Google Git
Sign in
webrtc/src/HEAD
  1. f8f44e9 Update WebRTC code version (2026-06-06T04:08:08). by webrtc-version-updater · 8 hours ago lkgr main master
  2. d370c22 Hdrext: Replace ABSL_DEPRECATE_AND_INLINE with [[deprecated]] by Harald Alvestrand · 15 hours ago
  3. ef4a476 Audio: Apply ABWENoTWCC trial to ReconfigureBitrateObserver by karllen.zheng@ringcentral.com · 25 hours ago
  4. 33a14aa Migrate from deprecated SSLFingerprint::CreateUnique* factories by Danil Chapovalov · 25 hours ago
  5. 40e0bfd Simplify RtpTransceiver and BaseChannel construction threading by Tommi · 26 hours ago
  6. 1c9a7db Revert "Merge ScopedFakeClock and ScopedBaseFakeClock" by Danil Chapovalov · 27 hours ago
  7. 590e27d Make ICE tiebreaker a construction time argument of Port by Philipp Hancke · 27 hours ago
  8. da4da9c Revert "Migrate test_support_unittests to rtc_test_suite" by Ilya Nikolaevskiy · 27 hours ago
  9. 48c80a6 Merge ScopedFakeClock and ScopedBaseFakeClock by Danil Chapovalov · 27 hours ago
  10. 605d431 fix(test): Handle worktrees in repo root detection scripts by Harald Alvestrand · 29 hours ago
  11. 706cbfa Remove libjingle_peerconnection_api by Jeremy Leconte · 30 hours ago
  12. d63cf5a Add TargetTransferRate.is_bwe_limited and support in googcc and scream by Per K · 30 hours ago
  13. 7c6a6a3 Update WebRTC code version (2026-06-05T04:10:06). by webrtc-version-updater · 32 hours ago
  14. fc7f8509 Add settling delay to CroppingWindowCapturer by Alexander Cooper · 2 days ago
  15. b5cf1fa [WGC] Fix frame size synchronization by Alexander Cooper · 2 days ago
  16. e72179b Migrate test_support_unittests to rtc_test_suite by Jeremy Leconte · 2 days ago
  17. 43f43f8 Delete SSLFingerprint factories returning raw pointer with ownership by Danil Chapovalov · 2 days ago
  18. 5c92dc0 Move api/stats/OWNERS constraints in the parent folder. by Jeremy Leconte · 2 days ago
  19. daefd4f Add some sequence checkers to simulator streams. by Rasmus Brandt · 2 days ago
  20. 15af077 RtpHeaderExtensionId: Clean up usage in pc/rtp_transceiver_unittest.cc by Harald Alvestrand · 2 days ago
  21. 3a013ef Merge rtc_stats and rtc_stats_api build targets. by Jeremy Leconte · 2 days ago
  22. 5a6c17b Add initial DefaultVideoJitterTiming class. by Åsa Persson · 2 days ago
  23. c952a43 Delete deprecated test::DirectTransport constructor by Danil Chapovalov · 2 days ago
  24. 50a60b7 Reenable markdown autoformat by Danil Chapovalov · 2 days ago
  25. e61e8bd [visualizer] Add RTCP bitrate time series and packet overhead to total graphs by Per K · 2 days ago
  26. 6f7e623 PT: Fix test failures and PT mapping conflicts under WebRTC-PayloadTypesInTransport by Harald Alvestrand · 3 days ago
  27. 97473db Remove SetOptions and other unused methods from channel interfaces by Tommi · 3 days ago
  28. b25fa31 Cleanup non-deprecated test::DirectTransport constructor by Danil Chapovalov · 3 days ago
  29. 08007e5 Relax BasicPortAllocator to take TaskQueue instead of Thread by Danil Chapovalov · 3 days ago
  30. caf8ef4 Stop relying on the generic libjingle_peerconnection_api dependency. by Jeremy Leconte · 3 days ago
  31. 08a923a Configure render delay constant via constructor of VCMTiming. by Åsa Persson · 3 days ago
  32. 9616223 Store const audio/video options inside RtpTransceiver on construction by Tommi · 3 days ago
  33. 3661efd Define spaceship <=> operator for StrongAlias by Harald Alvestrand · 3 days ago
  34. 53ca0d6 Deprecate implicit conversion from RtpHeaderExtensionId to int by Harald Alvestrand · 3 days ago
  35. 7f22a4a Perform `{Assembled,Rendered}Margin` calculations in ms-level precision. by Rasmus Brandt · 3 days ago
  36. f8b4a06 RtpHeaderExtensionId: Clean up usage in logging, pc, and video by Harald Alvestrand · 3 days ago
  37. 206ec73 Update WebRTC code version (2026-06-03T04:07:23). by webrtc-version-updater · 3 days ago
  38. 713b237 RtpHeaderExtensionId: Clean up usage in pc/media_session_unittest.cc by Harald Alvestrand · 4 days ago
  39. 6388de4 RtpHeaderExtensionId: Clean up usage in various tests by Harald Alvestrand · 4 days ago
  40. e1e37d7 RtpHeaderExtensionId: Clean up usage in tests and engines by Harald Alvestrand · 4 days ago
  41. c50fceb PT: Replicate legacy answer negotiation for directionality by Harald Alvestrand · 4 days ago
  42. 047dd98 RtpHeaderExtensionId: Clean up usage in remaining tests and build files by Harald Alvestrand · 4 days ago
  43. 4babf81 Make Video Encoder V2 config structs classes with getter/setters. by Erik Språng · 4 days ago
  44. 7bdb4a0 cleanup: Remove several deprecated symbols by Harald Alvestrand · 4 days ago
  45. ec9a800 Fix tests to respect command line field trials by Harald Alvestrand · 4 days ago
  46. 2f3d2c0 Reland "Apply global audio options at the engine level" by Tommi · 4 days ago
  47. 3b1eab8 Namespace: Fix tests that didn't wrap properly by Harald Alvestrand · 4 days ago
  48. 8a3d684 Update WebRTC code version (2026-06-02T04:09:49). by webrtc-version-updater · 4 days ago
  49. f2c276e Deprecate VideoFrameMetadata::GetFrameDependencies by Philipp Hancke · 5 days ago
  50. b1dde84 [SCReAMv2] Inline reference window backoff calculation due to delay in ScreamV2 by Per K · 5 days ago
  51. fc8ed82 Ignore internal transport packets in DatagramConnection by Tommi · 5 days ago
  52. a0773d3 SCReAM v2: Parse TransportPacketsFeedback once into ScreamFeedback by Per K · 5 days ago
  53. 25e13f1 PT: Fix video and audio RED codec handling in TypedCodecVendor and CodecVendor by Harald Alvestrand · 5 days ago
  54. cd56c39 Refactor state caching for SctpDataChannel observers by Tommi · 5 days ago
  55. e895914 Update WebRTC code version (2026-06-01T04:07:44). by webrtc-version-updater · 5 days ago
  56. ef40ea6 Fix race condition in unsignaled stream creation when worker!=network by Tommi · 6 days ago
  57. 9017801 Add SFrame packet buffer for RTP-level frame assembly by kwasniow · 6 days ago
  58. 777159a Update WebRTC code version (2026-05-30T04:04:21). by webrtc-version-updater · 7 days ago
  59. 3a230b4 Rename LOG_ERROR to RTC_LOG_ERROR by Boris Tsirkin · 8 days ago
  60. f88b3b1 Make video_options_ const in SdpOfferAnswerHandler by Tommi · 8 days ago
  61. cd5de07 Fix GetSingleActiveLayerPixels by Sergey Silkin · 8 days ago
  62. 94ebfb0 Fix race condition in CroppingWindowCapturer by Alexander Cooper · 8 days ago
  63. aa09cec Make audio_options_ const in SdpOfferAnswerHandler by Tommi · 8 days ago
  64. 3ee6393 Fix inference of scalability mode by Sergey Silkin · 8 days ago
  65. 74da57b Delete legacy_delay_estimator dead code by Sam Zackrisson · 8 days ago
  66. afc3b18 Clean up the finch experiment kUseHeuristicForFindingEditor by Palak Agarwal · 8 days ago
  67. f17e706 Revert "Apply global audio options at the engine level" by Tomas Gunnarsson · 8 days ago
  68. e737bee Mark ios_force_software_aec_HACK as deprecated by Tommi · 8 days ago
  69. 1488f77 Apply global audio options at the engine level by Tommi · 8 days ago
  70. 9d20c62 Cut lower than threshold audio at the end (not only at the beginning). by Mirko Bonadei · 8 days ago
  71. 98ad243 Remove unnecessary using webrtc:: directives by Philipp Hancke · 8 days ago
  72. f9accc1 HeaderExtensionId: Deprecate header extension functions taking int by Harald Alvestrand · 9 days ago
  73. a4c627e Roll chromium_revision 1a0f04a8da..36ff404216 (1637738:1637897) by chromium-webrtc-autoroll · 9 days ago
  74. e27f1de SCReAMv2: Replace EWMA loss rate filter with asymmetric step filter by Per K · 9 days ago
  75. 8f67a09 Roll chromium_revision a78b216bb6..1a0f04a8da (1637410:1637738) by chromium-webrtc-autoroll · 9 days ago
  76. ad88b60 Polish deprecated section in the style guide by Danil Chapovalov · 9 days ago
  77. 18e3afc Fix potential buffer underflow in handling of STUN_ATTR_GOOG_MISC_INFO by Jonas Oreland · 9 days ago
  78. 91719be rtc_event_log_visualizer: Fix crash due to infinite queue delay by Per K · 9 days ago
  79. 4dc35d5 Include Sframe library in libwebrtc by kwasniow · 9 days ago
  80. 868948f Remove VirtualSocketServer dependency on FakeClock as unused by Danil Chapovalov · 9 days ago
  81. 888fb56 Use StrongAlias for RTP header extension identification. by Harald Alvestrand · 9 days ago
  82. 5cbd174 SCReAMv2: Visualize newly lost, recovered, and CE marked packet events by Per K · 9 days ago
  83. a5c033d pc: ignore DTLS-decrypted packets in RtpTransport::OnReadPacket by Philipp Hancke · 9 days ago
  84. e366adb Single threaded RtpTransceiver construction by Tommi · 9 days ago
  85. 42bc0c0 MediaEngine: pass parameters-changed callback at construction by Tommi · 9 days ago
  86. b1e3a22 Update WebRTC code version (2026-05-28T04:08:22). by webrtc-version-updater · 9 days ago
  87. a9a0850 Roll chromium_revision 7b5d707169..a78b216bb6 (1636994:1637410) by chromium-webrtc-autoroll · 9 days ago
  88. 1191543 PT redesign: handle raw, change allocation and update golden tests by Harald Alvestrand · 10 days ago
  89. b733cfd Roll chromium_revision 63d703fa3d..7b5d707169 (1636747:1636994) by chromium-webrtc-autoroll · 10 days ago
  90. c18b45b Bind default sink to existing unsignaled receive stream by karllen.zheng@ringcentral.com · 10 days ago
  91. a0fb3c0 Allow media channel creation from the signaling thread by Tommi · 10 days ago
  92. 9da2684 NetEq: Align correlation buffer size in Merge with DspHelper expectations by Henrik Lundin · 10 days ago
  93. 85a0176 Enable dynamic speed controller by default, with new AV1 defaults. by Erik Språng · 10 days ago
  94. ad8ace8 Add testing.md to project GEMINI.md by Harald Alvestrand · 10 days ago
  95. ab604b9 refactor(pc): unify audio RED linking in payload type redesign by Harald Alvestrand · 10 days ago
  96. 0275baa Restrict number of actions in VP9 fuzzer by Sergey Silkin · 10 days ago
  97. 09fb29d Iterate over all VP9 GoF `pid_diff`s to determine frame references. by Philip Eliasson · 10 days ago
  98. 5f141a4 Roll chromium_revision 19b51b8218..63d703fa3d (1636642:1636747) by chromium-webrtc-autoroll · 10 days ago
  99. c611ffc Remove obsolete import of //build/config/chromeos/ui_mode.gni by Georg Neis · 10 days ago
  100. 6c973e0 Implement cryptex header extension negotiation by Philipp Hancke · 10 days ago