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  1. 4ee5e5f Disable VideoCaptureTest due to flakyness by Björn Terelius · 84 minutes ago main master
  2. 37fb647 Disable the roll of 'android_ndk' by Prashanth Swaminathan · 5 hours ago lkgr
  3. 36c945b Update WebRTC code version (2023-06-08T04:11:54). by webrtc-version-updater · 7 hours ago
  4. 9d9c3f4 [Analysis] Remove old threshold fields by Beining Chen · 12 hours ago
  5. 89f64b9 Make packet info optional and only set for primary packets in NetEq. by Jakob Ivarsson · 18 hours ago
  6. 9e639fa Migrate Android NDK to CIPD [1/2] by Prashanth Swaminathan · 19 hours ago
  7. fc260a18 Add method SetTimestamp in TransformableAudioFrameInterface by Palak Agarwal · 21 hours ago
  8. 3403acb av1: 8 threads for >720p and tiles config by Jerome Jiang · 21 hours ago
  9. d615704 Enable frame dropping in libaom AV1 encoder by Sergey Silkin · 23 hours ago
  10. a458fe5 Update WebRTC code version (2023-06-07T04:12:21). by webrtc-version-updater · 31 hours ago
  11. 09e0086 Remove ImplForTesting function from MediaChannel by Harald Alvestrand · 2 days ago
  12. bd66cfe Roll chromium_revision a5cd053713..a8db252505 (1153688:1153825) by chromium-webrtc-autoroll · 2 days ago
  13. 847208e Remove transitional shim classes by Harald Alvestrand · 2 days ago
  14. ade07ca Rename current flexfec implementation flexfec_03 by Yosef Twaik · 2 days ago
  15. 43df03d Fix spelling mistake ReplaceRemoteDescriptionAndCheckE*r*or by Philipp Hancke · 2 days ago
  16. 6d25e96 Roll chromium_revision 404afa6a86..a5cd053713 (1153573:1153688) by chromium-webrtc-autoroll · 2 days ago
  17. d3b71c7 Update WebRTC code version (2023-06-06T04:12:09). by webrtc-version-updater · 2 days ago
  18. e00a12f Roll chromium_revision 96ad22527d..404afa6a86 (1153423:1153573) by chromium-webrtc-autoroll · 2 days ago
  19. 8c4b9ea Remove references to AudioCodec and VideoCodec constructors by Florent Castelli · 3 days ago
  20. fd096da Roll chromium_revision 8f3397a259..96ad22527d (1153256:1153423) by chromium-webrtc-autoroll · 3 days ago
  21. 77c6230 Add create functions for voice media send and receive channels. by Harald Alvestrand · 3 days ago
  22. be316da Ensure that RTCErrorOr<T, E> doesn't require T to be default constructible by Florent Castelli · 3 days ago
  23. 0740048 Roll chromium_revision f28b824184..8f3397a259 (1152496:1153256) by chromium-webrtc-autoroll · 3 days ago
  24. b0ef5e4 Declare factory functions for video sender and receiver by Harald Alvestrand · 3 days ago
  25. 2f0c078 Split WebRtcVoiceChannel into Send and Receive classes by Harald Alvestrand · 3 days ago
  26. 1e04d61 Update WebRTC code version (2023-06-05T04:02:35). by webrtc-version-updater · 3 days ago
  27. 816f5b1 Create VP9Encoder with a VP9 codec object by Florent Castelli · 4 days ago
  28. 968e3c0 rtp_sender: fix typo with spatial_bitmask by Alfred E. Heggestad · 4 days ago
  29. 079ce25 Update WebRTC code version (2023-06-04T04:02:33). by webrtc-version-updater · 4 days ago
  30. e10f025 Update WebRTC code version (2023-06-03T04:02:02). by webrtc-version-updater · 5 days ago
  31. 5278b39 Add H264Encoder::Create() by Florent Castelli · 6 days ago
  32. 811e24a Move functionality from AudioCodec and VideoCodec into cricket::Codec by Florent Castelli · 6 days ago
  33. b8651de Roll chromium_revision d48b2929db..f28b824184 (1152392:1152496) by chromium-webrtc-autoroll · 6 days ago
  34. 54e95bc Propagate time of the last received packet with Timestamp type by Danil Chapovalov · 6 days ago
  35. 9a34d80 Apply the "shim" pattern for WebRtcVoiceEngine by Harald Alvestrand · 6 days ago
  36. b15a9f0 Fix perf tests. by Jeremy Leconte · 6 days ago
  37. 3488726 sdp: reject spec simulcast answers without the rid extension by Philipp Hancke · 6 days ago
  38. f785bd4 Split WebRtcVideoMediaChannel into Send and Receive by Harald Alvestrand · 6 days ago
  39. 4ad141e Add callback for send codec in audio too by Harald Alvestrand · 6 days ago
  40. 371b7af Roll chromium_revision 2478b63fb4..d48b2929db (1151892:1152392) by chromium-webrtc-autoroll · 6 days ago
  41. b29ee5b Run the same perf tests on all platforms. by Jeremy Leconte · 6 days ago
  42. 267040e Make native VideoTrack pointer public by Jonas Oreland · 6 days ago
  43. cfc1a3a Update vpython3 requests by Brian Sheedy · 6 days ago
  44. eeacddb Disable flaky PictureIdTests. by Jeremy Leconte · 6 days ago
  45. d454815 Use //third_party/cpu_features directly by Prashanth Swaminathan · 6 days ago
  46. dab505b Update WebRTC code version (2023-06-02T04:02:59). by webrtc-version-updater · 6 days ago
  47. 063b45b Roll chromium_revision faf350b988..2478b63fb4 (1151758:1151892) by chromium-webrtc-autoroll · 7 days ago
  48. dba22d3 Move transceiver iteration loop over to the signaling thread. by Tommi · 7 days ago
  49. 513ab0c Add a -d option to apply-iwyu by Harald Alvestrand · 7 days ago
  50. e24b34c Roll chromium_revision e26eb46a54..faf350b988 (1150524:1151758) by chromium-webrtc-autoroll · 7 days ago
  51. b93f69a In VideoCaptureV4L2 create the capture thread last in StartCapture by Andreas Pehrson · 7 days ago
  52. e44a155 Add third_party/cpu_features license path. by Jeremy Leconte · 7 days ago
  53. 2d59853 Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController. by Ying Wang · 7 days ago
  54. 3d6e88e Remove low_bandwidth_audio_test. by Jeremy Leconte · 7 days ago
  55. 6110fd9 Update WebRTC code version (2023-06-01T04:12:34). by webrtc-version-updater · 7 days ago
  56. cb85143 Fix duplicate 'unix' OS and latest-revision deps by Prashanth Swaminathan · 8 days ago
  57. 2197300 Update ReceiveStatistics to use Timestamp/TimeDelta to represent time by Danil Chapovalov · 8 days ago
  58. a9bba04 Updating AsyncAudioProcessing API, part 1. by Peter Hanspers · 8 days ago
  59. 56d69e2 Add //third_party/cpu_features to DEPS by Prashanth Swaminathan · 8 days ago
  60. c18f083 Split MediaChannel concrete functions to MediaChannelUtil by Harald Alvestrand · 8 days ago
  61. 94a9d55 Update WebRTC code version (2023-05-31T04:11:01). by webrtc-version-updater · 8 days ago
  62. b84fae6 Use sinf instead of std::sinf to improve libstdc++ compatibility by Li-Yu Yu · 9 days ago
  63. 9fa5057 Roll chromium_revision da88253915..e26eb46a54 (1150417:1150524) by chromium-webrtc-autoroll · 9 days ago
  64. 6acfbb0 Replace std::optional with absl::optional in RtpPacketHistory by Per K · 9 days ago
  65. d8098fb Delete struct RTCPReportBlock as no longer used by Danil Chapovalov · 9 days ago
  66. d8b88d8 Use the VideoMediaChannelShim for all cases by Harald Alvestrand · 9 days ago
  67. 428836d tools: fix small typo in python script by Alfred E. Heggestad · 9 days ago
  68. 4bf5238 sdp: reject BUNDLE with RTP header extension id collisions by Philipp Hancke · 9 days ago
  69. b184634 Run webrtc_perf_tests on Fuchsia os. by Jeremy Leconte · 9 days ago
  70. c73ea4f More systematic null checks before calling native methods by Xavier Lepaul · 9 days ago
  71. a3e9c0a Roll chromium_revision c90a8a46d7..da88253915 (1150306:1150417) by chromium-webrtc-autoroll · 9 days ago
  72. 97c9623 Make a shim object implementing the VideoMediaChannel interface by Harald Alvestrand · 9 days ago
  73. 4c1e959 Change flexfec header reader to parse according to updated RFC. by Yosef Twaik · 9 days ago
  74. e4a9a6d Update WebRTC code version (2023-05-30T04:02:06). by webrtc-version-updater · 9 days ago
  75. c5e4bcc Roll chromium_revision 599c746c73..c90a8a46d7 (1150194:1150306) by chromium-webrtc-autoroll · 10 days ago
  76. 4b14cb7 Roll chromium_revision fa2e063162..599c746c73 (1150086:1150194) by chromium-webrtc-autoroll · 10 days ago
  77. 4aaacb4 Update WebRTC code version (2023-05-29T04:03:50). by webrtc-version-updater · 10 days ago
  78. e641a97 In RtcpReceiver remove redundand way to represent RTCP report blocks by Danil Chapovalov · 11 days ago
  79. b9de471 Update WebRTC code version (2023-05-28T04:11:22). by webrtc-version-updater · 11 days ago
  80. 98185b9 Roll chromium_revision 99b12997bf..fa2e063162 (1150050:1150086) by chromium-webrtc-autoroll · 12 days ago
  81. a294353 Use type raw for video_codec_perf_tests. by Mirko Bonadei · 12 days ago
  82. 01c2efc Roll chromium_revision bddf6cbe18..99b12997bf (1149812:1150050) by chromium-webrtc-autoroll · 12 days ago
  83. 9bc8d05 Update WebRTC code version (2023-05-27T04:12:09). by webrtc-version-updater · 12 days ago
  84. 9ac543c Roll chromium_revision 1fc947a5da..bddf6cbe18 (1149703:1149812) by chromium-webrtc-autoroll · 13 days ago
  85. 87e74f9 Remove unused combined_audio_video_bwe. by Yury Yarashevich · 13 days ago
  86. 2bb686d Stop running low_bandwith_audio_tests. by Jeremy Leconte · 13 days ago
  87. 6490999 Roll chromium_revision aae661725b..1fc947a5da (1148994:1149703) by chromium-webrtc-autoroll · 13 days ago
  88. f0820ff Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay by Rasmus Brandt · 13 days ago
  89. 9caef2a Use a constant for invalid PipeWire file descriptor by Jan Grulich · 13 days ago
  90. 0f1a2c5 Change StreamDataCounters to use Timestamp instead of int64_t by Danil Chapovalov · 13 days ago
  91. 5f32fa4 Delete MediaBaseChannel class by Harald Alvestrand · 13 days ago
  92. 4f1dcbb doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c by Li-Yu Yu · 13 days ago
  93. f53b343 Cleanup RtcpTransceiver dependency on webrtc::Transport by Danil Chapovalov · 13 days ago
  94. 5f38949 Allow single-mline offers without BUNDLE group when using max-bundle by Philipp Hancke · 13 days ago
  95. dff6e25 Update WebRTC code version (2023-05-26T04:05:22). by webrtc-version-updater · 13 days ago
  96. 6e23fa5 Cleanup WebRTC-PayloadTypes-Lower-Dynamic-Range trial by Philipp Hancke · 14 days ago
  97. 56d1260 PipeWire video capture: split portal and PipeWire implementations by Jan Grulich · 14 days ago
  98. 2264e7a Fixes distortion in WGC screen capture path by henrika · 14 days ago
  99. 40a0fa9 Add new padding mode to RtpPacketHistory by Per K · 14 days ago
  100. 4206d31 [Analysis] Add new thresholds config schema by Beining Chen · 14 days ago