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  1. eba6831 Alias instead of reimplement rtc::RefCountedObject by Danil Chapovalov · 12 hours ago lkgr main master
  2. 639494b Add method CompactNtpIntervalToTimeDelta by Per K · 12 hours ago
  3. c2180102 Rename rtc::Packet to rtc::VirtualSocketPacket by Florent Castelli · 12 hours ago
  4. 3073c48 Move rtc_base/network/ecn_marking.h to api/transport by Per K · 12 hours ago
  5. a2205e3 Propagate the corruption_score metric to `RTCInboundRtpStreamStats`. by Emil Vardar · 14 hours ago
  6. 3f1a04a Comment unused variables in implemented functions 5\n by Dor Hen · 17 hours ago
  7. ca07d54 Comment unused variables in implemented functions 4\n by Dor Hen · 17 hours ago
  8. 6d58a43 Comment unused variables in implemented functions 3\n by Dor Hen · 17 hours ago
  9. 7a3c07b Cleanup: Move all comparator tests to codec_comparators_unittests by Harald Alvestrand · 15 hours ago
  10. d42640c Review style guide for freshness by Danil Chapovalov · 15 hours ago
  11. 29a4ada Update WebRTC code version (2024-10-22T04:06:00). by webrtc-version-updater · 22 hours ago
  12. 079a8b4 Refactor CongestionControllerFeedback logic by Per K · 2 days ago
  13. 10e4d86 Add helper to inject custom implementation of audio processing as factory by Danil Chapovalov · 2 days ago
  14. 929c02a Add IsSameRtpCodec method to Codec. by Åsa Persson · 5 days ago
  15. 6d815bd Let the existing TransportFeedback work with RFC8888 congesting control by Sun Shin · 5 days ago
  16. 78456fa Update WebRTC code version (2024-10-21T04:04:50). by webrtc-version-updater · 2 days ago
  17. 14dc9fb Update WebRTC code version (2024-10-20T04:05:53). by webrtc-version-updater · 3 days ago
  18. 915d555 Update WebRTC code version (2024-10-19T04:04:53). by webrtc-version-updater · 4 days ago
  19. 0bac2aa Use Payload Type suggester for all codec merging by Harald Alvestrand · 6 days ago
  20. cdc38b9 Remove unused field trial. by Emil Vardar · 7 days ago
  21. c382c84 fix h264 encoder don't generate template_structure after first keyframe by 林恩 · 6 days ago
  22. 93e1778 Prepare TransportFeedbackAdapter for RFC8888 by Per K · 5 days ago
  23. 03b2c9f Let ZeroOnFreeBuffer do the memcpy for DTLS-SRTP key extraction by Philipp Hancke · 9 days ago
  24. cecee51 Preserve the requested order for RTC event log plots by Björn Terelius · 5 days ago
  25. 337f6f2 remove proto_data_sources usages by Takuto Ikuta · 5 days ago
  26. ca932cb Update WebRTC code version (2024-10-18T04:06:16). by webrtc-version-updater · 5 days ago
  27. a5c5ff4 Disable WindowFinderTest.FindConsoleWindow due to flakiness by Bjorn Terelius · 5 days ago
  28. e5c3912 Remove unneccessary base64 includes and deps from pc/ by Philipp Hancke · 6 days ago
  29. 51fccaf Add dependency descriptor support for H264 when no template information by Brennan Waters · 6 days ago
  30. 27d3d74 Check return values in WindowFinderTest.FindConsoleWindow on win by Björn Terelius · 6 days ago
  31. ecb3ed7 Migrate CreateVoipEngine to take audio_processing_factory instead of audio_processing by Danil Chapovalov · 6 days ago
  32. b280cb95 Add a basic end-to-end test for corruption detection. by Fanny Linderborg · 6 days ago
  33. 44e17f3 Add value_type alias to EncodedImageBufferInterface by Emil Vardar · 6 days ago
  34. 3cc5835 Update WebRTC code version (2024-10-17T04:07:50). by webrtc-version-updater · 6 days ago
  35. f29fb25 Add begin()/end() to CopyOnWriteBuffer. by Peter Kasting · 7 days ago
  36. e8d27c7 PCLF: provide port allocator flags directly instead of providing only extra flags by Artem Titov · 7 days ago
  37. 049b43b [reland] Comment unused variables in implemented functions by Dor Hen · 8 days ago
  38. 558c2dc Change timestamps type from int64 to Timestamp in MediaReceiverInfo. by Olov Brändström · 8 days ago
  39. 9c21f63 Replace AudioProcessingBuilderForTesting with the BuiltinAudioProcessingFactory by Danil Chapovalov · 7 days ago
  40. db1c618 Remove legacy transitive include for RtcEventLogOutputFile by Philipp Hancke · 2 weeks ago
  41. d356228 Do not change crypto options in peer_connection.cc by Emil Vardar · 12 days ago
  42. 183a522 Enable corruption detection when the encrypted extension is present by Emil Vardar · 8 days ago
  43. e486aed Fix misaligned read in physical_socket_server by Philipp Hancke · 11 days ago
  44. 2dc95ba Add BuiltinAudioProcessingFactory by Danil Chapovalov · 8 days ago
  45. 6d433af Roll chromium_revision 03821003e7..030af8fbf0 (1368333:1368717) by chromium-webrtc-autoroll · 8 days ago
  46. 2a569a2 For test peer start/stop AEC dump using peer connection factory api by Danil Chapovalov · 8 days ago
  47. 2ee43d6 Callback to NetworkStateEstimateObserver before NetworkLinkRtcpObserver by Per K · 8 days ago
  48. 129f228 Post corruption score aggregation to worker thread. by Emil Vardar · 8 days ago
  49. 6f99dba Roll chromium_revision e82ec81396..03821003e7 (1368150:1368333) by chromium-webrtc-autoroll · 8 days ago
  50. 3fc43e2 Add missing field-trial in Vp9EncoderReferencesFuzzer by Ilya Nikolaevskiy · 9 days ago
  51. 41d9a01 Prepare function for deprecation by Fanny Linderborg · 9 days ago
  52. 01a9264 Remove the iLBC audio codec by Alessio Bazzica · 14 days ago
  53. ad49112 Introduce AudioProcessingFactory interface by Danil Chapovalov · 9 days ago
  54. 1deb4f8 Roll chromium_revision 822a2bcac0..e82ec81396 (1367727:1368150) by chromium-webrtc-autoroll · 9 days ago
  55. ce88f7f add DTLSSrtpTransport/SrtpTransport integration test by Philipp Hancke · 11 days ago
  56. d8bddfe Split up the call/video_stream_api target by Harald Alvestrand · 9 days ago
  57. 36e5979 Work around stdc++ bug with clang by Patrick Reynolds · 11 days ago
  58. e19dd3c Add MB config for use_custom_libcxx=false by Mirko Bonadei · 9 days ago
  59. 0aae211 Update WebRTC code version (2024-10-12T04:05:35). by webrtc-version-updater · 11 days ago
  60. d299c6a Roll chromium_revision ffc62772df..822a2bcac0 (1367619:1367727) by chromium-webrtc-autoroll · 11 days ago
  61. 6caca65 Reland "Spanify SRTP key export" by Philipp Hancke · 12 days ago
  62. a82975f Roll chromium_revision fd540ab771..ffc62772df (1366971:1367619) by chromium-webrtc-autoroll · 11 days ago
  63. bd42ee8 Refactor FindMatchingCodec by Harald Alvestrand · 12 days ago
  64. 346cf7c Add support for frame pair corruption score calculation. by Emil Vardar · 3 weeks ago
  65. 7b1b7a0 Update WebRTC code version (2024-10-11T04:05:16). by webrtc-version-updater · 12 days ago
  66. 60c3fea Fix header length and set layer_id/temporal_id with lowest value of aggregated NALU for AP packet in H265 RTP packetizer by qwu16 · 4 weeks ago
  67. f1849d9 Roll chromium_revision 1f8c3616d8..fd540ab771 (1366802:1366971) by chromium-webrtc-autoroll · 12 days ago
  68. 11d3f0d Roll chromium_revision aa68dfe997..1f8c3616d8 (1366639:1366802) by chromium-webrtc-autoroll · 13 days ago
  69. ae40039 Add comparators unittest, and abandon MatchesForSdp by Harald Alvestrand · 13 days ago
  70. 3203b62 Add AbslStringify for cricket::Codec by Harald Alvestrand · 13 days ago
  71. 366f205 Change rtc_executable template to depend on absl_full when built with chromium by Danil Chapovalov · 13 days ago
  72. b74268e Update TODOs to the correct format. by Olov Brändström · 13 days ago
  73. 518bd61 Forward the corruption score from the decoder to ReceiveStatisticsProxy by Fanny Linderborg · 13 days ago
  74. 2d75cd3 Roll chromium_revision ccf648df91..aa68dfe997 (1365600:1366639) by Jeremy Leconte · 13 days ago
  75. d180aaa Update WebRTC code version (2024-10-10T04:05:53). by webrtc-version-updater · 13 days ago
  76. 19bbd6f Move some codec-comparing functions to a single file. by Harald Alvestrand · 14 days ago
  77. fea3280 update format of recently added TODOs. by Olov Brändström · 14 days ago
  78. f95278f Revert "Allow sending to separate payload types for each simulcast index." by Jeremy Leconte · 13 days ago
  79. 042359f Revert "Roll chromium_revision ccf648df91..dcb20ee7c2 (1365600:1365760)" by Jeremy Leconte · 13 days ago
  80. 51b6826 Add an environment clock timestamp to SenderReportStats. by Olov Brändström · 14 days ago
  81. eb0ba6b Add sprang as api/video OWNER by Rasmus Brandt · 14 days ago
  82. 6099b64 Improve error message for tests comparing RTP header extensions. by Emil Vardar · 2 weeks ago
  83. 81d5ab8 Add field trial to enable negotiation of encrypted RTP header extensions by Emil Vardar · 3 weeks ago
  84. 32590ef Revert "Spanify SRTP key export" by Jeremy Leconte · 14 days ago
  85. 8fad80b Update WebRTC code version (2024-10-09T04:06:05). by webrtc-version-updater · 14 days ago
  86. 1c6d509 Roll chromium_revision ccf648df91..dcb20ee7c2 (1365600:1365760) by chromium-webrtc-autoroll · 2 weeks ago
  87. f055f9b Advise to use [[deprecated]], not ABSL_DEPRECATED by Harald Alvestrand · 2 weeks ago
  88. 65ae324 Spanify SRTP key export by Philipp Hancke · 3 weeks ago
  89. bcb19c0 Allow sending to separate payload types for each simulcast index. by Shigemasa Watanabe · 2 weeks ago
  90. c7f9426 Roll chromium_revision 8f3f021772..ccf648df91 (1363170:1365600) by chromium-webrtc-autoroll · 2 weeks ago
  91. e1adfc0 Rename FrameToRender to OnFrameToRender by Fanny Linderborg · 2 weeks ago
  92. b9c4c24 rename timestamps to show epoch by Olov Brändström · 2 weeks ago
  93. e3819f6 Fix java errors that used to be disabled. by Jeremy Leconte · 2 weeks ago
  94. a507a08 Calculate corruption score once the frame is decoded by Fanny Linderborg · 2 weeks ago
  95. fb43116 Improve SDP negotiation for mixed encrypted/unencrypted offers. by Emil Vardar · 2 weeks ago
  96. fec6f8d Cleanup duplicated log streaming operators by Danil Chapovalov · 2 weeks ago
  97. ebd3732 Remove support for logging types via ToLogString extension by Danil Chapovalov · 2 weeks ago
  98. 215401f Reland "Add a FrameToRender argument struct as input to FrameToRender" by Fanny Linderborg · 2 weeks ago
  99. 3d2e730 Update WebRTC code version (2024-10-08T04:06:02). by webrtc-version-updater · 2 weeks ago
  100. b507daf Refactor NetEq delay constraint logic. by Jakob Ivarsson · 2 weeks ago