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gerrit-public.fairphone.software / platform / external / webrtc / 55bcf0f087cf01dfd6230b02a665d2a88fb9d150
  1. 55bcf0f Fix -Wformat error in Win-Clang build (take 2) by hans · 9 years ago
  2. 013e83b Fix -Wformat error in Win-Clang build by Niklas Enbom · 9 years ago
  3. cf846ad Adding stub files needed for https://codereview.webrtc.org/1507973003/ by Taylor Brandstetter · 9 years ago
  4. 7c73bdb Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor. by deadbeef · 9 years ago
  5. ed83edc Roll chromium_revision 2e451bf..026b937 (364330:364421) by kjellander · 9 years ago
  6. 6a6f089 in rtp_rtcp module: by danilchap · 9 years ago
  7. 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
  8. 61a90f9 clang/win: Fix -Wextra warnings in webrtc. by thakis · 9 years ago
  9. 5c1def8 modules/rtp_rtcp/include folder cleared of lint warnings by danilchap · 9 years ago
  10. 796cfaf Add VideoCodec::PreferDecodeLate by perkj · 9 years ago
  11. 4d68208 Reduce the runtime of some ACM tests in modules_tests by Henrik Lundin · 9 years ago
  12. c490e01 Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to by nisse · 9 years ago
  13. b8b6fbb lint build/include errors fixed in rtp_rtcp module by danilchap · 9 years ago
  14. 90b9fc9 Roll chromium_revision a02d286..2e451bf (364268:364330) by kjellander · 9 years ago
  15. 866df66 Typo fix: Enable a bunch of tests that were accidentally disabled by kwiberg · 9 years ago
  16. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  17. 0f2e939 Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors. by jbauch · 9 years ago
  18. 162abd3 lint whitespace warning removed from most rtp_rtcp/source/ files by danilchap · 9 years ago
  19. 84e78f9 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. by terelius · 9 years ago
  20. 0b3d7ee Prevent RTCP SR to be sent with bogus timestamp. by mflodman · 9 years ago
  21. 48bf238 Some further minor bitexact APM echo suppressor refactoring by peah · 9 years ago
  22. 5ba58c6 Roll chromium_revision dad6346..a02d286 (363782:364268) by kjellander · 9 years ago
  23. a6e4328 Remove unnecessary test code on Windows. by Tommi · 9 years ago
  24. 70625e5 Enable cpplint for webrtc/examples and fix all uncovered cpplint errors. by jbauch · 9 years ago
  25. 2e5fe31 Remove myself from root_files watchlist. by andrew · 9 years ago
  26. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  27. ee40821 WebRTC: Update set of known root certificates by Guo-wei Shieh · 9 years ago
  28. b14f001 Some minor (bitexact) AEC echo suppressor refactoring by peah · 9 years ago
  29. 434aca8 Add empty placeholder files for remote audio tracks. by tommi · 9 years ago
  30. afeb438 Moved code into the lowest level of EchoSuppression by peah · 9 years ago
  31. d1590b2 Lint clean video/ and add lint presubmit check. by mflodman · 9 years ago
  32. 4cf61dd NetEq: Add codec name and RTP timestamp rate to DecoderInfo by henrik.lundin · 9 years ago
  33. 3980d46 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). by hbos · 9 years ago
  34. af3b9cb Removing DrMemory suppresssion on PushResampler. by minyuel · 9 years ago
  35. 5eb4988 [rtp_rtcp] Lint build/header_guard errors fixed by danilchap · 9 years ago
  36. 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
  37. d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  38. bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
  39. d02b0fa Add oldest rotation type option to RTCFileLogger by haysc · 9 years ago
  40. 5e465c3 Make NoiseSuppression not a processing component (bit exact). by solenberg · 9 years ago
  41. 1a9d615 Add tracing to public PeerConnection methods. by Peter Boström · 9 years ago
  42. 2d63680 Roll chromium_revision 9dfb3a1..dad6346 (363718:363782) by kjellander · 9 years ago
  43. 7b2f762 Don't call SetPreviewFormat if capturing to textures. by perkj · 9 years ago
  44. edd8fef Add new view that renders local video using AVCaptureLayerPreview. by haysc · 9 years ago
  45. 70f9903 Make HighPassFilter not a ProcessingComponent anymore (bit exact). by solenberg · 9 years ago
  46. 246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
  47. 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
  48. e10c82d Deletes temporary files that are generated in several ACM unittests. by ivoc · 9 years ago
  49. d7b7ae8 Add encode/decode time tracing to audio_coding. by Peter Boström · 9 years ago
  50. 9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
  51. cd6f539 Revert of RTCCertificate::Expires() and ::HasExpired() implemented (patchset #5 id:140001 of https://codereview.webrtc.org/1494103003/ ) by hbos · 9 years ago
  52. fe32a76 Create fuzzer tests for audio decoders by Henrik Lundin · 9 years ago
  53. ffea13c PRESUBMIT: change native API check from warning to information. by kjellander · 9 years ago
  54. 20ef654 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). by hbos · 9 years ago
  55. 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
  56. 88eeac4 Adding video_processing to presubmit lint check by mflodman · 9 years ago
  57. 4654d20 Add test which verifies that the RTP header extensions are set correctly for FEC packets. by Stefan Holmer · 9 years ago
  58. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  59. 99ab944 Clang format of video_processing folder. by mflodman · 9 years ago
  60. a440c6f Roll chromium_revision 3b8be21..9dfb3a1 (363445:363718) by kjellander · 9 years ago
  61. 3868692 Free SCTP data channels asynchronously in PeerConnection. by deadbeef · 9 years ago
  62. 46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
  63. 6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  64. 84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
  65. 0f490a5 Add logs when stun or turn host lookup is completed. by Honghai Zhang · 9 years ago
  66. cd4003f Use @webrtc.org addresses for OWNERS. by Peter Boström · 9 years ago
  67. cf890bc Roll gtest-parallel. by Peter Boström · 9 years ago
  68. 0608dc5 Roll chromium_revision 4918765..3b8be21 (363393:363445) by kjellander · 9 years ago
  69. 5f6deaf Remove unused RTP-header parser. by Peter Boström · 9 years ago
  70. ab82cbb Disable RtcEventLogTest.DropOldEvents on memcheck. by Peter Boström · 9 years ago
  71. 03671cb Use existing parser in ReceivesAndRetransmitsNack. by Peter Boström · 9 years ago
  72. fc47ed6 rtcp::Rrtr block moved into own file and got Parse function by Danil Chapovalov · 9 years ago
  73. 1aa420b Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead. by asapersson · 9 years ago
  74. 9d69c3f Return a copy of the supported RTP header extensions instead of a reference. by Stefan Holmer · 9 years ago
  75. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  76. 03f80eb Refactor EglBase configuration. by nisse · 9 years ago
  77. a856542 Initial VideoProcessing refactoring. by mflodman · 9 years ago
  78. 2512f44 Roll chromium_revision 292ab9f..4918765 (363376:363393) by kjellander · 9 years ago
  79. c9f1cb8 Roll chromium_revision 72c3265..292ab9f (363365:363376) by kjellander · 9 years ago
  80. 34a7054 Roll chromium_revision 626eecf..72c3265 (363027:363365) by kjellander · 9 years ago
  81. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  82. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  83. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  84. 381b421 Ping backup connection at a slower rate by Honghai Zhang · 9 years ago
  85. 45b0efd Stop sending stun binding requests after certain amount of time. by honghaiz · 9 years ago
  86. 9e1b992 Clear old decoders after recreating the receiver. by Peter Boström · 9 years ago
  87. 97f7e13 rtcp::ReceiverReport moved into own file and got Parse function by Danil Chapovalov · 9 years ago
  88. 7c704b8 Use webrtc/base/logging.h in stefan@'s ownership. by Peter Boström · 9 years ago
  89. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  90. fcdcf4a Disable RtcEventLogTest.DropOldEvents on DrMemory. by Peter Boström · 9 years ago
  91. 66f7f4e Roll chromium_revision d3aa9b1..626eecf (362950:363027) by kjellander · 9 years ago
  92. fd59523 Add webrtc/base to deprecated APIs. by kjellander · 9 years ago
  93. bc32ab4 Remove 'video_engine_core_unittests' binary. by Peter Boström · 9 years ago
  94. ff24c04 Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. by Åsa Persson · 9 years ago
  95. 1a5cf6e Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine). by Fredrik Solenberg · 9 years ago
  96. f7c5776 Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket. by Erik Språng · 9 years ago
  97. 9cf0c3d Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient. by Ivo Creusen · 9 years ago
  98. d048aa0 Make the audio codecs' GN targets self-sufficient by Henrik Lundin · 9 years ago
  99. b4a1ae5 Add separate send-side UMA stats for screenshare and video. by sprang · 9 years ago
  100. 29e3003 Bring back baremetal trybots to the default set. by kjellander@webrtc.org · 9 years ago