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  1. d44ecfe Update WebRTC code version (2024-09-22T04:04:45). by webrtc-version-updater · 2 hours ago main master
  2. 76821c8 Update WebRTC code version (2024-09-21T04:04:38). by webrtc-version-updater · 26 hours ago lkgr
  3. bdc6693 Revert "h264: fix first_packet_in_frame logic for multislice in a single rtp packet" by Gao Chun · 2 days ago
  4. b6ee51b Don't restrict max simulcast layers when `requested_resolution` is used. by Henrik Boström · 2 days ago
  5. 2f106d6 Add FrameInstrumentationGenerator to VideoStreamEncoder by Fanny Linderborg · 2 days ago
  6. 64e8e64 Revert "Reland "Return audio stats regarless if we have a codec."" by Tomas Lundqvist · 3 days ago
  7. f566dee Make requested_resolution throw on invalid dimensions. by Henrik Boström · 3 days ago
  8. 3aa47cf PipeWire camera: get max FPS for each format when specified as list by Jan Grulich · 3 days ago
  9. 1b8a7b2 Update WebRTC code version (2024-09-20T04:04:47). by webrtc-version-updater · 2 days ago
  10. 9a65339 srtp: spanify key setters by Philipp Hancke · 8 days ago
  11. 36f153e Apply include-cleaner to api direct files (2/2). by Jeremy Leconte · 10 days ago
  12. c1dc8ab Remove non-span NAL unit splitter and SPS parser by Philipp Hancke · 5 days ago
  13. 4595711 Revert "Disable TLS session ticket for DTLS" by Mirko Bonadei · 3 days ago
  14. e2952a0 Eliminate a pointless IsEnabled helper by Harald Alvestrand · 3 days ago
  15. 2548d22 WebRTC-TaskQueue-ReplaceLibeventWithStdlib: Launch stdlib task queue. by Markus Handell · 4 days ago
  16. d2123d9 Associate payload_type with rid by Shigemasa Watanabe · 2 weeks ago
  17. bba1a2e Propagate Environment to RtpPacketHistory by Joachim Reiersen · 4 days ago
  18. 5cf1285 Update WebRTC code version (2024-09-19T04:07:15). by webrtc-version-updater · 3 days ago
  19. e77d751 Disable TLS session ticket for DTLS by Philipp Hancke · 5 days ago
  20. 54903b4 Delete transient suppression code by Hanna Silen · 4 days ago
  21. 0fe3a61 Remove clang 3.7 and fuchsia specific flags by Philipp Hancke · 4 days ago
  22. 4a201de Add support for corruption classification. by Emil Vardar · 4 days ago
  23. f045dbd Modify sequence index on key frames by Fanny Linderborg · 4 days ago
  24. 1c4c165 Update WebRTC code version (2024-09-18T04:05:33). by webrtc-version-updater · 4 days ago
  25. b08a045 fix missing deps for proto compile actions by Takuto Ikuta · 5 days ago
  26. bbea923 Removed unused absl::InlinedVector. by Taylor Brandstetter · 5 days ago
  27. 17ffd36 Remove IntKeyTypeFamilyToKeyType by Philipp Hancke · 9 days ago
  28. 825e4f1 VideoAdapter: Interpret requested resolution as max restriction. by Henrik Boström · 5 days ago
  29. 52ea2c3 Propagate FieldTrialsView to query WebRTC-StableTargetRate field trial by Danil Chapovalov · 5 days ago
  30. e81ba30 Increase AV1 QP threshold for quality convergence from 40 to 60. by Sergey Silkin · 5 days ago
  31. 098c128 Explicitly use the Opus DTX encoder state. by Lionel Koenig · 6 days ago
  32. d153de6 Add payload type assignment to offer/answer generation. by Harald Alvestrand · 5 days ago
  33. a1ed306 Cleanup unused members in RtpRtcp::Configuration by Danil Chapovalov · 13 days ago
  34. 2957588 Export scalability mode helper APIs. by Qiu Jianlin · 9 days ago
  35. 4b51217 Make purple bots happy: Shorten TEST_P names. by Henrik Boström · 5 days ago
  36. 59d592e Replace list usage with set for files accumulation in PRESUBMIT to by Dor Hen · 6 days ago
  37. f3a33c0 Prepend all RTCMacros.h includes/imports with the relative path from repo root by Dor Hen · 6 days ago
  38. de6225b Don't crash on failed EGL makeCurrent attempts by Raman Budny · 11 days ago
  39. ce69c73 Clobber caches on Windows by Mirko Bonadei · 5 days ago
  40. 18486c5 Make GetSourcesVideo test wait for two frames by Harald Alvestrand · 6 days ago
  41. cbf5122 Avoid signaling requested_resolution back to the adapting source. by Henrik Boström · 6 days ago
  42. 8487d32 Remove all use of AcmReceiver from WebRTC by Henrik Lundin · 9 days ago
  43. 6e312e5 install libsrtp log handler by Philipp Hancke · 4 weeks ago
  44. 1320982 Remove SrtpTransport MaybeSetKeyParams and ParseKeyParams by Philipp Hancke · 10 days ago
  45. 2b5f7cb Adjust `requested_resolution` to match frame's aspect ratio. by Henrik Boström · 9 days ago
  46. 13e377b Update WebRTC code version (2024-09-13T04:07:27). by webrtc-version-updater · 9 days ago
  47. 08ec444 Roll chromium_revision 3b552b31ee..3b70d6f26c (1354345:1354985) by chromium-webrtc-autoroll · 9 days ago
  48. ec38238 Ensure the AudioCodingModule is reset when sending is stopped. by Lionel Koenig · 10 days ago
  49. 6aab4cc Change cricket::Codec default id from 0 to -1 by Harald Alvestrand · 10 days ago
  50. dfd8f57 Adds a WebRTC.DesktopCapture.Win.WgcDirtyRegionSupport UMA for diagnostic purposes. by henrika · 10 days ago
  51. 97c594f Add field trial for late PT allocation by Harald Alvestrand · 10 days ago
  52. 1859109 Specify in which RTP packet corruption score will be sent on. by Emil Vardar · 10 days ago
  53. fb0da3a Increase test coverage of InitialFrameDropper vs. ScaleResolutionDownBy by Jonas Oreland · 10 days ago
  54. 0d31d7b Increase test coverage of InitialFrameDropper vs. RequestedResolution by Jonas Oreland · 10 days ago
  55. ad4d3e9 Update WebRTC code version (2024-09-12T04:08:20). by webrtc-version-updater · 10 days ago
  56. ca368dd Roll chromium_revision 5c2bd4f9ef..3b552b31ee (1353980:1354345) by chromium-webrtc-autoroll · 10 days ago
  57. 9e0f2fe Roll chromium_revision 6c19d4f358..5c2bd4f9ef (1353847:1353980) by chromium-webrtc-autoroll · 11 days ago
  58. 97d0427 Add converters for corruption detection structs by Fanny Linderborg · 11 days ago
  59. e25b15e Update ownership of PCLF documentation. by Jeremy Leconte · 11 days ago
  60. 51a2bd1 Allow sdk/objc owners to approve sdk/BUILD.gn by Danil Chapovalov · 3 weeks ago
  61. e88a961 Roll chromium_revision 91acefc7c4..6c19d4f358 (1353678:1353847) by chromium-webrtc-autoroll · 11 days ago
  62. 2fb369a Refresh g3doc/implementation_basics.md by Harald Alvestrand · 12 days ago
  63. 254bd32 Update when/how `requested_resolution` throws for invalid parameters. by Henrik Boström · 11 days ago
  64. 1bd331f Ensure <netinet/in.h> is included by using rtc_base/ip_address.h. by Jeremy Leconte · 11 days ago
  65. 47d48a2 Update WebRTC code version (2024-09-11T04:05:44). by webrtc-version-updater · 11 days ago
  66. 6e8cff4 Roll chromium_revision 817ee7871b..91acefc7c4 (1353554:1353678) by chromium-webrtc-autoroll · 11 days ago
  67. 3accb4c Roll chromium_revision 56088b275c..817ee7871b (1353390:1353554) by chromium-webrtc-autoroll · 11 days ago
  68. e184c56 Roll chromium_revision 5dc6c1eec4..56088b275c (1353232:1353390) by chromium-webrtc-autoroll · 12 days ago
  69. 83d1f9a Ensure <sys/socket.h> is included by using "rtc_base/net_helpers.h". by Jeremy Leconte · 12 days ago
  70. 84273f5 Specify max number of consecutive drops using time units by Sergey Silkin · 3 weeks ago
  71. a986514 Roll chromium_revision 4a8f19d868..5dc6c1eec4 (1353126:1353232) by chromium-webrtc-autoroll · 12 days ago
  72. 28ce65c Apply include-cleaner to api direct files by Dor Hen · 3 weeks ago
  73. 21c456e Update WebRTC code version (2024-09-10T04:06:53). by webrtc-version-updater · 12 days ago
  74. 4ea6534 Roll chromium_revision c339b49443..4a8f19d868 (1353018:1353126) by chromium-webrtc-autoroll · 12 days ago
  75. 110f7db Roll chromium_revision 33ef804c4e..c339b49443 (1352775:1353018) by chromium-webrtc-autoroll · 12 days ago
  76. dc56a36 Use PayloadTypePicker in WebRtcVoiceEngine by Harald Alvestrand · 2 weeks ago
  77. 927244d Set MID in AudioReceiveChannel by Harald Alvestrand · 2 weeks ago
  78. 27db338 Roll chromium_revision c03ff62a28..33ef804c4e (1351560:1352775) by chromium-webrtc-autoroll · 13 days ago
  79. 0f61f60 Mock call to os.path.isdir in roll_deps_test. by Björn Terelius · 13 days ago
  80. 76aa330 Implement ObjCVideoEncoderFactory::QueryCodecSupport by Danil Chapovalov · 13 days ago
  81. 0acbb77 Pass Environment into RtcpSender by Danil Chapovalov · 13 days ago
  82. 363dc19 SimulcastToSvcConverter: Allow not setting scalability mode on frame by Ilya Nikolaevskiy · 13 days ago
  83. 02113a2 Pass Environment into RtcpReceiver by Danil Chapovalov · 2 weeks ago
  84. 3652dd3 Review documentation and update review date by Artem Titov · 13 days ago
  85. 65b59a9 Prepend webrtc ns to StrJoin calls in dcsctp ns by Dor Hen · 14 days ago
  86. 26146bb Add support for screencast with temporal layering to SvcRateAllocator by Sergey Silkin · 13 days ago
  87. 405f343 Update WebRTC code version (2024-09-07T04:08:12). by webrtc-version-updater · 2 weeks ago
  88. 6f64ae1 Extract corruption detection message to its own target by Fanny Linderborg · 2 weeks ago
  89. 65b46da dcsctp: Don't send FORWARD-TSN in its own chunk by Victor Boivie · 2 weeks ago
  90. 7929ef5 dcsctp: Add test for stream reset pending by Victor Boivie · 3 weeks ago
  91. c9aaf11 Remove use of AcmReceiver in ChannelReceive by Henrik Lundin · 2 weeks ago
  92. 3ad2c8d Make getNumObservers @VisibleForTesting so that it can be tested outside of package org.webrtc by Jonas Oreland · 2 weeks ago
  93. 9f096a8 Allow VideoEncoderSoftwareFallbackWrapper to return SIMULCAST_PARAMS_NOT_SUPPORTED by Ilya Nikolaevskiy · 2 weeks ago
  94. c7da857 Fix lint issues in pacing/ by Björn Terelius · 3 weeks ago
  95. f92f39e Increase the default maximum jitter buffer size to 200 packets for Android. by [email protected] · 2 weeks ago
  96. 4334cdf Reland "Return audio stats regarless if we have a codec." by Jakob Ivarsson · 2 weeks ago
  97. 5913803 Update WebRTC code version (2024-09-06T04:04:53). by webrtc-version-updater · 2 weeks ago
  98. f5c5fb9 Roll chromium_revision 040c638bdb..c03ff62a28 (1351313:1351560) by chromium-webrtc-autoroll · 2 weeks ago
  99. e922cd1 Use Environment instead of Clock in ModuleRtpRtcp and its RTP subcomponents by Danil Chapovalov · 2 weeks ago
  100. e94c7da Revert "Return audio stats regarless if we have a codec." by Jakob Ivarsson‎ · 2 weeks ago