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gerrit-public.fairphone.software / platform / external / webrtc / 5f0099596439a34a7534bf5557423db1f622989f
  1. 5f00995 Using unit classes in AimdRateControl. by Sebastian Jansson · 6 years ago
  2. 50b8426 Roll chromium_revision 2f3cca903d..7d53bc243c (609431:609559) by chromium-webrtc-autoroll · 6 years ago
  3. f85b6d2 Roll chromium_revision 9508bd7fec..2f3cca903d (609314:609431) by chromium-webrtc-autoroll · 6 years ago
  4. b6787bc Using data unit classes in DelayBasedBwe. by Sebastian Jansson · 6 years ago
  5. 2e0c655 [Sanitizers] Don't retry failed tests. by Yves Gerey · 6 years ago
  6. b22f077 Adds FieldTrialConstrained class. by Sebastian Jansson · 6 years ago
  7. 76f5750 Roll chromium_revision 3efc758c50..9508bd7fec (609210:609314) by chromium-webrtc-autoroll · 6 years ago
  8. 85340ce Move rtc::scoped_refptr to api/. by Mirko Bonadei · 6 years ago
  9. 52e69d7 Explicitly specify color space enum indices by Johannes Kron · 6 years ago
  10. 3a83748 New loss-based bandwidth control mechanism. by Christoffer Rodbro · 6 years ago
  11. 26e88b0 Replace RTC_DCHECK by RTC_DCHECK_RUN_ON for worker thread. by Niels Möller · 6 years ago
  12. 2058d52 Disabling test StunPortTest.TestPrepareAddressHostname on WIN. by Alex Loiko · 6 years ago
  13. eb13484 Remove ChannelSendState by Fredrik Solenberg · 6 years ago
  14. c3313a3 Make api:create_peerconnection_factory public. by Mirko Bonadei · 6 years ago
  15. c5e8be3 Remove ChannelReceiveState by Fredrik Solenberg · 6 years ago
  16. 72bba62 Adds shared base class for data units. by Sebastian Jansson · 6 years ago
  17. d474672 Make rtc_event_log protos publicly visible. by Mirko Bonadei · 6 years ago
  18. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  19. 3cf8f3e Adding empty api:create_peerconnection_factory. by Mirko Bonadei · 6 years ago
  20. 2ee41fe Disabling test StunPortTest.TestPrepareAddressHostname on WIN. by Alex Loiko · 6 years ago
  21. 95adedb Always compile VP9 source files. by Mirko Bonadei · 6 years ago
  22. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  23. 601504c in RtcpTransceiver remove workaround for old bug in RtcpReceiver by Danil Chapovalov · 6 years ago
  24. c3bd2fb Roll chromium_revision 92e84c81c1..3efc758c50 (608282:609210) by chromium-webrtc-autoroll · 6 years ago
  25. 0a8bd9c Adds clamping to TimeDelta. by Sebastian Jansson · 6 years ago
  26. b5f8201 Adds scalar division to DataRate. by Sebastian Jansson · 6 years ago
  27. 8ef5793 Switch from RTC_DISABLE_VP9 to RTC_ENABLE_VP9. by Mirko Bonadei · 6 years ago
  28. bd6ffaf Fix small issues that stops the Chromium DEPS roll. by Patrik Höglund · 6 years ago
  29. 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
  30. 8c1e73b Don't add empty extension list in event log parser. by Sebastian Jansson · 6 years ago
  31. 1eebec9 Fix data race in channel_send.cc by Piotr (Peter) Slatala · 6 years ago
  32. b5bb513 Disable RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection by Yves Gerey · 6 years ago
  33. 6eb8a16 Exposing audio and video engines directly. by Sebastian Jansson · 6 years ago
  34. eee3920 Don't poll EncoderInfo from encoder twice per frame by Erik Språng · 6 years ago
  35. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  36. a32d7e2 Add default values for PlayoutDelay in RTPVideoHeader. by Niels Möller · 6 years ago
  37. 7dbb7c3 Adding missing build target for audio_device_default. by Mirko Bonadei · 6 years ago
  38. fa0aa39 Removes templating from CompositeMediaEngine. by Sebastian Jansson · 6 years ago
  39. 84848f2 Adds interfaces for audio and video engines. by Sebastian Jansson · 6 years ago
  40. 2681523 Tweak ChannelSend interface, to make it closer to ChannelSendProxy by Niels Möller · 6 years ago
  41. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  42. 25a3a97 Android: ignore LintError for absent class files by Artem Titarenko · 6 years ago
  43. 3021342 Adding more owners to p2p by Jeroen de Borst · 6 years ago
  44. cc8e8bb Pass the media transport from JsepTransportController to Call. by Piotr (Peter) Slatala · 6 years ago
  45. 86336a5 Update FakeVp8Encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  46. 10aeb2a MediaTransportTests should use audio-only peer connection. by Piotr (Peter) Slatala · 6 years ago
  47. 0462948 Revert "Add ios bindings for PeerConnectionState." by Jonas Olsson · 6 years ago
  48. e78b465 Add version and UTC time fields to RTC event log. by Bjorn Terelius · 6 years ago
  49. f0db2e2 nit: Missing space in build_overrides/build.gni by Elad Alon · 6 years ago
  50. dd886082 AGC2 flags: remove deprecated fields. by Alessio Bazzica · 6 years ago
  51. a06bf85 Add a presubmit check for absl/memory/memory.h inclusion by tzik · 6 years ago
  52. 9514071 Android: Support externally aligned timestamps by Magnus Jedvert · 6 years ago
  53. 2277ac6 Adds OWNERS to rtc_base/experiments. by Sebastian Jansson · 6 years ago
  54. f01d8c8 Add android bindings for PeerConnectionState. by Jonas Olsson · 6 years ago
  55. 586725d Add ios bindings for PeerConnectionState. by Jonas Olsson · 6 years ago
  56. 58376f3 Make member internal::SynchronousMethodCall::e_ a non-pointer. by Niels Möller · 6 years ago
  57. a859d41 Increasing visibility of api/transport build targets. by Mirko Bonadei · 6 years ago
  58. dbb988b Change ReceiveStatistics to implement RtpPacketSinkInterface, part 2. by Niels Möller · 6 years ago
  59. 7af4ac8 Roll chromium_revision 4ffd688e44..92e84c81c1 (608180:608282) by chromium-webrtc-autoroll · 6 years ago
  60. c25d234 Adds OWNERS to api/transport. by Sebastian Jansson · 6 years ago
  61. d575a2d Roll chromium_revision 3d76a59d7d..4ffd688e44 (608069:608180) by chromium-webrtc-autoroll · 6 years ago
  62. 30599b0 Roll chromium_revision fbed28d429..3d76a59d7d (607938:608069) by chromium-webrtc-autoroll · 6 years ago
  63. b1e4775 Exposing rtcp report interval setting in objc api by Jiawei Ou · 6 years ago
  64. 83aa5ac Adding Microsoft Corporation (*@microsoft.com) to WebRTC AUTHORS by James Cadd · 6 years ago
  65. dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
  66. 1724a80 AEC3: Turn off the specific suppressor mode for stationary render by Per Åhgren · 6 years ago
  67. cc55032 Adding shampson to media/OWNERS. by Seth Hampson · 6 years ago
  68. 2464348 Don't reset RTT Backoff timeout on route change. by Sebastian Jansson · 6 years ago
  69. fdc635d Remove deprecated APIs from RTC event log parser. by Bjorn Terelius · 6 years ago
  70. 3bc696f Android EglRenderer: Replace unicoce character with ascii character by Magnus Jedvert · 6 years ago
  71. 76f9954 Remove the old RTC event log parser. by Bjorn Terelius · 6 years ago
  72. 38578ca Roll chromium_revision db720b4ab9..fbed28d429 (606025:607938) by chromium-webrtc-autoroll · 6 years ago
  73. a038e71 Less strict audio codec tests to accomodate opus switch to SSE. by Yves Gerey · 6 years ago
  74. fb6fd4b Fix lint errors for android manifests. by Yves Gerey · 6 years ago
  75. 6ef89e7 Rectify comment about 'build_with_chromium'. by Mirko Bonadei · 6 years ago
  76. c58c8a5 Adding mbonadei@ to build_overrides/OWNERS. by Mirko Bonadei · 6 years ago
  77. 42b715a Add visibility to ana config proto by Piotr (Peter) Slatala · 6 years ago
  78. 6dbf0e4 Remove all aliases to rtc::Thread by Danil Chapovalov · 6 years ago
  79. 428a160 Remove rtc_event_log2text by Bjorn Terelius · 6 years ago
  80. 95ca6e1 AudioSource allows implementations to return settings by Piotr (Peter) Slatala · 6 years ago
  81. bc4cf89 Run some peer connection end-to-end tests with an empty audio encoder factory by Karl Wiberg · 6 years ago
  82. de8e6e6 Refactor bitrate configuration in CallTest by Niels Möller · 6 years ago
  83. c7e3af1 Remove rtc_event_log2stats. by Bjorn Terelius · 6 years ago
  84. 8544799 Introduce DLOG to video and voiceengine. by Jonas Olsson · 6 years ago
  85. 318da51 Reland "Add support for screen sharing with PipeWire on Wayland" by Tomas Popela · 6 years ago
  86. 1e2542f AGC2: adding level estimation option (RMS or peak-based). by Alessio Bazzica · 6 years ago
  87. 44ca9a3 Allow usage of stringstream under examples/. by Mirko Bonadei · 6 years ago
  88. 105edca Remove some unused RentACodec static methods by Karl Wiberg · 6 years ago
  89. a33c895 AEC3: Corrected erroneous if-statement that always returned true by Per Åhgren · 6 years ago
  90. b739666 Add missing include of unistd.h by Niels Möller · 6 years ago
  91. 90e6745 Delete deprecated class WrappedI420Buffer by Niels Möller · 6 years ago
  92. f4ce0e4 Configs to run slow_tests. by Mirko Bonadei · 6 years ago
  93. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  94. fd20171 Adds setup of RTP Extensions in Scenario tests. by Sebastian Jansson · 6 years ago
  95. cb7eddb Add tests for cpu overuse scaling. by Åsa Persson · 6 years ago
  96. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  97. 4aeb35b Explicitly retain self in objc blocks to avoid compiler warning. by Jiawei Ou · 6 years ago
  98. 0c32e33 Allows change of fake encoder max rate in scenarios tests. by Sebastian Jansson · 6 years ago
  99. 985ee68 Add support for screenshare content type in scenario tests. by Sebastian Jansson · 6 years ago
  100. 2b101d2 Simplifies audio priority rate config in scenario tests. by Sebastian Jansson · 6 years ago