1. d29c689 Expose adaptive_ptime from Android SDK. by Yura Yaroshevich · 2 hours ago master
  2. d71b38e Update WebRTC code version (2021-04-19T04:03:03). by webrtc-version-updater · 4 hours ago lkgr
  3. d46a174 Expose adaptive_ptime from iOS SDK. by Yura Yaroshevich · 12 hours ago
  4. 7fa8d46 Slight code clarification in RemoveStoppedTransceivers. by Tomas Gunnarsson · 15 hours ago
  5. 0ee5bcf Update WebRTC code version (2021-04-18T04:03:49). by webrtc-version-updater · 28 hours ago
  6. e632402 Remove rtp data channel related code from media_channel.* by Tomas Gunnarsson · 2 days ago
  7. 18ac30c Update WebRTC code version (2021-04-17T04:04:03). by webrtc-version-updater · 2 days ago
  8. 983b620 Remove third_party/xstream from DEPS by Bjorn Terelius · 3 days ago
  9. 78aa5cd dcsctp: Ensure packet size doesn't exceed MTU by Victor Boivie · 3 days ago
  10. 7af57c6 Remove RTP data implementation by Harald Alvestrand · 3 days ago
  11. f981cb3 Add video/g3doc/stats.md to the doc site menu by Artem Titov · 3 days ago
  12. 15e078c Fix unsignalled ssrc race in WebRtcVideoChannel. by Henrik Boström · 3 days ago
  13. 882d007 Add documentation for video/stats. by Åsa Persson · 3 days ago
  14. 0131a4d Delete StreamAdapterInterface by Niels Möller · 3 days ago
  15. b291da8 Add conceptual docs for modules/video_coding by Rasmus Brandt · 3 days ago
  16. dd36198 Revert "Expose AV1 encoder&decoder from Android SDK." by Björn Terelius · 3 days ago
  17. 220a252 Delete unused class MessageBufferReader by Niels Möller · 3 days ago
  18. 6c127a1 Add Stable Writable Connection Ping Interval parameter to RTCConfiguration. by Derek Bailey · 3 days ago
  19. 74b1bbe Remove unused a gn variable related to gtk by Byoungchan Lee · 3 days ago
  20. a43528c Update WebRTC code version (2021-04-16T04:04:52). by webrtc-version-updater · 3 days ago
  21. 3ceb16e [Android] Set use_raw_android_executable explicitly for test() template. by Peter Kotwicz · 4 days ago
  22. 0f57e0b Make libjingle_peerconnection_metrics_default_jni available in Linux builds. by Mirko Bonadei · 4 days ago
  23. 9fea310 Fix crash in WindowCapturerWinGdi::CaptureFrame. by Austin Orion · 4 days ago
  24. a80c3e5 sctp: Reorganize build targets by Florent Castelli · 4 days ago
  25. 6c7c495 doc: fix ice metadata + spelling by Philipp Hancke · 4 days ago
  26. fedd502 Expose AV1 encoder&decoder from Android SDK. by Yura Yaroshevich · 4 days ago
  27. 572f50f Delete left-over references to AsyncInvoker by Niels Möller · 4 days ago
  28. affd219 Delete AsyncInvoker usage from SimulatedPacketTransport by Niels Möller · 4 days ago
  29. bc959b6 Remove enable_rtp_data_channel by Harald Alvestrand · 4 days ago
  30. fa8a946 Remove obsolete DCHECK in remote_audio_source.cc. by Henrik Boström · 4 days ago
  31. 17490b5 Fix regression in UsrSctpReliabilityTest by Niels Möller · 4 days ago
  32. 403e328 Fix build with rtc_libvpx_build_vp9=false by Byoungchan Lee · 4 days ago
  33. 980c460 AGC2: retuning and large refactoring by Alessio Bazzica · 5 days ago
  34. d28434b Configure GN to use python3 to exec_script. by Mirko Bonadei · 5 days ago
  35. dad500a Remove PacketBuffers internal mutex. by philipel · 5 days ago
  36. 61982a7 AGC2 lightweight noise floor estimator by Alessio Bazzica · 5 days ago
  37. 3ab7a55 Reformat pacer doc and add it into sitemap by Artem Titov · 5 days ago
  38. 9aec8c2 Use default rtp parameters to init wrappers in iOS by Yura Yaroshevich · 5 days ago
  39. 89f3dd5 Make RTC_LOG_THREAD_BLOCK_COUNT less spammy for known call counts by Tomas Gunnarsson · 5 days ago
  40. 5744b7f Fix formatting in sitemap.md by Artem Titov · 5 days ago
  41. 08d30a2 Add documentation for video/adaptation by Evan Shrubsole · 5 days ago
  42. 24bc419 Revert "Fix RTP header extension encryption" by Björn Terelius · 5 days ago
  43. dea5721 Adding g3doc for AudioProcessingModule (APM) by Per Åhgren · 5 days ago
  44. 9861f96 dcsctp: Add operators on TimeMs and DurationMs by Victor Boivie · 5 days ago
  45. 8181b4f Add conceptual documentation for NetEq. by Jakob Ivarsson · 5 days ago
  46. a743303 Fix RTP header extension encryption by Lennart Grahl · 5 days ago
  47. 84ba164 Change from sakal@webrtc.org to xalep@webrtc.org in OWNERS files. by Mirko Bonadei · 5 days ago
  48. c54f672 dcsctp: Fix post-review comments for DataTracker by Victor Boivie · 5 days ago
  49. 0498519 Add g3doc for audio coding module. by Minyue Li · 5 days ago
  50. 1fad94f Remove ErleUncertainty by Gustaf Ullberg · 5 days ago
  51. 77d73a6 Document SctpTransport by Harald Alvestrand · 5 days ago
  52. 1d2d169 Update WebRTC code version (2021-04-14T04:04:15). by webrtc-version-updater · 5 days ago
  53. e871e02 Add telemetry to measure usage, perf, and errors in Desktop Capturers. by Austin Orion · 5 days ago
  54. efcfa4b Roll chromium_revision 0bde1c5411..1a13f11499 (871876:872016) by chromium-webrtc-autoroll · 6 days ago
  55. 250fbb3 dcsctp: Make Sequence Number API more consistent by Victor Boivie · 6 days ago
  56. ce423ce Track last packet receive times in RtpVideoStreamReceiver instead of the PacketBuffer. by philipel · 6 days ago
  57. cd83ae2 Speed up FrameCombiner::Combine by 3x by Steve Anton · 6 days ago
  58. 32347b5 Add readme for pacing module by Erik Språng · 6 days ago
  59. 09c7f1e Add architecture section about PeerConnection test framework by Artem Titov · 6 days ago
  60. 79cbe69 Removes incorrect test expectation. by Erik Språng · 6 days ago
  61. 3db3a06 Adding g3doc for AudioDeviceModule (ADM) - part of the AudioEngine by henrika · 6 days ago
  62. df1edc9 API description: PeerConnection description by Harald Alvestrand · 6 days ago
  63. 11b3089 Roll chromium_revision 74f869d04b..0bde1c5411 (871745:871876) by chromium-webrtc-autoroll · 6 days ago
  64. 1fded2f dcsctp: Fix build dependencies by Florent Castelli · 6 days ago
  65. e082984 Add death test for WrappingAsyncResolver by Harald Alvestrand · 6 days ago
  66. a168bb9 Add index.md documentation page for PC level test framework by Artem Titov · 6 days ago
  67. 696cea0 Refactor some RtpSender-level tests into RtpRtcp-level tests by Erik Språng · 6 days ago
  68. 5fe0b37 Roll chromium_revision 7e70585ca5..74f869d04b (871605:871745) by chromium-webrtc-autoroll · 6 days ago
  69. c8cf0a6 Remove MDNS message implementation by Harald Alvestrand · 6 days ago
  70. eff79cf Roll chromium_revision 2dffe06711..7e70585ca5 (871492:871605) by chromium-webrtc-autoroll · 7 days ago
  71. 067dce7 Fix processing of dropped frame for runtime added participant by Andrey Logvin · 7 days ago
  72. dc53ce6 Revert "Add addr in error msg if stun sock sent with error" by Mirko Bonadei · 7 days ago
  73. 8093935 Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492) by Mirko Bonadei · 7 days ago
  74. 5051693 [Battery]: TaskQueuePacedSender not started by default. by Etienne Pierre-doray · 7 days ago
  75. 9ff75a6 Add addr in error msg if stun sock sent with error by Yura Yaroshevich · 7 days ago
  76. 3928e8f dcsctp: Disable packet fuzzers by Victor Boivie · 7 days ago
  77. 0aa1a19 Add module overview of ICE by Jonas Oreland · 7 days ago
  78. 9de39f6 Add titovartem@webrtc.org as owner for /g3doc by Artem Titov · 7 days ago
  79. 4af6f2b Move threading documentation for API into g3doc structure by Harald Alvestrand · 7 days ago
  80. a3575cb Remove tautological 'unsigned expr < 0' comparisons by Anton Bikineev · 7 days ago
  81. 22379fc sctp: Rename SctpTransport to UsrSctpTransport by Florent Castelli · 7 days ago
  82. 606bd6d dcsctp: Use correct field width for PPID by Victor Boivie · 7 days ago
  83. 9d60936 dcsctp: Fix relative dependency paths in timer/ by Victor Boivie · 7 days ago
  84. 1003219 srtp: compare key length to srtp policy key length by Philipp Hancke · 7 days ago
  85. 5691053 IceStatesReachCompletionWithRemoteHostname: disable on Linux. by Markus Handell · 7 days ago
  86. 9071957 Remove unused members in tests. by Åsa Persson · 7 days ago
  87. 55de292 Use relative paths for //net/dcsctp/public:socket. by Mirko Bonadei · 7 days ago
  88. 1cdeb0a addIceCandidate with callback into Android's SDK. by Yura Yaroshevich · 7 days ago
  89. f075917 Ensure TaskQueuePacedSender dont depend on PacketRouter by Per Kjellander · 7 days ago
  90. 1c73e03 Update WebRTC code version (2021-04-12T04:03:54). by webrtc-version-updater · 7 days ago
  91. cb70aa7 dcsctp: Add Reassembly Queue by Victor Boivie · 8 days ago
  92. 8a13d2c dcsctp: Add Traditional Reassembly Streams by Victor Boivie · 8 days ago
  93. b2d539b dcsctp: Add Data Tracker by Victor Boivie · 8 days ago
  94. 50fc1df dcsctp: Add SCTP packet corpus by Victor Boivie · 8 days ago
  95. a4da76a Update WebRTC code version (2021-04-10T04:03:36). by webrtc-version-updater · 9 days ago
  96. 061d898 Update WgcScreenSource* to use device indices instead of HMONITORs. by Austin Orion · 9 days ago
  97. f2f9bb6 Fixing a buffer copy issue in DesktopFrame by Joe Downing · 10 days ago
  98. fc5d276 Fix dropped frames not counted issue by Johannes Kron · 10 days ago
  99. 9410217 dcsctp: Add SCTP packet fuzzer by Victor Boivie · 10 days ago
  100. 3c31ee0 Reduce logging for PC supported codecs in PC level tests by Artem Titov · 10 days ago