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  1. eb79ac6 Update WebRTC code version (2026-06-13T04:08:28). by webrtc-version-updater · 7 hours ago lkgr main master
  2. 7f743b5 Roll chromium_revision f3b3c9f8fe..01106d5b5d (1644831:1646328) by chromium-webrtc-autoroll · 12 hours ago
  3. adaff0d Fix many more test binaries for force-test-env flag by Harald Alvestrand · 16 hours ago
  4. 473f762 Fix audio_engine_tests for force-test-env flag by Harald Alvestrand · 21 hours ago
  5. 799b632 Restrict max resolution in Vp9FrameBufferPool by Sergey Silkin · 23 hours ago
  6. 419cb40 Enforce simulcast RID limits and ignore duplicates by Tommi · 24 hours ago
  7. f8106e8 Fix modules_tests for force-test-env flag by Harald Alvestrand · 24 hours ago
  8. 55510f9 Clean up android build targets by Jeremy Leconte · 24 hours ago
  9. 7708fa5 Adding dct@ to WebRTC owners. by Mirko Bonadei · 25 hours ago
  10. 198a2b6 Add pre-reviewer skill containing code review guidelines+best practices by Tommi · 25 hours ago
  11. e3850ce stats: use correct IANA registry for SRTP cipher by Philipp Hancke · 26 hours ago
  12. 144bade Use injected clock in FakePacketTransport by Danil Chapovalov · 26 hours ago
  13. 75bf8162 Remove overly verbose DLOG in the ZeroHertzAdapterMode. by Philip Eliasson · 28 hours ago
  14. 724ea65 Avoid global time function in network_tester tool by Danil Chapovalov · 28 hours ago
  15. 0b0f9f6 [WebRTC] Migrate from Legacy Natives to Proxy Natives by Martin Kong · 2 days ago
  16. 1ae16ac Fix rtc_p2p_unittests for force-test-env flag by Harald Alvestrand · 2 days ago
  17. 081a7fb Fix rtc_media_unittests for force-test-env flag by Harald Alvestrand · 2 days ago
  18. d2f6e77 Fix peerconnection_unittests for force-test-env by Harald Alvestrand · 2 days ago
  19. efff37e Sframe: rename Encrypter/Decrypter to Encryptor/Decryptor by kwasniow · 2 days ago
  20. 2ecc6b6 Update deps of videocodec_test_mediacodec by Jeremy Leconte · 2 days ago
  21. 78c88b7 Propagate Environment into fake ice transport by Danil Chapovalov · 2 days ago
  22. c1ce35c p2p: remove deprecated SetIceTiebreaker by Philipp Hancke · 2 days ago
  23. 700ce61 Move jitter estimation logic from VideoStreamBufferController. by Åsa Persson · 2 days ago
  24. 364a5d5 Limit SSRCs in ssrc-group to 32 by Tommi · 2 days ago
  25. 4c122cc Fix rtc_pc_unittests for force-test-env flag by Harald Alvestrand · 2 days ago
  26. 9fc52c1 Make videocodec_test_mediacodec a rtc_test by Jeremy Leconte · 2 days ago
  27. 95d243d fix SVC tests flakiness on Android by Jeremy Leconte · 2 days ago
  28. 94c068b Update WebRTC code version (2026-06-11T04:08:41). by webrtc-version-updater · 2 days ago
  29. 71d8293 Add flag to allow enabling the zero hertz feature for non-screenshare content that also contains periods of static video. by Philip Eliasson · 3 days ago
  30. 99cd994 Roll chromium_revision c43135890e..f3b3c9f8fe (1644724:1644831) by chromium-webrtc-autoroll · 3 days ago
  31. 6586b60 Fix rtc_unittests for force-test-env flag by Harald Alvestrand · 3 days ago
  32. 18d2cdb Roll chromium_revision 1c39621c84..c43135890e (1644534:1644724) by chromium-webrtc-autoroll · 3 days ago
  33. 83cdac9 Integrate `RttSimulator` into video timing simulator by Rasmus Brandt · 3 days ago
  34. 6d9c533 Add documentation on SCReAM v2 implementation differences by Per K · 3 days ago
  35. c53df62 Add jitter estimation logic to `DefaultVideoJitterTiming`. by Åsa Persson · 3 days ago
  36. 63b38abc Roll chromium_revision 04a531fe41..1c39621c84 (1644409:1644534) by chromium-webrtc-autoroll · 3 days ago
  37. 41a7487 Bugfix: include `rtx_ssrc` in `all_known_ssrcs_` by Rasmus Brandt · 3 days ago
  38. 9811f2c Add `RttSimulator`. by Rasmus Brandt · 3 days ago
  39. b2215e3 Roll chromium_revision 4b6aacc5a8..04a531fe41 (1644257:1644409) by chromium-webrtc-autoroll · 3 days ago
  40. 02530eb Throttle log frequency for RTCP decryption errors. by Junichi Uekawa · 4 days ago
  41. e9988ad Roll chromium_revision 28acb7a53c..4b6aacc5a8 (1644111:1644257) by chromium-webrtc-autoroll · 4 days ago
  42. 0204f1f Pace CCFB using actual wire overhead and is_bandwidth_limited status by Per K · 4 days ago
  43. 6f0e783 Roll chromium_revision c7197fb1a3..28acb7a53c (1644006:1644111) by chromium-webrtc-autoroll · 4 days ago
  44. 02fc368 Roll chromium_revision d931749ef3..c7197fb1a3 (1643829:1644006) by chromium-webrtc-autoroll · 4 days ago
  45. c1a44ec Delete ThreadProcessingFakeClock as unused by Danil Chapovalov · 4 days ago
  46. 51728d2 Add force-test-environment flag and checks by Harald Alvestrand · 4 days ago
  47. 0179483 Roll chromium_revision 079ad5b815..d931749ef3 (1643675:1643829) by chromium-webrtc-autoroll · 4 days ago
  48. a432b11 Use ScopedTaskSafety in WebRtcSessionDescriptionFactory by Tommi · 4 days ago
  49. 7843cb4 Use CreateTestEnvironment in media tests by Harald Alvestrand · 4 days ago
  50. 420975b RtcpRttCalculator bugfix: do NOT reject reports whose dlsr or dlrr ==0. by Rasmus Brandt · 4 days ago
  51. f48df12 Update WebRTC code version (2026-06-09T04:04:20). by webrtc-version-updater · 4 days ago
  52. 9bac62c Roll chromium_revision 6a71381809..079ad5b815 (1643543:1643675) by chromium-webrtc-autoroll · 4 days ago
  53. 767e526 Roll chromium_revision 9d73023f14..6a71381809 (1643417:1643543) by chromium-webrtc-autoroll · 4 days ago
  54. 2a6c206 Roll chromium_revision d0363615d3..9d73023f14 (1643243:1643417) by chromium-webrtc-autoroll · 5 days ago
  55. c0f9917 Reland "Remove the deprecated libjingle_peerconnection target" by Jeremy Leconte · 5 days ago
  56. 1564410 Fix OOB write in RotateDesktopFrame and DXGI size mismatch by Alexander Cooper · 5 days ago
  57. 3e52f63 Revert "Remove the deprecated libjingle_peerconnection target" by Tomas Gunnarsson · 5 days ago
  58. c6c37c6 Fix possible uaf in WebRtcSessionDescriptionFactory::Post by Tommi · 5 days ago
  59. d89e0f4 Roll chromium_revision 36ff404216..d0363615d3 (1637897:1643243) by chromium-webrtc-autoroll · 5 days ago
  60. 1ea13f1 Use std::span in ToRtpCapabilities by Harald Alvestrand · 5 days ago
  61. 988fbb7 Reland "Merge ScopedFakeClock and ScopedBaseFakeClock" by Danil Chapovalov · 5 days ago
  62. 7060d28 Fix incorrect OwnedFactoryAndThreads teardown sequence by Tommi · 5 days ago
  63. bb4170d Handle possible deadlocks and hangs during PeerConnection teardown by Tommi · 5 days ago
  64. cbb1563 Remove 6 deprecated symbols by Harald Alvestrand · 5 days ago
  65. 77889f4 Remove the deprecated libjingle_peerconnection target by Jeremy Leconte · 5 days ago
  66. 61c57a1 Integrate DefaultVideoJitterTiming into VCMTiming. by Åsa Persson · 5 days ago
  67. 3b0cbbf Use new RtpHeaderExtensionCapability constructors by Harald Alvestrand · 5 days ago
  68. a5f338c Simplify WaitUntil assertions in tests by Harald Alvestrand · 5 days ago
  69. 8ba8406 Don't include private header <asm-generic/errno.h> by Jeremy Leconte · 5 days ago
  70. 0b08bb9 Add an allowlist for expand_directory by Jeremy Leconte · 5 days ago
  71. f295e88 Add `RtcpRttCalculator`. by Rasmus Brandt · 5 days ago
  72. 400089b [dcsctp] Treat forward tsn as payload when saving it for later processing by Danil Chapovalov · 5 days ago
  73. d82ea36 Initialize TaskQueuePacedSender with configuration at construction by Tommi · 5 days ago
  74. e5387e2 Modernize and simplify AudioState by Tommi · 5 days ago
  75. 57bc4f8 Reland "Migrate test_support_unittests to rtc_test_suite" by Jeremy Leconte · 5 days ago
  76. fd40e38 Construct Call on the signaling thread by Tommi · 5 days ago
  77. aa6281e Update WebRTC code version (2026-06-08T04:07:43). by webrtc-version-updater · 5 days ago
  78. c6bfde4 Associate RTX with preceding codec in redesign by Harald Alvestrand · 6 days ago
  79. a7fb01c Update WebRTC code version (2026-06-07T04:05:09). by webrtc-version-updater · 6 days ago
  80. daabde7 Remove TaskQueueBase::Current() from Call construction by Tommi · 7 days ago
  81. f8f44e9 Update WebRTC code version (2026-06-06T04:08:08). by webrtc-version-updater · 7 days ago
  82. d370c22 Hdrext: Replace ABSL_DEPRECATE_AND_INLINE with [[deprecated]] by Harald Alvestrand · 8 days ago
  83. ef4a476 Audio: Apply ABWENoTWCC trial to ReconfigureBitrateObserver by karllen.zheng@ringcentral.com · 8 days ago
  84. 33a14aa Migrate from deprecated SSLFingerprint::CreateUnique* factories by Danil Chapovalov · 8 days ago
  85. 40e0bfd Simplify RtpTransceiver and BaseChannel construction threading by Tommi · 8 days ago
  86. 1c9a7db Revert "Merge ScopedFakeClock and ScopedBaseFakeClock" by Danil Chapovalov · 8 days ago
  87. 590e27d Make ICE tiebreaker a construction time argument of Port by Philipp Hancke · 8 days ago
  88. da4da9c Revert "Migrate test_support_unittests to rtc_test_suite" by Ilya Nikolaevskiy · 8 days ago
  89. 48c80a6 Merge ScopedFakeClock and ScopedBaseFakeClock by Danil Chapovalov · 8 days ago
  90. 605d431 fix(test): Handle worktrees in repo root detection scripts by Harald Alvestrand · 8 days ago
  91. 706cbfa Remove libjingle_peerconnection_api by Jeremy Leconte · 8 days ago
  92. d63cf5a Add TargetTransferRate.is_bwe_limited and support in googcc and scream by Per K · 8 days ago
  93. 7c6a6a3 Update WebRTC code version (2026-06-05T04:10:06). by webrtc-version-updater · 8 days ago
  94. fc7f8509 Add settling delay to CroppingWindowCapturer by Alexander Cooper · 9 days ago
  95. b5cf1fa [WGC] Fix frame size synchronization by Alexander Cooper · 9 days ago
  96. e72179b Migrate test_support_unittests to rtc_test_suite by Jeremy Leconte · 9 days ago
  97. 43f43f8 Delete SSLFingerprint factories returning raw pointer with ownership by Danil Chapovalov · 9 days ago
  98. 5c92dc0 Move api/stats/OWNERS constraints in the parent folder. by Jeremy Leconte · 9 days ago
  99. daefd4f Add some sequence checkers to simulator streams. by Rasmus Brandt · 9 days ago
  100. 15af077 RtpHeaderExtensionId: Clean up usage in pc/rtp_transceiver_unittest.cc by Harald Alvestrand · 9 days ago