Log - 82ccfcf5cae798d21881e41a7123e9ca3016988a - platform/external/webrtc

82ccfcf Remove unused and rarely used LOG_ macros. by solenberg · 9 years ago
e22e1cb Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ ) by perkj · 9 years ago
40f349f [rtp_rtcp] Lint errors cleared from rtp_rtcp/test by danilchap · 9 years ago
3207916 Made EglBase an abstract class and cleaned up. by perkj · 9 years ago
03960d9 Roll chromium_revision 4bc4277..10bf0e1 (364953:365000) by kjellander · 9 years ago
bc14164 Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ ) by stefan · 9 years ago
b2f80e3 rtp_rtcp/test/BWEStandAlone deleted as obsolete by danilchap · 9 years ago
a78c021 Add APK targets to build libjingle_peerconnection_unittests for Android. by perkj · 9 years ago
17821db Wire up bandwidth limitation info to GetStats and adapt_reason. by asapersson · 9 years ago
ac921d7 Add "x"s in the end of a stripped IPv6 address string. by henrikg · 9 years ago
38bb8ad Add test for verifying configured key frame interval for VP9. by asapersson · 9 years ago
e5ae6f8 Correcting the check for the return code produced by by peah · 9 years ago
1d5c19d Address comments from code review 1505253004 by tommi · 9 years ago
4759bfb Roll chromium_revision 7de03ed..4bc4277 (364770:364953) by kjellander · 9 years ago
aa32c3e Update API for Objective-C RTCIceServer by hjon · 9 years ago
cb95f54e Remove pointless move() to fix build on clang/win. by Tommi · 9 years ago
66679dc Update WARN_UNUSED_RESULT macro to match Chromium's version. by tfarina · 9 years ago
be26c07 Roll gtest-parallel. by pbos · 9 years ago
b798f38 Roll chromium_revision 710285b..7de03ed (364599:364770) by kjellander · 9 years ago
f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
f67c548 Handle Turn error response to RefreshRequest, CreatePermissionRequest, and ChanelBindRequest by Honghai Zhang · 9 years ago
04e9146 Discard old-generation candidates when ICE restarts by Honghai Zhang · 9 years ago
43e4e23 Remove thread-id wraparounds in event tracing. by Peter Boström · 9 years ago
822bdf9 Remove cricket::VideoEncoderConfig. by Peter Boström · 9 years ago
4c1093b Add FEC producer fuzzing and a unittest for one of the issues found. by Stefan Holmer · 9 years ago
5b659c0 Special-case android-arm64 in codec bitexactness tests by kwiberg · 9 years ago
b562c33 Remove ancient VoE suppressions. by solenberg · 9 years ago
cb23c0d Adding Opus to RTPencode. by minyue · 9 years ago
71f5a9a This cl change VideoCaptureAndroid to handle CVO the same way when capturing to texture as when using ordinary byte buffers. by Per · 9 years ago
0b0a88b Add aecdump support to AppRTCDemo by aluebs · 9 years ago
4dfe332 Roll chromium_revision 026b937..710285b (364421:364599) by kjellander · 9 years ago
55bcf0f Fix -Wformat error in Win-Clang build (take 2) by hans · 9 years ago
013e83b Fix -Wformat error in Win-Clang build by Niklas Enbom · 9 years ago
cf846ad Adding stub files needed for https://codereview.webrtc.org/1507973003/ by Taylor Brandstetter · 9 years ago
7c73bdb Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor. by deadbeef · 9 years ago
ed83edc Roll chromium_revision 2e451bf..026b937 (364330:364421) by kjellander · 9 years ago
6a6f089 in rtp_rtcp module: by danilchap · 9 years ago
a1f567a Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ ) by deadbeef · 9 years ago
61a90f9 clang/win: Fix -Wextra warnings in webrtc. by thakis · 9 years ago
5c1def8 modules/rtp_rtcp/include folder cleared of lint warnings by danilchap · 9 years ago
796cfaf Add VideoCodec::PreferDecodeLate by perkj · 9 years ago
4d68208 Reduce the runtime of some ACM tests in modules_tests by Henrik Lundin · 9 years ago
c490e01 Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to by nisse · 9 years ago
b8b6fbb lint build/include errors fixed in rtp_rtcp module by danilchap · 9 years ago
90b9fc9 Roll chromium_revision a02d286..2e451bf (364268:364330) by kjellander · 9 years ago
866df66 Typo fix: Enable a bunch of tests that were accidentally disabled by kwiberg · 9 years ago
5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
0f2e939 Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors. by jbauch · 9 years ago
162abd3 lint whitespace warning removed from most rtp_rtcp/source/ files by danilchap · 9 years ago
84e78f9 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. by terelius · 9 years ago
0b3d7ee Prevent RTCP SR to be sent with bogus timestamp. by mflodman · 9 years ago
48bf238 Some further minor bitexact APM echo suppressor refactoring by peah · 9 years ago
5ba58c6 Roll chromium_revision dad6346..a02d286 (363782:364268) by kjellander · 9 years ago
a6e4328 Remove unnecessary test code on Windows. by Tommi · 9 years ago
70625e5 Enable cpplint for webrtc/examples and fix all uncovered cpplint errors. by jbauch · 9 years ago
2e5fe31 Remove myself from root_files watchlist. by andrew · 9 years ago
1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
ee40821 WebRTC: Update set of known root certificates by Guo-wei Shieh · 9 years ago
b14f001 Some minor (bitexact) AEC echo suppressor refactoring by peah · 9 years ago
434aca8 Add empty placeholder files for remote audio tracks. by tommi · 9 years ago
afeb438 Moved code into the lowest level of EchoSuppression by peah · 9 years ago
d1590b2 Lint clean video/ and add lint presubmit check. by mflodman · 9 years ago
4cf61dd NetEq: Add codec name and RTP timestamp rate to DecoderInfo by henrik.lundin · 9 years ago
3980d46 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). by hbos · 9 years ago
af3b9cb Removing DrMemory suppresssion on PushResampler. by minyuel · 9 years ago
5eb4988 [rtp_rtcp] Lint build/header_guard errors fixed by danilchap · 9 years ago
7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago
d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
d02b0fa Add oldest rotation type option to RTCFileLogger by haysc · 9 years ago
5e465c3 Make NoiseSuppression not a processing component (bit exact). by solenberg · 9 years ago
1a9d615 Add tracing to public PeerConnection methods. by Peter Boström · 9 years ago
2d63680 Roll chromium_revision 9dfb3a1..dad6346 (363718:363782) by kjellander · 9 years ago
7b2f762 Don't call SetPreviewFormat if capturing to textures. by perkj · 9 years ago
edd8fef Add new view that renders local video using AVCaptureLayerPreview. by haysc · 9 years ago
70f9903 Make HighPassFilter not a ProcessingComponent anymore (bit exact). by solenberg · 9 years ago
246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
e10c82d Deletes temporary files that are generated in several ACM unittests. by ivoc · 9 years ago
d7b7ae8 Add encode/decode time tracing to audio_coding. by Peter Boström · 9 years ago
9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
cd6f539 Revert of RTCCertificate::Expires() and ::HasExpired() implemented (patchset #5 id:140001 of https://codereview.webrtc.org/1494103003/ ) by hbos · 9 years ago
fe32a76 Create fuzzer tests for audio decoders by Henrik Lundin · 9 years ago
ffea13c PRESUBMIT: change native API check from warning to information. by kjellander · 9 years ago
20ef654 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). by hbos · 9 years ago
325b345 There was an old scaling for CNG 48 kHz in the code, from the time where Audio Coding Module didn't have full 48 kHz support. This CL removes the scaling. by Tina le Grand · 9 years ago
88eeac4 Adding video_processing to presubmit lint check by mflodman · 9 years ago
4654d20 Add test which verifies that the RTP header extensions are set correctly for FEC packets. by Stefan Holmer · 9 years ago
03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
99ab944 Clang format of video_processing folder. by mflodman · 9 years ago
a440c6f Roll chromium_revision 3b8be21..9dfb3a1 (363445:363718) by kjellander · 9 years ago
3868692 Free SCTP data channels asynchronously in PeerConnection. by deadbeef · 9 years ago
46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
0f490a5 Add logs when stun or turn host lookup is completed. by Honghai Zhang · 9 years ago
cd4003f Use @webrtc.org addresses for OWNERS. by Peter Boström · 9 years ago
cf890bc Roll gtest-parallel. by Peter Boström · 9 years ago
0608dc5 Roll chromium_revision 4918765..3b8be21 (363393:363445) by kjellander · 9 years ago
5f6deaf Remove unused RTP-header parser. by Peter Boström · 9 years ago