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gerrit-public.fairphone.software / platform / external / webrtc / 89f874eb39d668d2a71f45ad54542a4a88a806a6
  1. 89f874e Add offer_extmap_allow_mixed to RTCConfiguration by Johannes Kron · 6 years ago
  2. 5ae3a02 Revert "Run robolectric tests for Android on several Android API versions" by Danil Chapovalov · 6 years ago
  3. 20f60f0 Fuzzer crash in AGC2. by Alex Loiko · 6 years ago
  4. cfe3b6a Remove most of api/ortc/. by Jonas Olsson · 6 years ago
  5. 8584667 Fix overflow for high bitrates in BitrateProber by Johannes Kron · 6 years ago
  6. 09102a0 Revert "Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus."" by Yves Gerey · 6 years ago
  7. 0b1b5c1 Hide RtcEvent members behind accessors by Elad Alon · 6 years ago
  8. eb809f3 Event logs - separate audio_level and voice_activity by Elad Alon · 6 years ago
  9. 466620b Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus." by Yves Gerey · 6 years ago
  10. 56a4b32 Rename fields in rtc_event_log2.proto by Elad Alon · 6 years ago
  11. a2eb0a7 Fix up an outdated comment in peerconnection_integrationtest.cc. by Bjorn Mellem · 6 years ago
  12. 7127f34 Signal Network route change in fake ice. by Piotr (Peter) Slatala · 6 years ago
  13. d95b0a2 Use delta-encoding in new WebRTC event logs by Elad Alon · 6 years ago
  14. 7246720 Clean up root OWNERS. by Patrik Höglund · 6 years ago
  15. e598e6b Run robolectric tests for Android on several Android API versions by Artem Titarenko · 6 years ago
  16. 9973fa8 Pass HdrMetadata between VideoFrame and EncodedImage for VP9 by Johannes Kron · 6 years ago
  17. 6c373cc Add support for audio in latency visualization. by Bjorn Terelius · 6 years ago
  18. d8aa9f9 Fix flaky JsepTransportControllerTests. by Jonas Olsson · 6 years ago
  19. ad1d9f0 Add RTP header extension for HDR metadata by Johannes Kron · 6 years ago
  20. ee45f90 In RTP to NTP estimator do not allow huge jumps in NTP timestamps by Ilya Nikolaevskiy · 6 years ago
  21. 06f6bc9 Reintroduce missing dependencies in libwebrtc.a library. by Yves Gerey · 6 years ago
  22. 175aa2e Implement data channels over media transport. by Bjorn Mellem · 6 years ago
  23. c2ebe21 Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" by Jiawei Ou · 6 years ago
  24. 0393c64 [Win/boringSSL] Add nasm as part of required dependencies. by Yves Gerey · 6 years ago
  25. ada077f Callback changes to media transport interface: by Piotr (Peter) Slatala · 6 years ago
  26. 87e1619 Add owners for media_transport_interface by Piotr (Peter) Slatala · 6 years ago
  27. d3438aa Add ability to specify if rate controller of video encoder is trusted. by Erik Språng · 6 years ago
  28. 6528d8a In Android encoders, cache EncoderInfo in InitEncode. by Erik Språng · 6 years ago
  29. 260770c Delete rtc::Filesystem. Move needed functions to filerotatingstream.cc. by Niels Möller · 6 years ago
  30. b0550bd Eliminate use of EventWrapper from mac audio device by Niels Möller · 6 years ago
  31. c94b22e Add magjed/nisse/sprang/brandtr as api/video_codecs owners by Erik Språng · 6 years ago
  32. c5dd300 Introduce RtpPacket::GetExtension accessor that return result by Danil Chapovalov · 6 years ago
  33. 357f596 Split a separate codecs target off of :video_jni by Jonathan Yu · 6 years ago
  34. 5bb1ed6 Eliminate use of EventWrapper from ios audio device tests by Niels Möller · 6 years ago
  35. a33c7af Tolerate optional chunks in WAV files by Alessio Bazzica · 6 years ago
  36. c496d58 Add flag for fast jitter buffer playout in neteq simulation by Sam Zackrisson · 6 years ago
  37. e6c2c08 MsanUninitialized: restric type check to msan case. by Alessio Bazzica · 6 years ago
  38. c4e9825 Delete classes EventFactory and EventFactoryImpl. by Niels Möller · 6 years ago
  39. 2a74263 Make the bitrate_allocator param optional to prepare for its removal by Oleh Prypin · 6 years ago
  40. cd2e105 Reenable test RampUpTest.AudioTransportSequenceNumber by Niels Möller · 6 years ago
  41. 694ed17 Add a style rule about not using const optional<T>& arguments by Karl Wiberg · 6 years ago
  42. f0e7440 Add missing conditional defines to neteq test and tools targets by Sam Zackrisson · 6 years ago
  43. 689983f Deprecate EventFactory and delete all usage. by Niels Möller · 6 years ago
  44. 54b4924 Update H264 encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  45. 1060870 Update LibVpxVp8Encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  46. 727d164 Update VP9 encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  47. 5473a45 Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities by Florent Castelli · 6 years ago
  48. 75de46a Update SimulcastEncoderAdapter merging of EncoderInfo by Erik Språng · 6 years ago
  49. e6a2d94 Clear FrameBuffer if there were no frames received for 10 minutes by Ilya Nikolaevskiy · 6 years ago
  50. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  51. bdc6c40 Add field trial for target bitrate RTCP XR message. by Rasmus Brandt · 6 years ago
  52. d565918 Delete NullEventFactory by Niels Möller · 6 years ago
  53. e769ed9 Roll chromium_revision 38dcb5ed01..db720b4ab9 (605924:606025) by chromium-webrtc-autoroll · 6 years ago
  54. 50f60cb Rename software codec classes and move them into api/ by Jonathan Yu · 6 years ago
  55. ff7020a Remove non-default VideoEncoder::EncoderInfo() ctor by Erik Språng · 6 years ago
  56. 36d907b Update MockVideoEncoder with correct methods. by Erik Språng · 6 years ago
  57. 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  58. a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
  59. 7553c02 Update ObjCVideoEncoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  60. 7b3c76b Reland "Delete rtc::Pathname" by Niels Möller · 6 years ago
  61. 17fc7e2 Add counter to the end of FakeEncoder frames in order to make them unique. by Per Kjellander · 6 years ago
  62. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  63. 3ea7b83 Resolve the race condition between mDNS name registration and by Qingsi Wang · 6 years ago
  64. 8770ce7 Roll chromium_revision 03cf97f6d8..38dcb5ed01 (605818:605924) by chromium-webrtc-autoroll · 6 years ago
  65. bb091db Roll chromium_revision 793c8566ab..03cf97f6d8 (605715:605818) by chromium-webrtc-autoroll · 6 years ago
  66. 2cd3b4c Fixing bug in SimulatedNetwork where packets stop. by Sebastian Jansson · 6 years ago
  67. 0f54f21 Removes deprecated GetSentPacket from PacketResult. by Sebastian Jansson · 6 years ago
  68. dc98b9b AEC3: Corrected include by Per Åhgren · 6 years ago
  69. c564a7b Roll chromium_revision 7841106b37..793c8566ab (605607:605715) by chromium-webrtc-autoroll · 6 years ago
  70. 8ffd710 Update Android encoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  71. 020e583 AEC3: Compensate comfort noise level for loss due to filter bank by Gustaf Ullberg · 6 years ago
  72. 83b00f0 AEC3: Computation of comfort noise gains from suppression gains by Gustaf Ullberg · 6 years ago
  73. 34fc346 Add support for computing iOS code coverage by Artem Titarenko · 6 years ago
  74. 277b6ea Isolating APM API build target: adding dummy :api target. by Alessio Bazzica · 6 years ago
  75. 3ddaf3c Revert "Add support for screen sharing with PipeWire on Wayland" by Patrik Höglund · 6 years ago
  76. 82c07ea Tune huge video frames detection threshold for GetStats googHugeFramesSent stat by Ilya Nikolaevskiy · 6 years ago
  77. 4f3cc6e Make VideoSendStreamTest.NoPaddingWhenVideoIsMuted less flaky by Erik Språng · 6 years ago
  78. a8f5461 nit: Use make_unique in rtp_video_stream_receiver.cc by Elad Alon · 6 years ago
  79. 27f3172 Simplify use of events in TestAudioDevice by Niels Möller · 6 years ago
  80. 361dbc1 Android: Add option to set presentation timestamp in EglRenderer by Magnus Jedvert · 6 years ago
  81. 967f7d5 Add audio level to CSRC class by Jonas Oreland · 6 years ago
  82. df351f4 Update FakeEncoder to use EncoderInfo by Erik Språng · 6 years ago
  83. 254d3db Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc by tzik · 6 years ago
  84. fbf1683 Add HdrMetadata to VideoFrame by Johannes Kron · 6 years ago
  85. 4f0f3d5 Remove unused member variable - RTCPSender::using_nack_ by Elad Alon · 6 years ago
  86. 63ada78 Remove outdated TODO by Sam Zackrisson · 6 years ago
  87. 3ea1878 Add severity into RTC logging callbacks by Jiawei Ou · 6 years ago
  88. edfb883 Roll chromium_revision 11d7305a72..7841106b37 (605505:605607) by chromium-webrtc-autoroll · 6 years ago
  89. d7db17b Roll chromium_revision bf7ad46dee..11d7305a72 (605401:605505) by chromium-webrtc-autoroll · 6 years ago
  90. a9bbd86 Add a configuration parameter for using the media transport for data channels. by Bjorn Mellem · 6 years ago
  91. 41b5296 Roll chromium_revision c26ff44a53..bf7ad46dee (605286:605401) by chromium-webrtc-autoroll · 6 years ago
  92. ee49f70 Remove VideoEncoder::SetChannelParameters. by philipel · 6 years ago
  93. c22f551 Remove locks from AECM and move it into private_submodules_ by Sam Zackrisson · 6 years ago
  94. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  95. 0070655 Removing ancient and unused test scripts and data files by Henrik Lundin · 6 years ago
  96. fd1a2fb Set RtpRtcp config receive_only in voe::ChannelReceive by Niels Möller · 6 years ago
  97. aed3070 Replace GetScalingSettings() with GetEnocderInfo() in TestEncoder by Erik Språng · 6 years ago
  98. f418bcb Refactor RtpSender to use absl::string_view for payload name. by Niels Möller · 6 years ago
  99. 2634199 Move MovingAverage to rtc_base/numerics and update it. by Ilya Nikolaevskiy · 6 years ago
  100. a1ead6f Update EncoderProxy to use EncoderInfo by Erik Språng · 6 years ago