf8 Set VideoReceiveStream members in init list. by Peter Boström · 9 years ago
f174e3a [rtp_rtcp] rtcp::Tmmbr cleaned and got Parse function by danilchap · 9 years ago
48fa271 Made implicit casts in the echo canceller explicit. by peah · 9 years ago
1d04ac6 Untangle ViEChannel and ViEEncoder. by Peter Boström · 9 years ago
e449915 Measure encoding time on encode callbacks. by Peter Boström · 9 years ago
8e8908a Delete FrameInput method and FrameInputWrapper class. by nisse · 9 years ago
25d1f28 Fix race between Thread ctor/dtor and MessageQueueManager registrations. by jbauch · 9 years ago
988d31e Move gtest_prod_util.h out of webrtc/test tree. by kjellander · 9 years ago
a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
a713a40 Roll chromium_revision 4c670a4..6e376b8 (373575:373731) by kjellander · 9 years ago
b647aca Roll chromium_revision fbab684..4c670a4 (373504:373575) by kjellander · 9 years ago
ae95ff3 Add more logging and fix PTS overflow for HW decoder. by glaznev · 9 years ago
a92d6be rtcp::TmmbItem designed to replace RTCPUtility::RTCPPacketRTPFBTMMBRItem (for creating and parsing rtcp TMMBR/TMMBN packets) by danilchap · 9 years ago
20834ca Adds a nullptr check to prevent a rare crash when starting or stopping an RtcEventLog. by ivoc · 9 years ago
15ba624 Revert of Rename iOS test specs to match buildbot names. (patchset #1 id:1 of https://codereview.webrtc.org/1665783002/ ) by kjellander@webrtc.org · 9 years ago
daa672d Roll chromium_revision 28e68f8..fbab684 (373442:373504) by kjellander · 9 years ago
ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
2ddb8bd Avoid undefined behavior in vp8 screenshare_layers by sprang · 9 years ago
08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 9 years ago
8cb910d Delete backwards compatibility cruft from cricket::VideoFrame and VideoSourceInterface. by nisse · 9 years ago
c2148a5 Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics). by asapersson · 9 years ago
9031d63 Remove the network with empty name or NONE connection type from the network list. by honghaiz · 9 years ago
fc5fc1e Roll chromium_revision 609aa24..28e68f8 (373145:373442) by kjellander · 9 years ago
f2a2bf4 Stay writable after partial socket writes. by jbauch · 9 years ago
14d024d Do not notify networkconnect if the connection type is known. by Honghai Zhang · 9 years ago
45b683f Call static method getConnectionType using the class name. by Honghai Zhang · 9 years ago
5c35cf9 Re-enable RestartingSendStreamPreservesRtpState. by danilchap · 9 years ago
cedff02 Remove dead code from WebRtcVideoEngine2. by Peter Boström · 9 years ago
e03ac51 Implement NullVideoDecoder to avoid crash on unsupported decoders. by jbauch · 9 years ago
9dc5928 Ability to disable the effects of |rtc_use_h264| with DisableRtcUseH264. by hbos · 9 years ago