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gerrit-public.fairphone.software / platform / external / webrtc / f3ba6484e3f1044b885fd33c4622e4562ba2fec8
  1. f3ba648 Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format by Danil Chapovalov · 7 years ago
  2. 29f0d45 Delete ApplicationName and OrganizationName. by nisse · 7 years ago
  3. b008b45 Update webrtc/sdk/objc to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  4. 687bc3e Delete unused method Win32Filesystem::GetAppPathname. by nisse · 7 years ago
  5. 418b7d3 Increase number of unsignaled audio streams we handle to 4. by solenberg · 7 years ago
  6. c18c49b Roll chromium_revision 239d4798df..0ca6ede735 (478894:478917) by buildbot · 7 years ago
  7. f52ef71 Delete unused method FilesystemInterface::DeleteEmptyFolder. by nisse · 7 years ago
  8. f9fc4a5 Roll chromium_revision 97580dea94..239d4798df (478848:478894) by buildbot · 7 years ago
  9. 385a6e4 Roll chromium_revision 15b2b0b0e9..97580dea94 (478791:478848) by buildbot · 7 years ago
  10. c35c7de Fix play block size mismatch in Win audio device. by lliuu · 7 years ago
  11. 84da736 Roll chromium_revision 71baf2eb8f..15b2b0b0e9 (478645:478791) by buildbot · 7 years ago
  12. 22e0814 Update VirtualSocketServerTest to use a fake clock. by deadbeef · 7 years ago
  13. 36b1a5f Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 7 years ago
  14. 0703856 Add SafeClamp(), which accepts args of different types by kwiberg · 7 years ago
  15. d1114c7 Roll chromium_revision d59edeefb6..71baf2eb8f (478597:478645) by buildbot · 7 years ago
  16. 38018ba Merge BitrateControllerImpl::RtcpBandwidthObserverImpl into BitrateControllerImpl by Danil Chapovalov · 7 years ago
  17. 42742a5 Fall-back to OpenGL renderer if mac hardware doesn't support Metal by adam.fedor · 7 years ago
  18. 84b4d2c Use rtp_header_extension_map.h instead of rtp_header_extension.h by Danil Chapovalov · 7 years ago
  19. d3d8702 Roll chromium_revision 6dcccd8c3f..d59edeefb6 (478515:478597) by buildbot · 7 years ago
  20. 7f8369a Update expectation of OneBitrateObserverTwoRtcpObservers test: by Danil Chapovalov · 7 years ago
  21. f474c19 ACM tests: separate checksums for Android ARM64 clang and non-clang by Henrik Lundin · 7 years ago
  22. 39a41d9 Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr. by perkj · 7 years ago
  23. 7123029 List all device resolutions in AppRTCMobile settings by Anders Carlsson · 7 years ago
  24. c276ecf Update Android video buffers to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  25. f184138 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 7 years ago
  26. a8e781a Make rtc_event_log2text output header extensions by ilnik · 7 years ago
  27. 3fae628 Reland Refactored incoming bitrate estimator. by tschumim · 7 years ago
  28. 90e3190 Update webrtc/test to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  29. 72dbe2a Revert "Revert "Update video_coding/codecs to new VideoFrameBuffer interface"" by Magnus Jedvert · 7 years ago
  30. 29584c5 Roll chromium_revision 4b325fbec4..6dcccd8c3f (478514:478515) by buildbot · 7 years ago
  31. ef0a3ea Roll chromium_revision 5a101abbe0..4b325fbec4 (478513:478514) by buildbot · 7 years ago
  32. c8cac10 Roll chromium_revision 632b145c0e..5a101abbe0 (478512:478513) by buildbot · 7 years ago
  33. 7e120eb Roll chromium_revision 8e89b0b1a1..632b145c0e (478506:478512) by buildbot · 7 years ago
  34. d0fa397 Roll chromium_revision 999a40e458..8e89b0b1a1 (478482:478506) by buildbot · 7 years ago
  35. ad3a029 Roll chromium_revision 1b59498f08..999a40e458 (478431:478482) by buildbot · 7 years ago
  36. 995bad0 Roll chromium_revision 524fdc6e30..1b59498f08 (478357:478431) by buildbot · 7 years ago
  37. c131bf9 Enable webrtc_nonparallel_tests on iOS simulator by kjellander · 7 years ago
  38. b82487b Roll chromium_revision f7c1799c98..524fdc6e30 (478294:478357) by buildbot · 7 years ago
  39. be767e0 Remove default impl of Attach/DetachAecDump. by Alex Loiko · 7 years ago
  40. 12149bd Roll chromium_revision 06a62c1231..f7c1799c98 (478256:478294) by buildbot · 7 years ago
  41. 76d29f9 Fix Channel::GetSendCodec when used together with SetEncoder. by ossu · 7 years ago
  42. 7fdd067 Roll chromium_revision f8c224c31c..06a62c1231 (478239:478256) by buildbot · 7 years ago
  43. 461c940 ObjC: Rename VideoToolbox/decoder.cc to VideoToolbox/decoder.mm by Magnus Jedvert · 7 years ago
  44. b4ab381 Use the configured remote ssrc instead of relying on the first received packet RtpStreamReceiver. by stefan · 7 years ago
  45. fee994c Ensure the openGLContext is current before trying to reshape the viewport by adam.fedor · 7 years ago
  46. b1f2ff9 Rename class RtpStreamReceiver --> RtpVideoStreamReceiver. by nisse · 7 years ago
  47. e2baffb Create a UIApplication when running tests on iOS. by Kári Tristan Helgason · 7 years ago
  48. fae6d09 Roll chromium_revision 74ece38823..f8c224c31c (478141:478239) by buildbot · 7 years ago
  49. 85dcaea Roll chromium_revision 423c0eff45..74ece38823 (478099:478141) by buildbot · 7 years ago
  50. 6baee78 Add missing #include <cerrno> in string_to_number.cc by hugoh · 7 years ago
  51. 46537a3 Avoiding cascaded software echo cancellers by Per Åhgren · 7 years ago
  52. 7412fe6 Roll chromium_revision 2108fde0a1..423c0eff45 (478041:478099) by buildbot · 7 years ago
  53. dc4f7f5 Roll chromium_revision 88476a9f88..2108fde0a1 (477979:478041) by buildbot · 7 years ago
  54. 59154ed Roll chromium_revision 61a28216c8..88476a9f88 (477949:477979) by buildbot · 7 years ago
  55. 20e4a73 MockAecDump and integration tests between AecDump and AudioProcessing by aleloi · 7 years ago
  56. 317005a Revert of Periodically update codec bit/frame rate settings. (patchset #2 id:160001 of https://codereview.webrtc.org/2924023002/ ) by sprang · 7 years ago
  57. d3a8119 Roll chromium_revision 53a49d4c81..61a28216c8 (477934:477949) by buildbot · 7 years ago
  58. cf705c5 Reland of Protect new header extension by field trial experiment to allow hardcoding it in SDP (patchset #1 id:1 of https://codereview.webrtc.org/2922723002/ ) by ilnik · 7 years ago
  59. cdafeda Reland of Periodically update codec bit/frame rate settings. (patchset #1 id:1 of https://codereview.webrtc.org/2923993002/ ) by sprang · 7 years ago
  60. 1066b13 Remove deprecated AudioMixerImpl creation method. by Alex Loiko · 7 years ago
  61. d0244c2 Add RSID-based demuxing to RtpDemuxer by eladalon · 7 years ago
  62. 5c4897f Roll chromium_revision 3c550cc859..53a49d4c81 (477916:477934) by buildbot · 7 years ago
  63. 15dcb38 Make error resilience configurable through VideoCodecVP9 resilience setting (removes hard coded value in vp9_impl.cc). by asapersson · 7 years ago
  64. 04ca637 Make 'aleloi@' OWNER of webrtc/modules/audio_processing by Alex Loiko · 7 years ago
  65. 75b68b9 Delete webrtc/call.h (replaced with webrtc/call/call.h). by nisse · 7 years ago
  66. 02ed201 AcmReceiver: Make a member variable const by Henrik Lundin · 7 years ago
  67. 88f94fa Revert "Update video_coding/codecs to new VideoFrameBuffer interface" by Guido Urdaneta · 7 years ago
  68. e566e17 Add new screenshare full stack test with limited queue. by sprang · 7 years ago
  69. 097ad90 Roll chromium_revision ac66f89e4b..3c550cc859 (477875:477916) by buildbot · 7 years ago
  70. 807736e Revert of Refactored incoming bitrate estimator. (patchset #8 id:140001 of https://codereview.webrtc.org/2917873002/ ) by tschumim · 7 years ago
  71. b9ed108 Roll chromium_revision db06e65dcd..ac66f89e4b (477837:477875) by buildbot · 7 years ago
  72. ff5e5c0 Roll chromium_revision 1caed9e9b4..db06e65dcd (477766:477837) by buildbot · 7 years ago
  73. 353f065 Roll chromium_revision 19f44c261d..1caed9e9b4 (477714:477766) by buildbot · 7 years ago
  74. d9432c2 Roll chromium_revision 98dee77021..19f44c261d (477661:477714) by buildbot · 7 years ago
  75. 4c72cf4 Revert of Conversational speech tool, simualtor + unit tests (patchset #12 id:220001 of https://codereview.webrtc.org/2790933002/ ) by charujain · 7 years ago
  76. 6b648c4 The simulator puts into action the schedule of speech turns encoded in a MultiEndCall instance. The output is a set of audio track pairs. There is one set for each speaker and each set contains one near-end and one far-end audio track. The tracks are directly written into wav files instead of creating them in memory. To speed up the creation of the output wav files, *all* the source audio tracks (i.e., the atomic speech turns) are pre-loaded. by alessiob · 7 years ago
  77. bb28b35 Roll chromium_revision c6f978a173..98dee77021 (477619:477661) by buildbot · 7 years ago
  78. 5fc8bf8 Refactored incoming bitrate estimator. by tschumim · 7 years ago
  79. 20ebf4e Update video_coding/codecs to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  80. 9932e25 ObjC: Marshal all VideoTrackSource methods to the signaling thread by Magnus Jedvert · 7 years ago
  81. 5390c48 Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ ) by sprang · 7 years ago
  82. 11c89f5 Roll chromium_revision 1f3b0bc457..c6f978a173 (477597:477619) by buildbot · 7 years ago
  83. 6431e21 Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended. by sprang · 7 years ago
  84. 2038df4 Deleting unused build target. by mbonadei · 7 years ago
  85. 8b337b6 Remove outdated warning suppressions. by Kári Tristan Helgason · 7 years ago
  86. 946923a Remove webrtc deps from AppRTCMobile. by Kári Tristan Helgason · 7 years ago
  87. 1e15a99 MediaCodecVideoEncoder: Add QP stats to Encoded callback for VP9 and turn on quality scaling. by asapersson · 7 years ago
  88. 3e35a3c iOS: Update storage bucket for iOS64_Debug bot. by Henrik Kjellander · 7 years ago
  89. 7f0ddb0 Roll chromium_revision 99c4a934d4..1f3b0bc457 (477570:477597) by buildbot · 7 years ago
  90. 6b9653e ObjC: Pass in frame resolution to GL shaders by Magnus Jedvert · 7 years ago
  91. f53c4cd Delete rtc_event_log/ringbuffer.h by terelius · 7 years ago
  92. 5665caf Roll chromium_revision 24fca6fb8a..99c4a934d4 (477537:477570) by buildbot · 7 years ago
  93. 1387476 Compare adapt up/down request with sink_wants_ in VideoSourceProxy methods to make sure it is higher/lower than last request. by asapersson · 7 years ago
  94. 23ec19d Add fuzzer for vp9 qp parser. by asapersson · 7 years ago
  95. 5f200f6 Roll chromium_revision a6ac8624bc..24fca6fb8a (477491:477537) by buildbot · 7 years ago
  96. 832e67a Roll chromium_revision 5c265a308a..a6ac8624bc (477423:477491) by buildbot · 7 years ago
  97. 1c9fa21 Roll chromium_revision b1a5432280..5c265a308a (477354:477423) by buildbot · 7 years ago
  98. cc385d5 Roll chromium_revision a99122347a..b1a5432280 (477301:477354) by buildbot · 7 years ago
  99. 30fb370 Roll chromium_revision ee6a8849e4..a99122347a (477267:477301) by buildbot · 7 years ago
  100. 8cde0ab Roll chromium_revision 354a45f6eb..ee6a8849e4 (477247:477267) by buildbot · 7 years ago