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gerrit-public.fairphone.software / platform / external / webrtc / 7ed35f46433e6da91e7843eab20bad92a8f2b145
  1. 7ed35f4 Replacing WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP with WEBRTC_ENABLE_PROTOBUF. by minyue-webrtc · 7 years ago
  2. 10e1f75 Roll chromium_revision 9061a92f5c..4f7c2dc196 (478958:478995) by buildbot · 7 years ago
  3. 2986033 Remove webrtcvideoengine2.h by eladalon · 7 years ago
  4. 659a010 Delete old include file webrtc/video_frame.h. by nisse · 7 years ago
  5. a65ad22 Delete unused method FilesystemInterface::GetFileTime. by nisse · 7 years ago
  6. 8c6afef Make sure UI methods get called on the main thread by adam.fedor · 7 years ago
  7. fdfeb83 Declaring rtc_base_approved dep on webrtc_common by mbonadei · 7 years ago
  8. 7339712 Removing backward compatible header by mbonadei · 7 years ago
  9. a735d4e Roll chromium_revision 0ca6ede735..9061a92f5c (478917:478958) by buildbot · 7 years ago
  10. 2c9f9f2 Only create H264 frames if there are no gaps in the packet sequence number. by philipel · 7 years ago
  11. fc30975 Access UIApplication on main thread by Anders Carlsson · 7 years ago
  12. 5b383c0 Revert "Update webrtc/sdk/objc to new VideoFrameBuffer interface" by Magnus Jedvert · 7 years ago
  13. 1edbda0 Don't hardcode gn target path for licence generation. by Kári Tristan Helgason · 7 years ago
  14. f3ba648 Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format by Danil Chapovalov · 7 years ago
  15. 29f0d45 Delete ApplicationName and OrganizationName. by nisse · 7 years ago
  16. b008b45 Update webrtc/sdk/objc to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  17. 687bc3e Delete unused method Win32Filesystem::GetAppPathname. by nisse · 7 years ago
  18. 418b7d3 Increase number of unsignaled audio streams we handle to 4. by solenberg · 7 years ago
  19. c18c49b Roll chromium_revision 239d4798df..0ca6ede735 (478894:478917) by buildbot · 7 years ago
  20. f52ef71 Delete unused method FilesystemInterface::DeleteEmptyFolder. by nisse · 7 years ago
  21. f9fc4a5 Roll chromium_revision 97580dea94..239d4798df (478848:478894) by buildbot · 7 years ago
  22. 385a6e4 Roll chromium_revision 15b2b0b0e9..97580dea94 (478791:478848) by buildbot · 7 years ago
  23. c35c7de Fix play block size mismatch in Win audio device. by lliuu · 7 years ago
  24. 84da736 Roll chromium_revision 71baf2eb8f..15b2b0b0e9 (478645:478791) by buildbot · 7 years ago
  25. 22e0814 Update VirtualSocketServerTest to use a fake clock. by deadbeef · 7 years ago
  26. 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
  27. 0703856 Add SafeClamp(), which accepts args of different types by kwiberg · 7 years ago
  28. d1114c7 Roll chromium_revision d59edeefb6..71baf2eb8f (478597:478645) by buildbot · 7 years ago
  29. 38018ba Merge BitrateControllerImpl::RtcpBandwidthObserverImpl into BitrateControllerImpl by Danil Chapovalov · 7 years ago
  30. 42742a5 Fall-back to OpenGL renderer if mac hardware doesn't support Metal by adam.fedor · 7 years ago
  31. 84b4d2c Use rtp_header_extension_map.h instead of rtp_header_extension.h by Danil Chapovalov · 7 years ago
  32. d3d8702 Roll chromium_revision 6dcccd8c3f..d59edeefb6 (478515:478597) by buildbot · 7 years ago
  33. 7f8369a Update expectation of OneBitrateObserverTwoRtcpObservers test: by Danil Chapovalov · 7 years ago
  34. f474c19 ACM tests: separate checksums for Android ARM64 clang and non-clang by Henrik Lundin · 7 years ago
  35. 39a41d9 Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr. by perkj · 7 years ago
  36. 7123029 List all device resolutions in AppRTCMobile settings by Anders Carlsson · 7 years ago
  37. c276ecf Update Android video buffers to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  38. f184138 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 7 years ago
  39. a8e781a Make rtc_event_log2text output header extensions by ilnik · 7 years ago
  40. 3fae628 Reland Refactored incoming bitrate estimator. by tschumim · 7 years ago
  41. 90e3190 Update webrtc/test to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  42. 72dbe2a Revert "Revert "Update video_coding/codecs to new VideoFrameBuffer interface"" by Magnus Jedvert · 7 years ago
  43. 29584c5 Roll chromium_revision 4b325fbec4..6dcccd8c3f (478514:478515) by buildbot · 7 years ago
  44. ef0a3ea Roll chromium_revision 5a101abbe0..4b325fbec4 (478513:478514) by buildbot · 7 years ago
  45. c8cac10 Roll chromium_revision 632b145c0e..5a101abbe0 (478512:478513) by buildbot · 7 years ago
  46. 7e120eb Roll chromium_revision 8e89b0b1a1..632b145c0e (478506:478512) by buildbot · 7 years ago
  47. d0fa397 Roll chromium_revision 999a40e458..8e89b0b1a1 (478482:478506) by buildbot · 7 years ago
  48. ad3a029 Roll chromium_revision 1b59498f08..999a40e458 (478431:478482) by buildbot · 7 years ago
  49. 995bad0 Roll chromium_revision 524fdc6e30..1b59498f08 (478357:478431) by buildbot · 7 years ago
  50. c131bf9 Enable webrtc_nonparallel_tests on iOS simulator by kjellander · 7 years ago
  51. b82487b Roll chromium_revision f7c1799c98..524fdc6e30 (478294:478357) by buildbot · 7 years ago
  52. be767e0 Remove default impl of Attach/DetachAecDump. by Alex Loiko · 7 years ago
  53. 12149bd Roll chromium_revision 06a62c1231..f7c1799c98 (478256:478294) by buildbot · 7 years ago
  54. 76d29f9 Fix Channel::GetSendCodec when used together with SetEncoder. by ossu · 7 years ago
  55. 7fdd067 Roll chromium_revision f8c224c31c..06a62c1231 (478239:478256) by buildbot · 7 years ago
  56. 461c940 ObjC: Rename VideoToolbox/decoder.cc to VideoToolbox/decoder.mm by Magnus Jedvert · 7 years ago
  57. b4ab381 Use the configured remote ssrc instead of relying on the first received packet RtpStreamReceiver. by stefan · 7 years ago
  58. fee994c Ensure the openGLContext is current before trying to reshape the viewport by adam.fedor · 7 years ago
  59. b1f2ff9 Rename class RtpStreamReceiver --> RtpVideoStreamReceiver. by nisse · 7 years ago
  60. e2baffb Create a UIApplication when running tests on iOS. by Kári Tristan Helgason · 7 years ago
  61. fae6d09 Roll chromium_revision 74ece38823..f8c224c31c (478141:478239) by buildbot · 7 years ago
  62. 85dcaea Roll chromium_revision 423c0eff45..74ece38823 (478099:478141) by buildbot · 7 years ago
  63. 6baee78 Add missing #include <cerrno> in string_to_number.cc by hugoh · 7 years ago
  64. 46537a3 Avoiding cascaded software echo cancellers by Per Åhgren · 7 years ago
  65. 7412fe6 Roll chromium_revision 2108fde0a1..423c0eff45 (478041:478099) by buildbot · 7 years ago
  66. dc4f7f5 Roll chromium_revision 88476a9f88..2108fde0a1 (477979:478041) by buildbot · 7 years ago
  67. 59154ed Roll chromium_revision 61a28216c8..88476a9f88 (477949:477979) by buildbot · 7 years ago
  68. 20e4a73 MockAecDump and integration tests between AecDump and AudioProcessing by aleloi · 7 years ago
  69. 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
  70. d3a8119 Roll chromium_revision 53a49d4c81..61a28216c8 (477934:477949) by buildbot · 7 years ago
  71. 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
  72. 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
  73. 1066b13 Remove deprecated AudioMixerImpl creation method. by Alex Loiko · 7 years ago
  74. d0244c2 Add RSID-based demuxing to RtpDemuxer by eladalon · 7 years ago
  75. 5c4897f Roll chromium_revision 3c550cc859..53a49d4c81 (477916:477934) by buildbot · 7 years ago
  76. 15dcb38 Make error resilience configurable through VideoCodecVP9 resilience setting (removes hard coded value in vp9_impl.cc). by asapersson · 7 years ago
  77. 04ca637 Make 'aleloi@' OWNER of webrtc/modules/audio_processing by Alex Loiko · 7 years ago
  78. 75b68b9 Delete webrtc/call.h (replaced with webrtc/call/call.h). by nisse · 7 years ago
  79. 02ed201 AcmReceiver: Make a member variable const by Henrik Lundin · 7 years ago
  80. 88f94fa Revert "Update video_coding/codecs to new VideoFrameBuffer interface" by Guido Urdaneta · 7 years ago
  81. e566e17 Add new screenshare full stack test with limited queue. by sprang · 7 years ago
  82. 097ad90 Roll chromium_revision ac66f89e4b..3c550cc859 (477875:477916) by buildbot · 7 years ago
  83. 807736e Revert of Refactored incoming bitrate estimator. (patchset #8 id:140001 of https://codereview.webrtc.org/2917873002/ ) by tschumim · 7 years ago
  84. b9ed108 Roll chromium_revision db06e65dcd..ac66f89e4b (477837:477875) by buildbot · 7 years ago
  85. ff5e5c0 Roll chromium_revision 1caed9e9b4..db06e65dcd (477766:477837) by buildbot · 7 years ago
  86. 353f065 Roll chromium_revision 19f44c261d..1caed9e9b4 (477714:477766) by buildbot · 7 years ago
  87. d9432c2 Roll chromium_revision 98dee77021..19f44c261d (477661:477714) by buildbot · 7 years ago
  88. 4c72cf4 Revert of Conversational speech tool, simualtor + unit tests (patchset #12 id:220001 of https://codereview.webrtc.org/2790933002/ ) by charujain · 7 years ago
  89. 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
  90. bb28b35 Roll chromium_revision c6f978a173..98dee77021 (477619:477661) by buildbot · 7 years ago
  91. 5fc8bf8 Refactored incoming bitrate estimator. by tschumim · 7 years ago
  92. 20ebf4e Update video_coding/codecs to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  93. 9932e25 ObjC: Marshal all VideoTrackSource methods to the signaling thread by Magnus Jedvert · 7 years ago
  94. 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
  95. 11c89f5 Roll chromium_revision 1f3b0bc457..c6f978a173 (477597:477619) by buildbot · 7 years ago
  96. 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
  97. 2038df4 Deleting unused build target. by mbonadei · 7 years ago
  98. 8b337b6 Remove outdated warning suppressions. by Kári Tristan Helgason · 7 years ago
  99. 946923a Remove webrtc deps from AppRTCMobile. by Kári Tristan Helgason · 7 years ago
  100. 1e15a99 MediaCodecVideoEncoder: Add QP stats to Encoded callback for VP9 and turn on quality scaling. by asapersson · 7 years ago