Log - eb45981165f982dd51425fad5ecb7ea9619063d3 - platform/external/webrtc

eb45981 Restoring behavior where PeerConnection tracks changes to MediaStreams. by deadbeef · 9 years ago
44f0819 Fixing bug where "mid" wasn't preserved across re-offers. by deadbeef · 9 years ago
c1316a1 Fix HPF initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1490333004/. by solenberg · 9 years ago
95d9851 Add speech encoder to the encoder stack specification struct by kwiberg · 9 years ago
7eb914d Fix incorrect comment by kwiberg · 9 years ago
78315b9 Reland of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1528043002/ ) by Peter Boström · 9 years ago
f9945b2 Only try to pair protocol matching candidates for creating connections. by Honghai Zhang · 9 years ago
949028f Make LevelEstimation not a ProcessingComponent. by solenberg · 9 years ago
5e0218c Revert of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1524993002/ ) by tommi · 9 years ago
5125433 Android: Refactor renderers to allow apps to inject custom shaders by Magnus Jedvert · 9 years ago
91941ae rtcp::VoipMetric block moved into own file and got Parse function by danilchap · 9 years ago
32d989b Disable transport sequence numbers for audio. by Stefan Holmer · 9 years ago
10aea22 Roll chromium_revision 53970fd..bff4606 (365141:365226) by kjellander · 9 years ago
377b5e6 enabled cpplint for the webrtc/modules/rtp_rtcp directory by danilchap · 9 years ago
6eca7e3 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( by tommi · 9 years ago
6db6cdc [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs by danilchap · 9 years ago
9638143 Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ ) by perkj · 9 years ago
e005cf2 [rtp_rtcp] SSRCDatabase class cleaned (including all lint errors) by danilchap · 9 years ago
5ea3da2 Base webrtc fuzzers on a template. by Peter Boström · 9 years ago
8f09f17 Simple CL to fix lint errors in webrtc/modules/remote_bitrate_estimator. Added the lint check for the folder to the presubmit script. by terelius · 9 years ago
498ae00 Disable ThreadTest.ThreeThreadsInvoke on DrMemory bots. by Stefan Holmer · 9 years ago
47a740b [rtp_rtcp] lint errors about rand() usage fixed. by danilchap · 9 years ago
2d36b92 Roll chromium_revision 10bf0e1..53970fd (365000:365141) by kjellander · 9 years ago
1588793 Fixing flaky LocalP2PTestSctpDataChannel test. by deadbeef · 9 years ago
c9be007 Fixing and re-enabling some flaky PeerConnection tests. by deadbeef · 9 years ago
bd29246 Reland of Free SCTP data channels asynchronously in PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1513143003/ ) by deadbeef · 9 years ago
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