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gerrit-public.fairphone.software / platform / external / webrtc / 8b5d9d86508835435f34b7b031d76d8ec4fec1b5
  1. 8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
  2. 4a2dd7a Roll chromium_revision 5825fead7b..6931f4c0d0 (610108:610209) by chromium-webrtc-autoroll · 6 years ago
  3. 540ef28 Adds OnReceivedUplinkAllocation method to AudioEncoder. by Sebastian Jansson · 6 years ago
  4. 6736df1 Moves BitrateAllocationUpdate to api. by Sebastian Jansson · 6 years ago
  5. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  6. e4cccae Removed ability to set CryptoOptions through PeerConnectionFactory from bindings. by Benjamin Wright · 6 years ago
  7. a526ae6 Roll chromium_revision 92f8c5b2a2..5825fead7b (609994:610108) by chromium-webrtc-autoroll · 6 years ago
  8. 5eae1d9 Remove legacy SetTargetTransferRateObserver by Piotr (Peter) Slatala · 6 years ago
  9. 37227be Add check for media transport and bundle policy by Piotr (Peter) Slatala · 6 years ago
  10. 47dfdca Create 'MaybeCreateMediaTransport' function by Piotr (Peter) Slatala · 6 years ago
  11. 64bfcde Add sakal@ to OWNERS in android tests / aarproject directories. by Sami Kalliomäki · 6 years ago
  12. 4749e4e Move HdrMetadata to ColorSpace by Johannes Kron · 6 years ago
  13. ecf6315 AGC2 adaptive digital: remove unnecessary flag. by Alessio Bazzica · 6 years ago
  14. 8da7b35 AGC2 adaptive digital false by default by Alessio Bazzica · 6 years ago
  15. cfddbb7 Add ios bindings for PeerConnectionState. by Jonas Olsson · 6 years ago
  16. 49a7843 Don't restart streams in scenario tests. by Sebastian Jansson · 6 years ago
  17. 0e4dfcb Roll chromium_revision 16e6b25329..92f8c5b2a2 (609893:609994) by chromium-webrtc-autoroll · 6 years ago
  18. 59a01b0 Set Framerate in RTCVideoEncoderH264 by Qiang Chen · 6 years ago
  19. 2b5b0e9 Disabling ScreenDrawerTest.TwoScreenDrawerLocks by Alex Loiko · 6 years ago
  20. c4d5642 Revert "Default to dlopening the PipeWire." by Oleh Prypin · 6 years ago
  21. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  22. a13be01 Default to dlopening the PipeWire. by Tomas Popela · 6 years ago
  23. c68d282 Add test PeerConnectionIntegrationTest.MediaTransportBidirectionalAudio by Niels Möller · 6 years ago
  24. 89c94b9 Adds target bandwidth to BitrateAllocator. by Sebastian Jansson · 6 years ago
  25. 66eedce Roll chromium_revision 7d53bc243c..16e6b25329 (609559:609893) by chromium-webrtc-autoroll · 6 years ago
  26. bd04f4a Increase buffer level threshold in VP8/9 tests. by Sergey Silkin · 6 years ago
  27. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  28. 38332cd Add RTCP and simulcast support for RTCRtpReceiver::getParameters() by Florent Castelli · 6 years ago
  29. 4bc6045 Add output directory option for audioproc_f data dump files. by Alessio Bazzica · 6 years ago
  30. 388e4e9 Make RTC_LOG_FILE_LINE use its parameters by Jonas Olsson · 6 years ago
  31. c20b82a Remove unused variables in RtcEventAudioXStreamConfig::Copy() by Bjorn Terelius · 6 years ago
  32. 22b70ff Move VideoCodecType from common_types.h to api/video/video_codec_type.h by Niels Möller · 6 years ago
  33. 22ff1a4 Fix threshold in VideoCodecTestLibvpx.ChangeFramerateVP9. by Mirko Bonadei · 6 years ago
  34. 6817038 APM audioproc_f: flag for AGC2 adaptive level estimator. by Alessio Bazzica · 6 years ago
  35. 44974e1 AEC3: Adding a correction factor for the Erle estimation that depends on the portion of the filter that is currently in use. by Jesús de Vicente Peña · 6 years ago
  36. 985a1f3 Add const or GUARDED_BY on a few ChannelSend members by Niels Möller · 6 years ago
  37. 5f00995 Using unit classes in AimdRateControl. by Sebastian Jansson · 6 years ago
  38. 50b8426 Roll chromium_revision 2f3cca903d..7d53bc243c (609431:609559) by chromium-webrtc-autoroll · 6 years ago
  39. f85b6d2 Roll chromium_revision 9508bd7fec..2f3cca903d (609314:609431) by chromium-webrtc-autoroll · 6 years ago
  40. b6787bc Using data unit classes in DelayBasedBwe. by Sebastian Jansson · 6 years ago
  41. 2e0c655 [Sanitizers] Don't retry failed tests. by Yves Gerey · 6 years ago
  42. b22f077 Adds FieldTrialConstrained class. by Sebastian Jansson · 6 years ago
  43. 76f5750 Roll chromium_revision 3efc758c50..9508bd7fec (609210:609314) by chromium-webrtc-autoroll · 6 years ago
  44. 85340ce Move rtc::scoped_refptr to api/. by Mirko Bonadei · 6 years ago
  45. 52e69d7 Explicitly specify color space enum indices by Johannes Kron · 6 years ago
  46. 3a83748 New loss-based bandwidth control mechanism. by Christoffer Rodbro · 6 years ago
  47. 26e88b0 Replace RTC_DCHECK by RTC_DCHECK_RUN_ON for worker thread. by Niels Möller · 6 years ago
  48. 2058d52 Disabling test StunPortTest.TestPrepareAddressHostname on WIN. by Alex Loiko · 6 years ago
  49. eb13484 Remove ChannelSendState by Fredrik Solenberg · 6 years ago
  50. c3313a3 Make api:create_peerconnection_factory public. by Mirko Bonadei · 6 years ago
  51. c5e8be3 Remove ChannelReceiveState by Fredrik Solenberg · 6 years ago
  52. 72bba62 Adds shared base class for data units. by Sebastian Jansson · 6 years ago
  53. d474672 Make rtc_event_log protos publicly visible. by Mirko Bonadei · 6 years ago
  54. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  55. 3cf8f3e Adding empty api:create_peerconnection_factory. by Mirko Bonadei · 6 years ago
  56. 2ee41fe Disabling test StunPortTest.TestPrepareAddressHostname on WIN. by Alex Loiko · 6 years ago
  57. 95adedb Always compile VP9 source files. by Mirko Bonadei · 6 years ago
  58. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  59. 601504c in RtcpTransceiver remove workaround for old bug in RtcpReceiver by Danil Chapovalov · 6 years ago
  60. c3bd2fb Roll chromium_revision 92e84c81c1..3efc758c50 (608282:609210) by chromium-webrtc-autoroll · 6 years ago
  61. 0a8bd9c Adds clamping to TimeDelta. by Sebastian Jansson · 6 years ago
  62. b5f8201 Adds scalar division to DataRate. by Sebastian Jansson · 6 years ago
  63. 8ef5793 Switch from RTC_DISABLE_VP9 to RTC_ENABLE_VP9. by Mirko Bonadei · 6 years ago
  64. bd6ffaf Fix small issues that stops the Chromium DEPS roll. by Patrik Höglund · 6 years ago
  65. 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
  66. 8c1e73b Don't add empty extension list in event log parser. by Sebastian Jansson · 6 years ago
  67. 1eebec9 Fix data race in channel_send.cc by Piotr (Peter) Slatala · 6 years ago
  68. b5bb513 Disable RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection by Yves Gerey · 6 years ago
  69. 6eb8a16 Exposing audio and video engines directly. by Sebastian Jansson · 6 years ago
  70. eee3920 Don't poll EncoderInfo from encoder twice per frame by Erik Språng · 6 years ago
  71. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  72. a32d7e2 Add default values for PlayoutDelay in RTPVideoHeader. by Niels Möller · 6 years ago
  73. 7dbb7c3 Adding missing build target for audio_device_default. by Mirko Bonadei · 6 years ago
  74. fa0aa39 Removes templating from CompositeMediaEngine. by Sebastian Jansson · 6 years ago
  75. 84848f2 Adds interfaces for audio and video engines. by Sebastian Jansson · 6 years ago
  76. 2681523 Tweak ChannelSend interface, to make it closer to ChannelSendProxy by Niels Möller · 6 years ago
  77. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  78. 25a3a97 Android: ignore LintError for absent class files by Artem Titarenko · 6 years ago
  79. 3021342 Adding more owners to p2p by Jeroen de Borst · 6 years ago
  80. cc8e8bb Pass the media transport from JsepTransportController to Call. by Piotr (Peter) Slatala · 6 years ago
  81. 86336a5 Update FakeVp8Encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  82. 10aeb2a MediaTransportTests should use audio-only peer connection. by Piotr (Peter) Slatala · 6 years ago
  83. 0462948 Revert "Add ios bindings for PeerConnectionState." by Jonas Olsson · 6 years ago
  84. e78b465 Add version and UTC time fields to RTC event log. by Bjorn Terelius · 6 years ago
  85. f0db2e2 nit: Missing space in build_overrides/build.gni by Elad Alon · 6 years ago
  86. dd886082 AGC2 flags: remove deprecated fields. by Alessio Bazzica · 6 years ago
  87. a06bf85 Add a presubmit check for absl/memory/memory.h inclusion by tzik · 6 years ago
  88. 9514071 Android: Support externally aligned timestamps by Magnus Jedvert · 6 years ago
  89. 2277ac6 Adds OWNERS to rtc_base/experiments. by Sebastian Jansson · 6 years ago
  90. f01d8c8 Add android bindings for PeerConnectionState. by Jonas Olsson · 6 years ago
  91. 586725d Add ios bindings for PeerConnectionState. by Jonas Olsson · 6 years ago
  92. 58376f3 Make member internal::SynchronousMethodCall::e_ a non-pointer. by Niels Möller · 6 years ago
  93. a859d41 Increasing visibility of api/transport build targets. by Mirko Bonadei · 6 years ago
  94. dbb988b Change ReceiveStatistics to implement RtpPacketSinkInterface, part 2. by Niels Möller · 6 years ago
  95. 7af4ac8 Roll chromium_revision 4ffd688e44..92e84c81c1 (608180:608282) by chromium-webrtc-autoroll · 6 years ago
  96. c25d234 Adds OWNERS to api/transport. by Sebastian Jansson · 6 years ago
  97. d575a2d Roll chromium_revision 3d76a59d7d..4ffd688e44 (608069:608180) by chromium-webrtc-autoroll · 6 years ago
  98. 30599b0 Roll chromium_revision fbed28d429..3d76a59d7d (607938:608069) by chromium-webrtc-autoroll · 6 years ago
  99. b1e4775 Exposing rtcp report interval setting in objc api by Jiawei Ou · 6 years ago
  100. 83aa5ac Adding Microsoft Corporation (*@microsoft.com) to WebRTC AUTHORS by James Cadd · 6 years ago