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gerrit-public.fairphone.software / platform / external / webrtc / c5dd3009b4b9b1f3be845fe8982db7dd4c268929
  1. c5dd300 Introduce RtpPacket::GetExtension accessor that return result by Danil Chapovalov · 6 years ago
  2. 357f596 Split a separate codecs target off of :video_jni by Jonathan Yu · 6 years ago
  3. 5bb1ed6 Eliminate use of EventWrapper from ios audio device tests by Niels Möller · 6 years ago
  4. a33c7af Tolerate optional chunks in WAV files by Alessio Bazzica · 6 years ago
  5. c496d58 Add flag for fast jitter buffer playout in neteq simulation by Sam Zackrisson · 6 years ago
  6. e6c2c08 MsanUninitialized: restric type check to msan case. by Alessio Bazzica · 6 years ago
  7. c4e9825 Delete classes EventFactory and EventFactoryImpl. by Niels Möller · 6 years ago
  8. 2a74263 Make the bitrate_allocator param optional to prepare for its removal by Oleh Prypin · 6 years ago
  9. cd2e105 Reenable test RampUpTest.AudioTransportSequenceNumber by Niels Möller · 6 years ago
  10. 694ed17 Add a style rule about not using const optional<T>& arguments by Karl Wiberg · 6 years ago
  11. f0e7440 Add missing conditional defines to neteq test and tools targets by Sam Zackrisson · 6 years ago
  12. 689983f Deprecate EventFactory and delete all usage. by Niels Möller · 6 years ago
  13. 54b4924 Update H264 encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  14. 1060870 Update LibVpxVp8Encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  15. 727d164 Update VP9 encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  16. 5473a45 Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities by Florent Castelli · 6 years ago
  17. 75de46a Update SimulcastEncoderAdapter merging of EncoderInfo by Erik Språng · 6 years ago
  18. e6a2d94 Clear FrameBuffer if there were no frames received for 10 minutes by Ilya Nikolaevskiy · 6 years ago
  19. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  20. bdc6c40 Add field trial for target bitrate RTCP XR message. by Rasmus Brandt · 6 years ago
  21. d565918 Delete NullEventFactory by Niels Möller · 6 years ago
  22. e769ed9 Roll chromium_revision 38dcb5ed01..db720b4ab9 (605924:606025) by chromium-webrtc-autoroll · 6 years ago
  23. 50f60cb Rename software codec classes and move them into api/ by Jonathan Yu · 6 years ago
  24. ff7020a Remove non-default VideoEncoder::EncoderInfo() ctor by Erik Språng · 6 years ago
  25. 36d907b Update MockVideoEncoder with correct methods. by Erik Språng · 6 years ago
  26. 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  27. a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
  28. 7553c02 Update ObjCVideoEncoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  29. 7b3c76b Reland "Delete rtc::Pathname" by Niels Möller · 6 years ago
  30. 17fc7e2 Add counter to the end of FakeEncoder frames in order to make them unique. by Per Kjellander · 6 years ago
  31. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  32. 3ea7b83 Resolve the race condition between mDNS name registration and by Qingsi Wang · 6 years ago
  33. 8770ce7 Roll chromium_revision 03cf97f6d8..38dcb5ed01 (605818:605924) by chromium-webrtc-autoroll · 6 years ago
  34. bb091db Roll chromium_revision 793c8566ab..03cf97f6d8 (605715:605818) by chromium-webrtc-autoroll · 6 years ago
  35. 2cd3b4c Fixing bug in SimulatedNetwork where packets stop. by Sebastian Jansson · 6 years ago
  36. 0f54f21 Removes deprecated GetSentPacket from PacketResult. by Sebastian Jansson · 6 years ago
  37. dc98b9b AEC3: Corrected include by Per Åhgren · 6 years ago
  38. c564a7b Roll chromium_revision 7841106b37..793c8566ab (605607:605715) by chromium-webrtc-autoroll · 6 years ago
  39. 8ffd710 Update Android encoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  40. 020e583 AEC3: Compensate comfort noise level for loss due to filter bank by Gustaf Ullberg · 6 years ago
  41. 83b00f0 AEC3: Computation of comfort noise gains from suppression gains by Gustaf Ullberg · 6 years ago
  42. 34fc346 Add support for computing iOS code coverage by Artem Titarenko · 6 years ago
  43. 277b6ea Isolating APM API build target: adding dummy :api target. by Alessio Bazzica · 6 years ago
  44. 3ddaf3c Revert "Add support for screen sharing with PipeWire on Wayland" by Patrik Höglund · 6 years ago
  45. 82c07ea Tune huge video frames detection threshold for GetStats googHugeFramesSent stat by Ilya Nikolaevskiy · 6 years ago
  46. 4f3cc6e Make VideoSendStreamTest.NoPaddingWhenVideoIsMuted less flaky by Erik Språng · 6 years ago
  47. a8f5461 nit: Use make_unique in rtp_video_stream_receiver.cc by Elad Alon · 6 years ago
  48. 27f3172 Simplify use of events in TestAudioDevice by Niels Möller · 6 years ago
  49. 361dbc1 Android: Add option to set presentation timestamp in EglRenderer by Magnus Jedvert · 6 years ago
  50. 967f7d5 Add audio level to CSRC class by Jonas Oreland · 6 years ago
  51. df351f4 Update FakeEncoder to use EncoderInfo by Erik Språng · 6 years ago
  52. 254d3db Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc by tzik · 6 years ago
  53. fbf1683 Add HdrMetadata to VideoFrame by Johannes Kron · 6 years ago
  54. 4f0f3d5 Remove unused member variable - RTCPSender::using_nack_ by Elad Alon · 6 years ago
  55. 63ada78 Remove outdated TODO by Sam Zackrisson · 6 years ago
  56. 3ea1878 Add severity into RTC logging callbacks by Jiawei Ou · 6 years ago
  57. edfb883 Roll chromium_revision 11d7305a72..7841106b37 (605505:605607) by chromium-webrtc-autoroll · 6 years ago
  58. d7db17b Roll chromium_revision bf7ad46dee..11d7305a72 (605401:605505) by chromium-webrtc-autoroll · 6 years ago
  59. a9bbd86 Add a configuration parameter for using the media transport for data channels. by Bjorn Mellem · 6 years ago
  60. 41b5296 Roll chromium_revision c26ff44a53..bf7ad46dee (605286:605401) by chromium-webrtc-autoroll · 6 years ago
  61. ee49f70 Remove VideoEncoder::SetChannelParameters. by philipel · 6 years ago
  62. c22f551 Remove locks from AECM and move it into private_submodules_ by Sam Zackrisson · 6 years ago
  63. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  64. 0070655 Removing ancient and unused test scripts and data files by Henrik Lundin · 6 years ago
  65. fd1a2fb Set RtpRtcp config receive_only in voe::ChannelReceive by Niels Möller · 6 years ago
  66. aed3070 Replace GetScalingSettings() with GetEnocderInfo() in TestEncoder by Erik Språng · 6 years ago
  67. f418bcb Refactor RtpSender to use absl::string_view for payload name. by Niels Möller · 6 years ago
  68. 2634199 Move MovingAverage to rtc_base/numerics and update it. by Ilya Nikolaevskiy · 6 years ago
  69. a1ead6f Update EncoderProxy to use EncoderInfo by Erik Språng · 6 years ago
  70. bf0d0c1 Add IPv6 configuration parameters to iOS API by Uladzislau Susha · 6 years ago
  71. 842a2a8 Roll chromium_revision 4e7c87b55c..c26ff44a53 (605184:605286) by chromium-webrtc-autoroll · 6 years ago
  72. e7547d5 Move MemoryStream to separate source files, and to a test target. by Niels Möller · 6 years ago
  73. 9f878f6 Roll chromium_revision b58a03341b..4e7c87b55c (605082:605184) by chromium-webrtc-autoroll · 6 years ago
  74. 671341a Roll chromium_revision 35f882550d..b58a03341b (604980:605082) by chromium-webrtc-autoroll · 6 years ago
  75. 1bc0b9d Roll chromium_revision e842ab5f98..35f882550d (604874:604980) by chromium-webrtc-autoroll · 6 years ago
  76. 2039ee7 Revert "Delete rtc::Pathname" by Qingsi Wang · 6 years ago
  77. 273d029 Implement data channel methods in LoopbackMediaTransport. by Bjorn Mellem · 6 years ago
  78. 0367d1a Adds a field trial parameter to configure waiting time before sending Nack packets. by Ying Wang · 6 years ago
  79. e401863 Change to RtcEvent::Copy by Elad Alon · 6 years ago
  80. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  81. 42e7d9c Enable rtc event log in *_loopback tools running with renderers by Ilya Nikolaevskiy · 6 years ago
  82. f8ba95e Add field trial for vp8 cpu speed configuration for arm. by Åsa Persson · 6 years ago
  83. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  84. 6bf2054 Roll chromium_revision 734e273d43..e842ab5f98 (604373:604874) by chromium-webrtc-autoroll · 6 years ago
  85. aa3c1cc Delete _strnicmp. Uses replaced with abseil functions. by Niels Möller · 6 years ago
  86. 41f00de Fix chromium roll by Artem Titov · 6 years ago
  87. 6b9dec0 Delete rtc::Pathname by Niels Möller · 6 years ago
  88. d4a68bd Implement Injectable Audio Codecs for the Java SDK. by Lennart Kolmodin · 6 years ago
  89. 3e4c77f Fix AGC2 fixed-adaptive gain controllers order. by Alessio Bazzica · 6 years ago
  90. 096d016 Update MultiplexEncoderAdapter to use EncoderInfo by Erik Språng · 6 years ago
  91. 58df0ad Add guards to VideoCaptureDS::Init for when pins are null by Andreas Pehrson · 6 years ago
  92. 9b5b070 Use EncoderInfo in SimulcastEncoderAdapter by Erik Språng · 6 years ago
  93. 4eb4112 Plug-in media transport state listener by Piotr (Peter) Slatala · 6 years ago
  94. 189013b Update QualityTestVideoEncoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  95. 449afd9 Updated ScopedVideoEncoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  96. 5e78461 Make the extra seturation margin configurable. by Alex Loiko · 6 years ago
  97. b1e031a JitterEstimator: Remove old LowRate exp and add trial for upper bound. by Erik Språng · 6 years ago
  98. 15ca5a9 Add implicit conversion between rtc:PacketTime and int64_t. by Niels Möller · 6 years ago
  99. 96965ae Add ability to enable frame dumping decoder via field trial. by Erik Språng · 6 years ago
  100. fe45da4 Remove WebRTC-VP8-GfBoost field trial. by philipel · 6 years ago