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  1. 8e55dca Set the URL in README.chromium files for G.711 and G.722 by Tomas Lundqvist · 26 hours ago lkgr main master
  2. f56007b Update WebRTC code version (2025-02-06T04:03:14). by webrtc-version-updater · 14 hours ago
  3. 0f4f802 Skip owned window with WS_EX_LAYERED attribute when capturing with GDI. by fizzfang · 33 hours ago
  4. 70b65bc Remove deprecated ReceiveSideCongestionController ctor by Per K · 33 hours ago
  5. 3164c2a Restructure PeerConnection tests not to create PortAllocator directly by Danil Chapovalov · 2 days ago
  6. e46d8e4 Update WebRTC code version (2025-02-05T04:02:29). by webrtc-version-updater · 2 days ago
  7. b1ec813 Expose direct access to PeerConnection in PeerConnectionWrapper helper by Danil Chapovalov · 2 days ago
  8. e828c6d red: remove hardcoded parameters in favor of taking them from the codec by Philipp Hancke · 9 days ago
  9. 9c56cb3 Add include for <optional> by Takuto Ikuta · 7 days ago
  10. 0533b5e Add set_timestamp() method to RTCStats. by Henrik Boström · 3 days ago
  11. 9cc5bc8 Remove rust dependency on android. by Jeremy Leconte · 3 days ago
  12. a6f3549 [cpp23] Remove use of std::aligned_storage in webrtc by Victor Hugo Vianna Silva · 3 days ago
  13. c262375 Add test for preferring RTX payload to be "primary codec + 1". by Henrik Boström · 3 days ago
  14. ae24807 [ObjC] Avoid usage of variable after move in RTCNetworkMonitor. by Yury Yarashevich · 3 days ago
  15. f80562d [ObjC] Validate and store strong ref to peer_connection before use. by Yury Yarashevich · 7 days ago
  16. b60a5ab Choose RTX codec PT in lower range if codec is in lower range by Henrik Boström · 3 days ago
  17. 1181edda [ObjC] Fix strong reference check in RTCNetworkMonitor. by Yury Yarashevich · 7 days ago
  18. 6f17d09 [ObjC] Init NSMutableDictionary with capacity. by Yury Yarashevich · 3 days ago
  19. 34c15bc Restructure PeerConnectionBundleTest helper not to create PortAllocator by Danil Chapovalov · 6 days ago
  20. 9830de9 Update WebRTC code version (2025-02-01T04:05:45). by webrtc-version-updater · 6 days ago
  21. f68df0b Restore primary/rtx payload type assignment logic by Philipp Hancke · 7 days ago
  22. c58a767 Reland "Get DeviceScaleFactor for the captured monitor/screen" by Palak Agarwal · 6 days ago
  23. 18b94b5 [rtc_base] Replace manual element initialization and movement with C++17 standard functions by Ho Cheung · 12 days ago
  24. de17350 Revert "Reland "Allow sending to separate payload types for each simulcast index."" by Jonas Oreland · 6 days ago
  25. 9a40734 Revert "Get DeviceScaleFactor for the captured monitor/screen" by Mirko Bonadei · 6 days ago
  26. 45ebd33 Update WebRTC code version (2025-01-31T04:06:50). by webrtc-version-updater · 7 days ago
  27. e20fbb0 Get DeviceScaleFactor for the captured monitor/screen by Palak Agarwal · 8 days ago
  28. d643be9 Add a render error callback from AudioDeviceIOS to AudioDeviceModuleIOS. by Peter Hanspers · 7 days ago
  29. 4b39cb3 Reland "Move piggybacking controller from P2PTC to DTLS transport" by Jonas Oreland · 7 days ago
  30. 4de5839 Revert "Move piggybacking controller from P2PTC to DTLS transport" by Jonas Oreland · 7 days ago
  31. 29e639e Move piggybacking controller from P2PTC to DTLS transport by Philipp Hancke · 8 days ago
  32. feabcdb Reduce redundant memory allocation when capturing a single monitor. by fizzfang · 13 days ago
  33. eb688d6 Remove dependency to NetworkStateEstimator from TransportSequenceNumberFeedbackGenerator by Per K · 8 days ago
  34. 3155346 Reland "Remove rtc_p2p" by Jonas Oreland · 8 days ago
  35. c627895 Revert "Remove rtc_p2p" by Jonas Oreland · 8 days ago
  36. a347fdf Update WebRTC code version (2025-01-29T04:03:12). by webrtc-version-updater · 9 days ago
  37. a05ad63 Remove rtc_p2p by Jonas Oreland · 9 days ago
  38. 406d195 Move the rtc_p2p file last in its BUILD file by Harald Alvestrand · 9 days ago
  39. 5342220 Make corruption_detection_message publicly visible by Fanny Linderborg · 9 days ago
  40. 87b7c1a Reduce warning logging when minimum playout delay exceed maximum by Danil Chapovalov · 9 days ago
  41. 4a21048 DTLS 1.3 - patch 5 by Jonas Oreland · 9 days ago
  42. d8fea51 Revert "Cleanup usage of the global field trials in the PeerConnectionE2EQualityTest helper" by Danil Chapovalov · 9 days ago
  43. 1a72c0c Move a test from media_session_unittest to codec_vendor_unittest by Harald Alvestrand · 9 days ago
  44. 2bebeaf Remove unused create_call.cc by Evan Shrubsole · 10 days ago
  45. fb31c99 Update WebRTC code version (2025-01-28T04:03:17). by webrtc-version-updater · 10 days ago
  46. bea4459 desktop_capture: Fix Xrandr / Xrender order by Florent Castelli · 10 days ago
  47. 26617be Make AV1 even payload size default-on when packetizer is used directly by Danil Chapovalov · 2 weeks ago
  48. 13170bd Refactor media_session to move codec handling to new class by Harald Alvestrand · 10 days ago
  49. 6b70995 VideoEncoder: rtc::StringBuilder instead of rtc::SimpleStringBuilder. by Henrik Boström · 10 days ago
  50. 39da6f3 Move corruption_detection_message from common_video to api/transport/rtp by Fanny Linderborg · 13 days ago
  51. a97304c Cleanup usage of the global field trials in the PeerConnectionE2EQualityTest helper by Danil Chapovalov · 2 weeks ago
  52. 9ff254e srtp: stop using private libsrtp function to determine packet index by Philipp Hancke · 7 weeks ago
  53. a9b40ae Update WebRTC code version (2025-01-26T04:03:33). by webrtc-version-updater · 12 days ago
  54. 7b8b0f6 Update WebRTC code version (2025-01-25T04:02:12). by webrtc-version-updater · 13 days ago
  55. 5090eaf Reland "srtp: spanify Protect + Unprotect" by Philipp Hancke · 2 weeks ago
  56. 4e8c984 Obfuscate private keys in unit tests to avoid false lint errors by Philipp Hancke · 14 days ago
  57. ede69fd Make IsSameRtpCodecIgnoringLevel work for any codec. by Henrik Boström · 13 days ago
  58. f2ecdd7 Use ElementsAreArray in corruption detection unittests by Fanny Linderborg · 13 days ago
  59. 1cc5e54 Add missing newline by Fanny Linderborg · 13 days ago
  60. 589acd5 dtls-stun piggybacking: make it compatible with DTLS 1.3 by Philipp Hancke · 2 weeks ago
  61. eafee5e fix: h26x packet buffer video artifacts by k-wasniowski · 2 weeks ago
  62. 63d3cf0 Update WebRTC code version (2025-01-24T04:10:20). by webrtc-version-updater · 14 days ago
  63. a70cc78 Make mid_ a private member variable by Tommi · 2 weeks ago
  64. 7fe307d Update WebRTC code version (2025-01-23T04:07:51). by webrtc-version-updater · 2 weeks ago
  65. a2a528c Add PortAllocator min/max ports to JAVA API by Youjie Zhou · 3 weeks ago
  66. 79e5e72 Add unidirectional codec support ("offer to send" use case). by Henrik Boström · 2 weeks ago
  67. 49ac6b7 Reland "Allow sending to separate payload types for each simulcast index." by Henrik Boström · 2 weeks ago
  68. 1f9e604 Start deprecation process for non-Optional datachannel parameters by Harald Alvestrand · 2 weeks ago
  69. 0bebca5 Remove gunit.h EXPECT/ASSERT..WAIT macros by Evan Shrubsole · 2 weeks ago
  70. f62fbe9 Update WebRTC code version (2025-01-22T04:05:41). by webrtc-version-updater · 2 weeks ago
  71. 046c979 Delete reference to "no_build_hooks" GN variable (part 2) by Andrew Grieve · 2 weeks ago
  72. 76c8f30 Replace use of .name in test code with .mid() by Tommi · 3 weeks ago
  73. 7a0bdb6 Update PeerConnectionSdpMethods::AddRemoteCandidate by Tommi · 3 weeks ago
  74. 3fef8b2 Adding an error callback to AudioDeviceModuleIOS. by Peter Hanspers · 2 weeks ago
  75. 32f3c6c Add AbslStringify for RtcErrorType and RtcErrorDetail by Harald Alvestrand · 2 weeks ago
  76. a6bccab [DVQA] Dont try to render a 'superfluous' frame. by Jeremy Leconte · 2 weeks ago
  77. 283a84d Add matchers for RTCError, rename old matcher for RTCErrorOr. by Henrik Boström · 2 weeks ago
  78. 860a13c Misc improvements to RtpTransceiver unit tests and test utils. by Henrik Boström · 2 weeks ago
  79. ee7371f Update WebRTC code version (2025-01-21T04:06:38). by webrtc-version-updater · 2 weeks ago
  80. b4127b5 Roll chromium_revision dcda5ff9c0..3462a5bab8 (1408528:1408687) by chromium-webrtc-autoroll · 2 weeks ago
  81. d621b41 Make WebRTC-Video-AV1EvenPayloadSizes default-on. by Erik Språng · 2 weeks ago
  82. d481136 Fix hw decoder rendering delay after frame resize by Anna Lemehova · 3 weeks ago
  83. fa73a2e Convert timeouts in integration_test_helpers to TimeDelta by Evan Shrubsole · 2 weeks ago
  84. f1b3e3e Replace gunit.h macros with WaitUntil in modules/ by Evan Shrubsole · 3 weeks ago
  85. 2a858e2 Migrate last uses of gunit.h macros by Evan Shrubsole · 2 weeks ago
  86. 4f56e15 Roll chromium_revision 48223dfc0a..dcda5ff9c0 (1408397:1408528) by chromium-webrtc-autoroll · 2 weeks ago
  87. 9165a9b Disable OpenSSL tests needing a fake clock when boringssl is not used by Philipp Hancke · 3 weeks ago
  88. e6890ad Update WebRTC code version (2025-01-20T04:06:42). by webrtc-version-updater · 3 weeks ago
  89. 0908c9b Update WebRTC code version (2025-01-19T04:03:25). by webrtc-version-updater · 3 weeks ago
  90. 9aeeb61 Roll chromium_revision bb864a5b8d..48223dfc0a (1408296:1408397) by chromium-webrtc-autoroll · 3 weeks ago
  91. a85040f Revert "Reland "Use Payload Type suggester for all codec merging"" by Fabian Reddig · 3 weeks ago
  92. 48ca8b3 Roll chromium_revision 62907d98e8..bb864a5b8d (1408002:1408296) by chromium-webrtc-autoroll · 3 weeks ago
  93. 1305eb9 Update WebRTC code version (2025-01-18T04:08:19). by webrtc-version-updater · 3 weeks ago
  94. 5f27925 Roll chromium_revision 8bf5d05e2b..62907d98e8 (1407755:1408002) by chromium-webrtc-autoroll · 3 weeks ago
  95. 9f68535 Fix setParameters() throwing when level-id does not match. by Henrik Boström · 3 weeks ago
  96. b0038dd Replace gunit.h macros with WaitUntil in P2P by Evan Shrubsole · 3 weeks ago
  97. d959303 Replace gunit.h macros with WaitUntil in rtc_base/ by Evan Shrubsole · 3 weeks ago
  98. 762753d Slight restriction of access to ContentInfo and prefer mid to name. by Tommi · 3 weeks ago
  99. 88833e6 Update video stats documentation. by Åsa Persson · 3 weeks ago
  100. ceb5a3b Roll chromium_revision 839b9b8bb4..8bf5d05e2b (1406733:1407755) by chromium-webrtc-autoroll · 3 weeks ago