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gerrit-public.fairphone.software / platform / external / webrtc / 53e048d83aa47de7bf98d1f10c667bd0e79551e7
  1. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 6 years ago
  2. aa6b24f Delete DefaultAudioProcessingFactory in the Java layer by Sam Zackrisson · 6 years ago
  3. a4eeeff Fixing API list in native-api.md (adding media/base and media/engine). by Mirko Bonadei · 6 years ago
  4. fedc00c Optional: Use nullopt and implicit construction in /call by Oskar Sundbom · 7 years ago
  5. 6f49106 Block more check_includes overrides from being added. by Patrik Höglund · 6 years ago
  6. c77b528 Adds usage of RTC_LOG macros in JNI audio code on Android. by henrika · 7 years ago
  7. 6f9c01f Delete use of pthread_cond_timedwait_relative_np. by Niels Möller · 6 years ago
  8. 10b96eb Optional: Use nullopt and implicit construction in /modules/pacing by Oskar Sundbom · 6 years ago
  9. b3fb938 Remove obsolete TODO. by Patrik Höglund · 6 years ago
  10. e5a93a7 Making rtc_tools:unpack_aecdump publicly visible. by Mirko Bonadei · 6 years ago
  11. 448691c Remove the old videosinkinterface.h. by Patrik Höglund · 6 years ago
  12. adc1e9b Remove old videosourceinterface files. by Patrik Höglund · 7 years ago
  13. f075c5f Delete obsolete header file transportchannelimpl.h. by Niels Möller · 6 years ago
  14. d3c642b Fix typo in the include path of ooura_fft.h by Jiawei Ou · 7 years ago
  15. ef65ef1 Fire OnAddTrack when Unified Plan is enabled by Steve Anton · 6 years ago
  16. 02ee47c Signal track ID correctly when Unified Plan semantics selected by Steve Anton · 6 years ago
  17. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 6 years ago
  18. d4d92b6 Roll chromium_revision b7c48e4a30..ec629e65bf (528253:528433) by Autoroller · 6 years ago
  19. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 6 years ago
  20. a0f5f60 Revert "Revert "Make some more targets publicly visible"" by Per Kjellander · 6 years ago
  21. a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 6 years ago
  22. 0e6d2f5 Use the filter delay to use the proper render block in the AEC3 AecState by Per Åhgren · 7 years ago
  23. 77a47e0 Handle any exception in Camera.setPreviewTexture. by Yura Yaroshevich · 6 years ago
  24. c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 6 years ago
  25. 55d1809 Revert "Make some more targets publicly visible" by Per Kjellander · 6 years ago
  26. b84a909 Revert "Add completion callbacks to RTCCameraVideoCapturer start/stop operations" by Erik Språng · 6 years ago
  27. 60d1792 Make some more targets publicly visible by Karl Wiberg · 6 years ago
  28. 12d3a93 Remove luci win trybot for now. by Patrik Höglund · 6 years ago
  29. b4c188d Added logging of the maximum observed API call jitter in AEC3 by Per Åhgren · 7 years ago
  30. ce804dd Remove gradle from DEPS. by Edward Lemur · 6 years ago
  31. 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 6 years ago
  32. 84804b0 Make getNativePeerConnection public. by Sami Kalliomäki · 7 years ago
  33. e23a9e8 Add completion callbacks to RTCCameraVideoCapturer start/stop operations by Gustavo Garcia · 7 years ago
  34. 438663e DCHECKS added to GetSimulcastConfig. by Seth Hampson · 7 years ago
  35. 7e60de2 Enable orphan checks for all .h files. by Patrik Höglund · 7 years ago
  36. 873e565 Remove obsolete TODO. by Patrik Höglund · 7 years ago
  37. 14de824 Stop using std::random_shuffle in favor of std::shuffle. by Mirko Bonadei · 7 years ago
  38. 0e1817b Roll chromium_revision 5da649a4e0..b7c48e4a30 (528145:528253) by Autoroller · 6 years ago
  39. 002f921 Inline default constructors for MediaChannel structs by Steve Anton · 7 years ago
  40. dfe9ffc Added active field to constructor and ToString() of VideoStream. by Seth Hampson · 7 years ago
  41. 43bd705 Roll chromium_revision 1fa7f9b489..5da649a4e0 (528002:528145) by Autoroller · 7 years ago
  42. 5fcc7b5 Roll chromium_revision 95e51a93db..1fa7f9b489 (527891:528002) by Autoroller · 7 years ago
  43. 8020ae8 Adding test to check for c++ in framework headers. by Peter Hanspers · 7 years ago
  44. 8e07c13 Optional: Use nullopt and implicit construction in /video by Oskar Sundbom · 7 years ago
  45. 62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
  46. 8bb8308 Revert "iOS: Don't upload perf results yet." by Edward Lemur · 7 years ago
  47. 86b893c Delete declaration of LogMultilineState and LogMultiline. by Niels Möller · 7 years ago
  48. 825249f Add final missing header files before enabling orphans check. by Patrik Höglund · 7 years ago
  49. 3460fa6 Use .empty() instead of '!= ""' by Edward Lemur · 7 years ago
  50. 7f331fa Add metric name for MinVideoAndAudioBitRate. by Edward Lemur · 7 years ago
  51. 6728003 Skip H246 scaling lists in SPS packets by Todd Wong · 7 years ago
  52. 81ca3bf Including rtc_base/flags.h after test/gtest.h. by Mirko Bonadei · 7 years ago
  53. 85e6f39 Roll chromium_revision 0bc7995b0c..95e51a93db (527790:527891) by Autoroller · 7 years ago
  54. 65d9b54 Roll chromium_revision e50425bed9..0bc7995b0c (527616:527790) by Autoroller · 7 years ago
  55. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  56. a9329db Bug Fix: Peers Cannot Communicate If One With Stereo Codec, One Not by Qiang Chen · 7 years ago
  57. e6aca63 Avoids audio crash in combination with invalid audio session on iOS. by henrika · 7 years ago
  58. 9a0a17f Make it possible to change the amplitude of the pulses generated by PulsedNoiseCapturer. by erikvarga@webrtc.org · 7 years ago
  59. 6206eef iOS: Don't upload perf results yet. by Edward Lemur · 7 years ago
  60. 730add8 Fix release shader resources in the right OpenGL context by Gustavo Garcia · 7 years ago
  61. e66572b Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
  62. 7be31af Roll chromium_revision 39876e57c3..e50425bed9 (526975:527616) by Autoroller · 7 years ago
  63. 99175c6 Add untracked headers to video_coding. by Patrik Höglund · 7 years ago
  64. c492bf1 Fix JSON format for reporting perf results. by Edward Lemur · 7 years ago
  65. c643204 Remove sparse BWE update field trial and instead enable it by default. by Bjorn Terelius · 7 years ago
  66. a114c88 Reland "Reland "Reland "Put internal video codec factories into separate target""" by Anders Carlsson · 7 years ago
  67. c9e74f5 Removing API forwarding headers. by Mirko Bonadei · 7 years ago
  68. 625c3b6 Add missing file to p2p. by Patrik Höglund · 7 years ago
  69. f641687 Forward fixing WebRTC to compile with Android NDK r16. by Mirko Bonadei · 7 years ago
  70. 2d6c76a Switch to using AddTrack with stream labels by Steve Anton · 7 years ago
  71. 43a723a Move AddTrack stats update to submethod by Steve Anton · 7 years ago
  72. dc8b5ab Remove dead code for media channel errors by Steve Anton · 7 years ago
  73. 8171211 Fix shadowed variable in ContentInfo by Steve Anton · 7 years ago
  74. 2767632 Expose RTCDtmfSender API via RTCRtpSender in ObjC SDK. by Yura Yaroshevich · 7 years ago
  75. bf56712 Exposed setOptions via RTCPeerConnectionFactory by Yura Yaroshevich · 7 years ago
  76. e0c2eea Simplifying video engine test code. by srte · 7 years ago
  77. b960e41 Add missing files to audio_coding by Patrik Höglund · 7 years ago
  78. 947f3fe Fix reporting of perf results on PlaysOutAudioAndVideoInSync* tests by Edward Lemur · 7 years ago
  79. 6d3ed71 Add missing audio_device files. by Patrik Höglund · 7 years ago
  80. dd3987f Add _[no]red suffix to RampUpTests. by Edward Lemur · 7 years ago
  81. 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
  82. 75ceef2 Pivot old stats generation to iterate senders/receivers by Harald Alvestrand · 7 years ago
  83. bb4bbfd Roll chromium_revision 941cdfd0c2..39876e57c3 (526830:526975) by Autoroller · 7 years ago
  84. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  85. 085bb64 Adds WebRTC.Audio.XXXRecordingDurationMs UMA stat on Android. by henrika · 7 years ago
  86. 4f024ef Renaming android_library android_manifest to android_manifest_for_lint. by Mirko Bonadei · 7 years ago
  87. 7613454 Reland: Fix circular dependency in BWE code. by Patrik Höglund · 7 years ago
  88. 9226f05 Add linux32_dbg to CQ. by Edward Lemur · 7 years ago
  89. 3b116ad Check keepAlive before calling nativeDataIsRecording (reland) by henrika · 7 years ago
  90. 81d2e3d Do not build h264 on Linux32 Debug bots. by Edward Lemur · 7 years ago
  91. 389a97c Fixing leaked reference from SCTP transport to DTLS/ICE transport. by Taylor Brandstetter · 7 years ago
  92. 1b17981 Roll chromium_revision 5cbb672076..941cdfd0c2 (526700:526830) by Autoroller · 7 years ago
  93. 8e126fc Remove RtcEventLoggingStarted and RtcEventLoggingStopped events. by Bjorn Terelius · 7 years ago
  94. c564d01 Fully remove win_msvc bots by Oleh Prypin · 7 years ago
  95. bf0aa42 Roll chromium_revision 677d1be0fd..5cbb672076 (526600:526700) by Autoroller · 7 years ago
  96. d1d6743 Use PrintResult* instead of printf in video/video_quality_test.cc by Edward Lemur · 7 years ago
  97. d420903 Add missing audio_device .h files. by Patrik Höglund · 7 years ago
  98. 35d2b7e Fix reporting of perf results on video full stack tests. by Edward Lemur · 7 years ago
  99. 731082c Reland: Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
  100. 9cc67dc Roll chromium_revision 2471b3de96..677d1be0fd (526492:526600) by Autoroller · 7 years ago