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gerrit.openfyde.cn / webrtc.googlesource.com / src / HEAD
  1. 2b88704 Roll chromium_revision e91245ed6d..b39fd8fad6 (1103355:1103515) by chromium-webrtc-autoroll · 2 years, 5 months ago main master
  2. 59251a9 Roll chromium_revision 73f1a09e0a..e91245ed6d (1103251:1103355) by chromium-webrtc-autoroll · 2 years, 5 months ago
  3. fd877d9 Consolidate TransformableVideoFrame mocks used inside webrtc by Tony Herre · 2 years, 5 months ago
  4. 168d11c Deprecate TransformableVideoFrame GetAdditionalData by Tony Herre · 2 years, 5 months ago
  5. 1f39162 Fill fps allocation for LibaomAv1Encoder correctly by Wan-Teh Chang · 2 years, 5 months ago
  6. 0c4c9be video_encoder.h: update kFullFramerate in comment by Wan-Teh Chang · 2 years, 5 months ago
  7. e1137d7 Delete deprecated variant of IncomingRtcpPacket function by Danil Chapovalov · 2 years, 5 months ago
  8. 16579cc Change MediaChannel to have a Role parameter by Harald Alvestrand · 2 years, 5 months ago
  9. d59c0c3 Roll chromium_revision 790425576f..73f1a09e0a (1103139:1103251) by chromium-webrtc-autoroll · 2 years, 5 months ago lkgr
  10. 24f7b2f Test what happens when asking for simulcast VP9 (not yet supported). by Henrik Boström · 2 years, 5 months ago
  11. d234cef Handling NetEqSetMinimumDelay events in neteq_rtpplay. by Jesús de Vicente Peña · 2 years, 5 months ago
  12. 5c06bef [infra] Remove goma properties by Junji Watanabe · 2 years, 5 months ago
  13. bce135a Add test coverage for legacy VP9 SVC with media flow. by Henrik Boström · 2 years, 5 months ago
  14. 7484e62 Roll chromium_revision b49d6c9ec1..790425576f (1103023:1103139) by chromium-webrtc-autoroll · 2 years, 5 months ago
  15. 713fdc0 Roll chromium_revision 513ae2fffc..b49d6c9ec1 (1102858:1103023) by chromium-webrtc-autoroll · 2 years, 5 months ago
  16. 318cf28 Fix Destruction inside WGC Callback by Alexander Cooper · 2 years, 5 months ago
  17. d8361ee Roll chromium_revision 5346f2f55a..513ae2fffc (1102735:1102858) by chromium-webrtc-autoroll · 2 years, 5 months ago
  18. 12ae52c Roll chromium_revision 2e99d26f79..5346f2f55a (1102579:1102735) by chromium-webrtc-autoroll · 2 years, 5 months ago
  19. 88ddfdb Verify codec and scalability mode in simulcast test. by Henrik Boström · 2 years, 5 months ago
  20. fd4ddd1 Add a simulcast test that verifies media is flowing on all layers. by Henrik Boström · 2 years, 5 months ago
  21. 617d89a Add capture time as identifier in webrtc::VideoFrame by Palak Agarwal · 2 years, 5 months ago
  22. cd3e1d0 Roll chromium_revision e182675fbb..2e99d26f79 (1098562:1102579) by Mirko Bonadei · 2 years, 5 months ago
  23. cd489a0 Fix autoroller not to miss writing to DEPS of variable changes. by Byoungchan Lee · 2 years, 5 months ago
  24. e8ac5af Teach autoroller to roll variables like fuchsia_version. by Byoungchan Lee · 2 years, 5 months ago
  25. b459dea Add ssrc to VideoFrameMetadata used in encoded transforms by Tony Herre · 2 years, 5 months ago
  26. 1f33a2b Add capture timestamps to test audio device. by Olov Brändström · 2 years, 5 months ago
  27. 51dbe82 setOfferedHeaderExtensions: stop any filtered extension by Philipp Hancke · 2 years, 6 months ago
  28. e9c3e51 Add a DEPS hook to download llvm-cov and llvm-profdata based on .gclient custom_vars. by Björn Terelius · 2 years, 5 months ago
  29. c7a0620 Add an ICE switch reason for a switch requested by an application. by Sameer Vijaykar · 2 years, 5 months ago
  30. 95d12ad Create unit test for the population of capture_start_ntp_time by Harald Alvestrand · 2 years, 5 months ago
  31. 4b0d6f9 Upgrade Linux MSan to Focal by Tom Anderson · 2 years, 5 months ago
  32. b0e1cb2 Adds WebRTC.DesktopCapture.Win.DirectXCapturerResult UMA by henrika · 2 years, 5 months ago
  33. fd29662 Fix typo in histogram name by Johannes Kron · 2 years, 5 months ago
  34. 5e7301f Remove rid and rrid from list of extensions that can be used for audio by Philipp Hancke · 2 years, 5 months ago
  35. c6ff4bc Do not transfer ownership of codecs to tester by Sergey Silkin · 2 years, 5 months ago
  36. be9b576 Move video video receiver transformable frame to modules/rtc_rtcp/source by Tony Herre · 2 years, 5 months ago
  37. b311f6a Add UMA histograms to track usage of fullscreen detection by Johannes Kron · 2 years, 5 months ago
  38. 85abbdf RtcEventLogImpl: Add test cases by Xuanxi Leng · 2 years, 6 months ago
  39. d1831cb Treat non DTLS/SCTP Protocol Based Data Channels as Unsupported Media by Dor Hen · 2 years, 6 months ago
  40. f0be3be Add pipewire/portal video capture support by Michael Olbrich · 2 years, 6 months ago
  41. fad9a6d Delete deprecated Create method and config from AudioCodingModule by Henrik Lundin · 2 years, 6 months ago
  42. 101c6aa Remove leftover function signatures. by Fredrik Solenberg · 2 years, 6 months ago
  43. 6c60f72 Refactor video codec testing stats by Sergey Silkin · 2 years, 6 months ago
  44. 97d1c34 Enable rotation tests marked as expected failures by Andreas Pehrson · 2 years, 6 months ago
  45. 65ab5fd Cleanup RemoteEstimatorProxy::IncomingPacket by Danil Chapovalov · 2 years, 6 months ago
  46. ba846cc Add a test that shows when channel_receive fires RR by Harald Alvestrand · 2 years, 6 months ago
  47. 84f7569 Break apart AudioCodingModule and AcmReceiver by Henrik Lundin · 2 years, 6 months ago
  48. c5455e7 Allow RTX ssrc to be updated on receive streams by Per K · 2 years, 6 months ago
  49. be03c09 Only serialize non-stopped RTP header extensions by Philipp Hancke · 2 years, 6 months ago
  50. 1f206b8 Use ArrayView in the IncomingRtcpPacket function. by Harald Alvestrand · 2 years, 6 months ago
  51. 16a8792 Propagate received video csrcs to encodedframe metadata by Tony Herre · 2 years, 6 months ago
  52. 0507fbd [Desktop Capture] Remove disabled test. by mark a. foltz · 2 years, 6 months ago
  53. cb4b0a6 Check FMA3 support before use it in SincResampler by Henrik Lundin · 2 years, 6 months ago
  54. 0b7184c Add possibility to set MetricsSet metadata. by Mirko Bonadei · 2 years, 6 months ago
  55. 217b384 Remove rtp header extension from config of Call audio and video receivers by Per K · 2 years, 6 months ago
  56. 3541732 Add a Config struct to AcmReceiver, and a ctor using it by Henrik Lundin · 2 years, 6 months ago
  57. 3274051 Add method to get FD for physical socket by Artem Titov · 2 years, 6 months ago
  58. d78f8e7 Fix doc path by Artem Titov · 2 years, 6 months ago
  59. e592283 Add 'metadata' field to MetricsSet proto. by Mirko Bonadei · 2 years, 6 months ago
  60. a617867 Reland "Migrate WebRTC documentation to new renderer" by Artem Titov · 2 years, 6 months ago
  61. 2367103 Add android-tiramisuprivacysandbox to DEPS. by Mirko Bonadei · 2 years, 6 months ago
  62. 3963a95 Enforce policy that SDP munging requires special approval by Harald Alvestrand · 2 years, 6 months ago
  63. 4822031 shared_screencast_stream: Set WL capturer id by Salman · 2 years, 6 months ago
  64. 9214718 Roll chromium_revision 2fd12db504..e182675fbb (1098462:1098562) by chromium-webrtc-autoroll · 2 years, 6 months ago
  65. dad91a6 Send periodic TransportFeedback based on extension version by Per K · 2 years, 6 months ago
  66. 2ded55e Cleanup Thread::BlockingCall by Danil Chapovalov · 2 years, 6 months ago
  67. 8155227 sdp: add rtcp-fb:* lines for common feedback by Philipp Hancke · 2 years, 6 months ago
  68. 68564bb [infra] Clean up mb_config.pyl after reclient migration by Junji Watanabe · 2 years, 6 months ago
  69. 942abaa Roll chromium_revision 7faf2e057e..2fd12db504 (1098356:1098462) by chromium-webrtc-autoroll · 2 years, 6 months ago
  70. f6a1f7e Roll chromium_revision ff5a43c15d..7faf2e057e (1098253:1098356) by chromium-webrtc-autoroll · 2 years, 6 months ago
  71. 863a07f Update WebRTC code version (2023-01-29T04:02:18). by webrtc-version-updater · 2 years, 6 months ago
  72. dc7a14b Roll chromium_revision 5195bd9e41..ff5a43c15d (1098136:1098253) by chromium-webrtc-autoroll · 2 years, 6 months ago
  73. 6ff7713 base_capturer_pipewire: Send frames via callback by Salman · 2 years, 6 months ago
  74. bf216a7 Roll chromium_revision d9876846b5..5195bd9e41 (1097987:1098136) by chromium-webrtc-autoroll · 2 years, 6 months ago
  75. 26340b0 desktop_capturer: Support frame rate negotiation via pipewire by Salman · 2 years, 6 months ago
  76. 4c4566a Roll chromium_revision 6da6f5ebed..d9876846b5 (1097837:1097987) by chromium-webrtc-autoroll · 2 years, 6 months ago
  77. 64ce699 Propagate Video CSRCs modified by an insertable streams frame transform by Tony Herre · 2 years, 6 months ago
  78. db20831 Update RTP timestamp based on capture timestamp when audio send stream is resumed. by Jakob Ivarsson · 2 years, 6 months ago
  79. 9d21eb3 Roll chromium_revision 1f33300013..6da6f5ebed (1097480:1097837) by chromium-webrtc-autoroll · 2 years, 6 months ago
  80. a2653bc Export some more symbols for use in chromium tests. by Sameer Vijaykar · 2 years, 6 months ago
  81. ca1cfd4 Add missing `src/third_party/jdk11` dependency and roll chromium into webrtc by landrey · 2 years, 6 months ago
  82. 0f2ce5c Revert "Migrate WebRTC documentation to new renderer" by Artem Titov · 2 years, 6 months ago
  83. b0d8a37 Ensure CallTest derived tests per default set min/max audio bitrate. by Per K · 2 years, 6 months ago
  84. 6c032cb in rtcp::TransportFeedback do not memorise all described packet by Danil Chapovalov · 2 years, 6 months ago
  85. dcb09ff Reset encoder when audio send stream is stopped. by Jakob Ivarsson · 2 years, 6 months ago
  86. 3eceaf4 Migrate WebRTC documentation to new renderer by Artem Titov · 2 years, 6 months ago
  87. 94d5f6a Add missing include by Bjorn Terelius · 2 years, 6 months ago
  88. 66efab2 Measure RTCPMuxPolicy at time of connect by Philipp Hancke · 2 years, 6 months ago
  89. 73e0cc8 Delete unused Audio Bwe integration test. by Per K · 2 years, 6 months ago
  90. cfbb247 Update WebRTC code version (2023-01-26T04:01:54). by webrtc-version-updater · 2 years, 6 months ago
  91. e15b9ff Add a basic unittest for webrtc::voe::ChannelReceive by Harald Alvestrand · 2 years, 6 months ago
  92. 664cf14 Reland "Delete PacketReceiver::DeliverPacket from all implementations" by Per K · 2 years, 6 months ago
  93. 7a67dce prefer absl::optional for rtx-time by Philipp Hancke · 2 years, 6 months ago
  94. 7c43d24 Roll chromium_revision a484be4b74..e5191e93ab (1096680:1096792) by chromium-webrtc-autoroll · 2 years, 6 months ago
  95. 5adc2b6 Correct RTCAudioPlayoutStats type and add kind field. by Fredrik Hernqvist · 2 years, 6 months ago
  96. 0c1c972 Fix gtest-output and resultdb for fuchsia by Christoffer Jansson · 2 years, 6 months ago
  97. 5683a12 Increase expiration timeout for Perf bots by landrey · 2 years, 6 months ago
  98. 5671c64 Stop overriding extensions in rampup tests by Per K · 2 years, 6 months ago
  99. 8773349 Update win10 mixins. by Jeremy Leconte · 2 years, 6 months ago
  100. 92dcc2d Roll chromium_revision 98774e2693..a484be4b74 (1096567:1096680) by chromium-webrtc-autoroll · 2 years, 6 months ago