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webrtc / src.git / 38bb8ad1ca93d28d6f0055bb5663dff95d82a1e8
  1. 38bb8ad Add test for verifying configured key frame interval for VP9. by asapersson · 9 years ago
  2. e5ae6f8 Correcting the check for the return code produced by by peah · 9 years ago
  3. 1d5c19d Address comments from code review 1505253004 by tommi · 9 years ago
  4. 4759bfb Roll chromium_revision 7de03ed..4bc4277 (364770:364953) by kjellander · 9 years ago
  5. aa32c3e Update API for Objective-C RTCIceServer by hjon · 9 years ago
  6. cb95f54e Remove pointless move() to fix build on clang/win. by Tommi · 9 years ago
  7. 66679dc Update WARN_UNUSED_RESULT macro to match Chromium's version. by tfarina · 9 years ago
  8. be26c07 Roll gtest-parallel. by pbos · 9 years ago
  9. b798f38 Roll chromium_revision 710285b..7de03ed (364599:364770) by kjellander · 9 years ago
  10. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  11. f67c548 Handle Turn error response to RefreshRequest, CreatePermissionRequest, and ChanelBindRequest by Honghai Zhang · 9 years ago
  12. 04e9146 Discard old-generation candidates when ICE restarts by Honghai Zhang · 9 years ago
  13. 43e4e23 Remove thread-id wraparounds in event tracing. by Peter Boström · 9 years ago
  14. 822bdf9 Remove cricket::VideoEncoderConfig. by Peter Boström · 9 years ago
  15. 4c1093b Add FEC producer fuzzing and a unittest for one of the issues found. by Stefan Holmer · 9 years ago
  16. 5b659c0 Special-case android-arm64 in codec bitexactness tests by kwiberg · 9 years ago
  17. b562c33 Remove ancient VoE suppressions. by solenberg · 9 years ago
  18. cb23c0d Adding Opus to RTPencode. by minyue · 9 years ago
  19. 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
  20. 0b0a88b Add aecdump support to AppRTCDemo by aluebs · 9 years ago
  21. 4dfe332 Roll chromium_revision 026b937..710285b (364421:364599) by kjellander · 9 years ago
  22. 55bcf0f Fix -Wformat error in Win-Clang build (take 2) by hans · 9 years ago
  23. 013e83b Fix -Wformat error in Win-Clang build by Niklas Enbom · 9 years ago
  24. cf846ad Adding stub files needed for https://codereview.webrtc.org/1507973003/ by Taylor Brandstetter · 9 years ago
  25. 7c73bdb Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor. by deadbeef · 9 years ago
  26. ed83edc Roll chromium_revision 2e451bf..026b937 (364330:364421) by kjellander · 9 years ago
  27. 6a6f089 in rtp_rtcp module: by danilchap · 9 years ago
  28. 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
  29. 61a90f9 clang/win: Fix -Wextra warnings in webrtc. by thakis · 9 years ago
  30. 5c1def8 modules/rtp_rtcp/include folder cleared of lint warnings by danilchap · 9 years ago
  31. 796cfaf Add VideoCodec::PreferDecodeLate by perkj · 9 years ago
  32. 4d68208 Reduce the runtime of some ACM tests in modules_tests by Henrik Lundin · 9 years ago
  33. c490e01 Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to by nisse · 9 years ago
  34. b8b6fbb lint build/include errors fixed in rtp_rtcp module by danilchap · 9 years ago
  35. 90b9fc9 Roll chromium_revision a02d286..2e451bf (364268:364330) by kjellander · 9 years ago
  36. 866df66 Typo fix: Enable a bunch of tests that were accidentally disabled by kwiberg · 9 years ago
  37. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  38. 0f2e939 Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors. by jbauch · 9 years ago
  39. 162abd3 lint whitespace warning removed from most rtp_rtcp/source/ files by danilchap · 9 years ago
  40. 84e78f9 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. by terelius · 9 years ago
  41. 0b3d7ee Prevent RTCP SR to be sent with bogus timestamp. by mflodman · 9 years ago
  42. 48bf238 Some further minor bitexact APM echo suppressor refactoring by peah · 9 years ago
  43. 5ba58c6 Roll chromium_revision dad6346..a02d286 (363782:364268) by kjellander · 9 years ago
  44. a6e4328 Remove unnecessary test code on Windows. by Tommi · 9 years ago
  45. 70625e5 Enable cpplint for webrtc/examples and fix all uncovered cpplint errors. by jbauch · 9 years ago
  46. 2e5fe31 Remove myself from root_files watchlist. by andrew · 9 years ago
  47. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  48. ee40821 WebRTC: Update set of known root certificates by Guo-wei Shieh · 9 years ago
  49. b14f001 Some minor (bitexact) AEC echo suppressor refactoring by peah · 9 years ago
  50. 434aca8 Add empty placeholder files for remote audio tracks. by tommi · 9 years ago
  51. afeb438 Moved code into the lowest level of EchoSuppression by peah · 9 years ago
  52. d1590b2 Lint clean video/ and add lint presubmit check. by mflodman · 9 years ago
  53. 4cf61dd NetEq: Add codec name and RTP timestamp rate to DecoderInfo by henrik.lundin · 9 years ago
  54. 3980d46 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). by hbos · 9 years ago
  55. af3b9cb Removing DrMemory suppresssion on PushResampler. by minyuel · 9 years ago
  56. 5eb4988 [rtp_rtcp] Lint build/header_guard errors fixed by danilchap · 9 years ago
  57. 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
  58. d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  59. bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
  60. d02b0fa Add oldest rotation type option to RTCFileLogger by haysc · 9 years ago
  61. 5e465c3 Make NoiseSuppression not a processing component (bit exact). by solenberg · 9 years ago
  62. 1a9d615 Add tracing to public PeerConnection methods. by Peter Boström · 9 years ago
  63. 2d63680 Roll chromium_revision 9dfb3a1..dad6346 (363718:363782) by kjellander · 9 years ago
  64. 7b2f762 Don't call SetPreviewFormat if capturing to textures. by perkj · 9 years ago
  65. edd8fef Add new view that renders local video using AVCaptureLayerPreview. by haysc · 9 years ago
  66. 70f9903 Make HighPassFilter not a ProcessingComponent anymore (bit exact). by solenberg · 9 years ago
  67. 246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
  68. 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
  69. e10c82d Deletes temporary files that are generated in several ACM unittests. by ivoc · 9 years ago
  70. d7b7ae8 Add encode/decode time tracing to audio_coding. by Peter Boström · 9 years ago
  71. 9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
  72. cd6f539 Revert of RTCCertificate::Expires() and ::HasExpired() implemented (patchset #5 id:140001 of https://codereview.webrtc.org/1494103003/ ) by hbos · 9 years ago
  73. fe32a76 Create fuzzer tests for audio decoders by Henrik Lundin · 9 years ago
  74. ffea13c PRESUBMIT: change native API check from warning to information. by kjellander · 9 years ago
  75. 20ef654 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). by hbos · 9 years ago
  76. 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
  77. 88eeac4 Adding video_processing to presubmit lint check by mflodman · 9 years ago
  78. 4654d20 Add test which verifies that the RTP header extensions are set correctly for FEC packets. by Stefan Holmer · 9 years ago
  79. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  80. 99ab944 Clang format of video_processing folder. by mflodman · 9 years ago
  81. a440c6f Roll chromium_revision 3b8be21..9dfb3a1 (363445:363718) by kjellander · 9 years ago
  82. 3868692 Free SCTP data channels asynchronously in PeerConnection. by deadbeef · 9 years ago
  83. 46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
  84. 6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  85. 84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
  86. 0f490a5 Add logs when stun or turn host lookup is completed. by Honghai Zhang · 9 years ago
  87. cd4003f Use @webrtc.org addresses for OWNERS. by Peter Boström · 9 years ago
  88. cf890bc Roll gtest-parallel. by Peter Boström · 9 years ago
  89. 0608dc5 Roll chromium_revision 4918765..3b8be21 (363393:363445) by kjellander · 9 years ago
  90. 5f6deaf Remove unused RTP-header parser. by Peter Boström · 9 years ago
  91. ab82cbb Disable RtcEventLogTest.DropOldEvents on memcheck. by Peter Boström · 9 years ago
  92. 03671cb Use existing parser in ReceivesAndRetransmitsNack. by Peter Boström · 9 years ago
  93. fc47ed6 rtcp::Rrtr block moved into own file and got Parse function by Danil Chapovalov · 9 years ago
  94. 1aa420b Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead. by asapersson · 9 years ago
  95. 9d69c3f Return a copy of the supported RTP header extensions instead of a reference. by Stefan Holmer · 9 years ago
  96. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  97. 03f80eb Refactor EglBase configuration. by nisse · 9 years ago
  98. a856542 Initial VideoProcessing refactoring. by mflodman · 9 years ago
  99. 2512f44 Roll chromium_revision 292ab9f..4918765 (363376:363393) by kjellander · 9 years ago
  100. c9f1cb8 Roll chromium_revision 72c3265..292ab9f (363365:363376) by kjellander · 9 years ago