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webrtc / src / / 412dc5f27e74af1bff54408921f49580e6c4f73e
  1. 412dc5f Clean-up of unused PacingBufferPushback feature. by Christoffer Rodbro · 6 years ago
  2. 556258a Fuzzer fix for multistream opus. by Alex Loiko · 6 years ago
  3. f4481c8 Adds GetStats to scenario receive streams. by Sebastian Jansson · 6 years ago
  4. 20789e4 Name changes inside VideoEncoder by Elad Alon · 6 years ago
  5. 0a8562e Forward LossNotification from RTCPReceiver to EncoderRtcpFeedback by Elad Alon · 6 years ago
  6. 1f44bc1 Delete local convenience alias Buffer by Niels Möller · 6 years ago
  7. 00d1adf Make audio_device_module_from_input_and_output visible. by Mirko Bonadei · 6 years ago
  8. 7dd83e2 Revert "Refactor FrameDecryptorInterface::Decrypt to use new API." by Henrik Boström · 6 years ago
  9. 88718fd Fix comment - ice transport is cleared on the networking thread. by Marina Ciocea · 6 years ago
  10. 7a53f66 Delete autoroll config for src/third_party/winsdk_samples by Niels Möller · 6 years ago
  11. f3736ed Datachannel: Use absl::optional for maxRetransmits and maxRetransmitTime. by Harald Alvestrand · 6 years ago
  12. 8581877 Delete interface class VideoCaptureExternal by Niels Möller · 6 years ago
  13. f73f7d6 Add thread safety annotations for some more PeerConnection members (part 13) by Karl Wiberg · 6 years ago
  14. 2108cc6 Added some more debugging logs by Benjamin Wright · 6 years ago
  15. b703db9 Fix and test CreateVideoStreamDecoder by Danil Chapovalov · 6 years ago
  16. 25b9612 Update visibility of java_audio_device_module_jni target by Paulina Hensman · 6 years ago
  17. f71362f Wire up RTCOutboundRtpStreamStats.totalEncodeTime. by Henrik Boström · 6 years ago
  18. ea7b4c5 Roll chromium_revision afc02859a5..6c8246bc15 (648860:648994) by chromium-webrtc-autoroll · 6 years ago
  19. 07f3279 Adding a restriction for legal RID values. by Amit Hilbuch · 6 years ago
  20. 19a94cb Roll chromium_revision 827b96d72e..afc02859a5 (648742:648860) by chromium-webrtc-autoroll · 6 years ago
  21. 642aa81 Refactor FrameDecryptorInterface::Decrypt to use new API. by Benjamin Wright · 6 years ago
  22. 6fdbba3 Roll chromium_revision 7987ebdc29..827b96d72e (648632:648742) by chromium-webrtc-autoroll · 6 years ago
  23. 7237c15 Fixing failing ScenarioTest for IOS ARM64 Debug builds. by Sebastian Jansson · 6 years ago
  24. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  25. e5b9416 Decoder for multistream Opus. by Alex Loiko · 6 years ago
  26. e9d2b4e Revert "Remove old audio device implementation." by Jeroen de Borst · 6 years ago
  27. 57f2a54 Revert "Remove TaskQueue constructor that uses GlobalTaskQueueFactory" by Florent Castelli · 6 years ago
  28. 14d1c9d Rename EncoderKeyFrameCallback back to EncoderRtcpFeedback by Elad Alon · 6 years ago
  29. 8b60e8b Give VideoSendStreamImpl access to RTP timestamps by Elad Alon · 6 years ago
  30. 0cfa4cb Remove old audio device implementation. by Kári Tristan Helgason · 6 years ago
  31. 4c6ca30 Update VideoStreamEncoder to use new VideoEncoder::SetRates() method. by Erik Språng · 6 years ago
  32. e893bfc Roll chromium_revision baccefbc73..7987ebdc29 (648532:648632) by chromium-webrtc-autoroll · 6 years ago
  33. 28d13cb Add RtpSequenceNumberMap::InsertFrame() by Elad Alon · 6 years ago
  34. 3fcc5be Remove unused members in VCMJitterEstimator. by Åsa Persson · 6 years ago
  35. 61e2753 Delete placeholder code for Windows Media Foundation capturer. by Niels Möller · 6 years ago
  36. 800a103 Fix timeout in rtcp_receiver_fuzzer - limit input length by Elad Alon · 6 years ago
  37. f59666b Fix potential bug due to malformed input by Johannes Kron · 6 years ago
  38. 1e2d436 Change PlayoutLatency setLatency zero-threshold value. by Ruslan Burakov · 6 years ago
  39. 8b9f511 Add stream labels into PeerConnection level smoke test. by Artem Titov · 6 years ago
  40. e0b9355 Move enum VideoType out of common_types.h by Niels Möller · 6 years ago
  41. 0d32a73 Fix naming in NetworkEmulationManager: endpoint_controller -> endpoint_container by Artem Titov · 6 years ago
  42. 5a0665b Make UDP receive buffer size configurable via field trial by Johannes Kron · 6 years ago
  43. d1c6085 Added FrameDecryptorInterface::Result constructor and IsOk() member function. by Benjamin Wright · 6 years ago
  44. f948eb6 Implement DefaultAudioQualityAnalyzer. by Mirko Bonadei · 6 years ago
  45. 0b2bf95 Roll chromium_revision 58ec016f06..baccefbc73 (648432:648532) by chromium-webrtc-autoroll · 6 years ago
  46. a5b8220 Roll chromium_revision 0a611f37b8..58ec016f06 (648329:648432) by chromium-webrtc-autoroll · 6 years ago
  47. 0694d1f Roll chromium_revision 37663bcca7..0a611f37b8 (648215:648329) by chromium-webrtc-autoroll · 6 years ago
  48. 72e9771 Add Result FrameDecryptorInterface::Decrypt by Benjamin Wright · 6 years ago
  49. 2a3cf05 Roll chromium_revision f91f825874..37663bcca7 (648096:648215) by chromium-webrtc-autoroll · 6 years ago
  50. ebd94f6 Using simulated time for GoogCC tests. by Sebastian Jansson · 6 years ago
  51. 8f32b6c AEC3: Enable usage of external delay estimator by Gustaf Ullberg · 6 years ago
  52. a553c72 Tune VideoCodecTestLibvpx.TemporalLayersVP8 thresholds. by philipel · 6 years ago
  53. 363fb7e Running scenario quality unit tests in simulated time. by Sebastian Jansson · 6 years ago
  54. 7b7485b Remove TaskQueue constructor that uses GlobalTaskQueueFactory by Danil Chapovalov · 6 years ago
  55. 59e875c Tune VideoCodecTestLibvpx.MultiresVP8 thresholds. by philipel · 6 years ago
  56. c46a999 Reduce flakiness of repeating task test. by Sebastian Jansson · 6 years ago
  57. ae2213b Delete compatibility alias webrtc::kI420 by Niels Möller · 6 years ago
  58. ff39312 Add ability to have multiple connected remote endpoints by Artem Titov · 6 years ago
  59. 5684af5 VideoSendStream::Stats::total_encode_time_ms added. by Henrik Boström · 6 years ago
  60. a556448 Don't recreate the VideoReceiveStream on SetFrameDecryptor in the MediaEngine. by Benjamin Wright · 6 years ago
  61. 144575b Roll chromium_revision e05071635b..f91f825874 (647992:648096) by chromium-webrtc-autoroll · 6 years ago
  62. 4e95e17 Roll chromium_revision b95224d1f4..e05071635b (647872:647992) by chromium-webrtc-autoroll · 6 years ago
  63. 0ae222a Roll chromium_revision 4aff3b8e0f..b95224d1f4 (647765:647872) by chromium-webrtc-autoroll · 6 years ago
  64. 1c747f5 Preparing VideoReceiveStream for move to TaskQueue. by Sebastian Jansson · 6 years ago
  65. f75d458 Roll chromium_revision e839e72818..4aff3b8e0f (647650:647765) by chromium-webrtc-autoroll · 6 years ago
  66. 50b8c39 Generalize the C-language Opus interface. by Alex Loiko · 6 years ago
  67. 21f6fd7 Add preemptive rate and preferred buffer size plots to event log visualizer. by Jakob Ivarsson · 6 years ago
  68. ec51ce0 AEC3: Remove unused config parameters by Gustaf Ullberg · 6 years ago
  69. 6c371ca Add OnLossNotification() to VideoEncoder and Vp8FrameBufferController by Elad Alon · 6 years ago
  70. f88aa97 Introduce RtpSequenceNumberMap by Elad Alon · 6 years ago
  71. 086b907 Update codecs/h264 owners. by Sergey Silkin · 6 years ago
  72. 1770ddb Remove obsolete TODO in default_encoded_image_data_injector.h by Artem Titov · 6 years ago
  73. 2579047 Roll chromium_revision 8524e2aded..e839e72818 (647205:647650) by chromium-webrtc-autoroll · 6 years ago
  74. 4bac79e Add SetJitterBufferMinimumDelay method to RtpReceiverInterface. by Ruslan Burakov · 6 years ago
  75. ada9b89 Added more refined benchmarking code for audioproc_f by Per Åhgren · 6 years ago
  76. 1c1b1ea Allow setting ALR values for screen content again by Erik Språng · 6 years ago
  77. 5982d00 Stop always predicting from last keyframe in the 3TL VP8 case. by philipel · 6 years ago
  78. ade945d Add ability to specify encoder bitrate multiplier in PC level tests by Artem Titov · 6 years ago
  79. fd720b2 Switch to SendTask instead of manually waiting for event. by Rasmus Brandt · 6 years ago
  80. 0682850 Roll chromium_revision 8a7888ec8b..8524e2aded (647072:647205) by chromium-webrtc-autoroll · 6 years ago
  81. 739506e Add thread safety annotations for some more PeerConnection members (part 12) by Karl Wiberg · 6 years ago
  82. c680c4a Revert "Running FrameBuffer on task queue." by Henrik Boström · 6 years ago
  83. fc6f3e58 Include duration of pauses into sum of squared frames duration. by Sergey Silkin · 6 years ago
  84. 78a5e96 Reland "Add thread guards to JsepTransport" by Harald Alvestrand · 6 years ago
  85. 33d2a91 Fix target bitrate RTCP messages behavior for SVC streams by Ilya Nikolaevskiy · 6 years ago
  86. dbfb58b Ignore ERROR_ACCESS_DENIED when stopping Windows platform threads. by Noah Richards · 6 years ago
  87. d9bf720 Add thread safety annotations for some more PeerConnection members (part 11) by Karl Wiberg · 6 years ago
  88. 288cbe8 Remove unused method in VCMInterFrameDelay. by Åsa Persson · 6 years ago
  89. caedb5d Revert "Add thread guards to JsepTransport" by Gustaf Ullberg · 6 years ago
  90. 0105dfc Roll chromium_revision 80b8e2fb67..8a7888ec8b (646925:647072) by chromium-webrtc-autoroll · 6 years ago
  91. 7e1db52 Add thread guards to JsepTransport by Harald Alvestrand · 6 years ago
  92. 12d4e67 Roll chromium_revision d5a4101d31..80b8e2fb67 (646704:646925) by chromium-webrtc-autoroll · 6 years ago
  93. 7d6a259 Adds fake clock unit test. by Sebastian Jansson · 6 years ago
  94. 27d5ad0 Fix mouse not being shared with Handgouts on Win10 by Julien Isorce · 6 years ago
  95. 9435c610 Expose TaskQueueFactory for webrtc::Call in peer connection api by Danil Chapovalov · 6 years ago
  96. 13943b7 Running FrameBuffer on task queue. by Sebastian Jansson · 6 years ago
  97. d98cbd8 Moves send side bandwidth estimation bandwidth cap inside class. by Sebastian Jansson · 6 years ago
  98. 5b84f67 Cleaner reading of field trials in GoogCcNetworkController. by Sebastian Jansson · 6 years ago
  99. 7a651c6e Add thread safety annotations for some more PeerConnection members (part 10) by Karl Wiberg · 6 years ago
  100. 62bb47f Splits network node into link and router. by Sebastian Jansson · 6 years ago