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  1. 0e231c1 Update WebRTC code version (2025-03-16T04:06:29). by webrtc-version-updater · 12 hours ago lkgr main master
  2. 9634696 Update WebRTC code version (2025-03-15T04:02:31). by webrtc-version-updater · 36 hours ago
  3. 99d1807 Move CorruptionDetectionFilterSettings into webrtc namespace. by Joachim Reiersen · 2 days ago
  4. b401d3d Add a callback to use StreamStats::rtp_stats as the source of truth. by Jeremy Leconte · 2 days ago
  5. c839686 Introduce SctpOptions struct as argument to SctpTransport::Start by Philipp Hancke · 2 days ago
  6. a22e244 Enable test that there are no duplicates in a CodecList by Harald Alvestrand · 2 days ago
  7. c2f1bc7 Split EmulatedNetworkInterface into two by Danil Chapovalov · 2 days ago
  8. c3cb9b7 Add .rustfmt.toml file by Mirko Bonadei · 2 days ago
  9. 64e4714 Add field trial that change g2g metric to use abs. capture time. by Olov Brändström · 2 days ago
  10. b922597 dtls-in-stun: Fix late SDP answer by Jonas Oreland · 2 days ago
  11. 024ceea Run DataChannelIntegrationTests w/o media enginge (for most tests) by Jonas Oreland · 2 days ago
  12. d5d9526 Update freshness for the h-cc-pairs style document by Danil Chapovalov · 2 days ago
  13. 42a1cde Update WebRTC code version (2025-03-14T04:09:46). by webrtc-version-updater · 3 days ago
  14. 41bc570 Enable 0Hz capture for test::FrameGeneratorCapturer. by Jeremy Leconte · 3 days ago
  15. c2c052c Refresh abseil-in-webrtc rules and documentation by Danil Chapovalov · 3 days ago
  16. c13956e Move all files in p2p/test to webrtc namespace by Evan Shrubsole · 3 days ago
  17. 0d3b384 Avoids log spam in AudioDeviceGeneric by henrika · 3 days ago
  18. b6909ee dtls-in-stun: Remove dtls restart in dtls-in-stun handling by Jonas Oreland · 3 days ago
  19. 2cf4492 Log VideoEncoderConfig::simulcast_layers[] by Sergey Silkin · 3 days ago
  20. 8990f2a Move rtc_certificate and rtc_certificate_generator to webrtc namespace by Evan Shrubsole · 3 days ago
  21. c713071 dtls_transport: Extend logging a tiny bit by Jonas Oreland · 3 days ago
  22. dd99114 Add RtcEventProcessor TieBreaker for LoggedRtcpPacketSenderReport. by Rasmus Brandt · 3 days ago
  23. d200661 Move sanitizer.h to webrtc namespace by Evan Shrubsole · 3 days ago
  24. 64b076f4 Move socket_address.h to webrtc namespace by Evan Shrubsole · 3 days ago
  25. e1912c5 Fix issue with protobuf that is blocking perf tests. by Mirko Bonadei · 3 days ago
  26. 63f9cc0 Apply include-cleaner to call/ by Danil Chapovalov · 3 days ago
  27. 1bd545a Revert "fix: h26x packet buffer video artifacts" by Philip Eliasson · 3 days ago
  28. e0cbfe4 Update WebRTC code version (2025-03-13T04:04:53). by webrtc-version-updater · 3 days ago
  29. 01f2ec6 Prevent VideoQualityMetricsReporter from crashing because of negative DataRate. by Jeremy Leconte · 4 days ago
  30. 7e54bb9 Remove unused constant by Rasmus Brandt · 4 days ago
  31. 762d77e Revert the deletion of WebRTC-VideoH26xPacketBuffer flag. by Henrik Boström · 4 days ago
  32. b2e88f9 Improve test outcomes for WebRTC-PayloadTypesInTransport by Harald Alvestrand · 4 days ago
  33. cc5db7a Remove deprecated non-Optional datachannel parameters by Harald Alvestrand · 4 days ago
  34. 31d20a6 Move moving_max_counter.h to webrtc namespace by Evan Shrubsole · 4 days ago
  35. 9278968 Delete deprecated functions in test EmulatedNetworkManagerInterface by Danil Chapovalov · 4 days ago
  36. 9ca15b6 Increase precision of high pass filter by Gustaf Ullberg · 4 days ago
  37. eb835d0 Move ssl_stream_adapter.h to webrtc namespace by Evan Shrubsole · 4 days ago
  38. 97feff2 Move function_view.h to webrtc namespace by Evan Shrubsole · 4 days ago
  39. b1d704b Move port_interface.h to webrtc namespace by Evan Shrubsole · 4 days ago
  40. 9f114ece Delete sigslot_tester by Evan Shrubsole · 4 days ago
  41. c9f82e6 Update WebRTC code version (2025-03-12T04:09:25). by webrtc-version-updater · 5 days ago
  42. 851bcba dtls-in-stun: Redo handling of when ice complete before dtls by Jonas Oreland · 5 days ago
  43. 6202bf1 Move ip_address.h to webrtc namespace by Evan Shrubsole · 5 days ago
  44. a0a83aa Allow test::EmulatedEndpoint to rebind to a different network by Danil Chapovalov · 5 days ago
  45. 0a76875 Skip assembling frames if the stream has been decoded past that point. by Philip Eliasson · 5 days ago
  46. a3866aa Refactoring: Look up codec vendor via RtpTransceiver by Harald Alvestrand · 5 days ago
  47. 8a9e08b Add QP threshold experiment for H265. by Henrik Boström · 5 days ago
  48. e1dc6d5 Deprecate sigslot_tester by Evan Shrubsole · 5 days ago
  49. b000e8e Move test_client.h to webrtc namespace by Evan Shrubsole · 5 days ago
  50. f13201d Temporarily disable two failing PCFullStackTests that hit DCHECKs. by Henrik Boström · 5 days ago
  51. 576ec5a PipeWire capture: Clean latest_available_frame on capture stop by Jan Grulich · 5 days ago
  52. 9e1b0ec Move stream.h to webrtc namespace by Evan Shrubsole · 6 days ago
  53. 377b69a Move test_certificate_verifier.h to webrtc namespace by Evan Shrubsole · 6 days ago
  54. 6738e57 Move base64.h to webrtc namespace by Evan Shrubsole · 6 days ago
  55. f36d45f Disable 'rtc_unittests' on iOS simulator. by Jeremy Leconte · 6 days ago
  56. d9232b3 Refactor VideoQualityMetricsReporter with no impact. by Jeremy Leconte · 6 days ago
  57. 47ddc6e Move test_echo_server.h to webrtc namespace by Evan Shrubsole · 6 days ago
  58. 4887bc3 Move socket_address_pair.h to webrtc namespace by Evan Shrubsole · 6 days ago
  59. 0a97878 Fix build with Pipewire 1.4 by K900 · 6 days ago
  60. 4203617 dtls-in-stun: Add new testcase by Jonas Oreland · 6 days ago
  61. 58c273b Move bit_buffer.h to webrtc namespace by Evan Shrubsole · 7 days ago
  62. c3fba45 Update WebRTC code version (2025-03-10T04:05:53). by webrtc-version-updater · 7 days ago
  63. 49ba3e2 Move byte_order.h to webrtc namespace by Evan Shrubsole · 7 days ago
  64. 5015b4d Move crc32.h to webrtc namespace by Evan Shrubsole · 7 days ago
  65. ccf5114 Update WebRTC code version (2025-03-09T04:08:42). by webrtc-version-updater · 8 days ago
  66. cec4b95 Update WebRTC code version (2025-03-08T04:06:00). by webrtc-version-updater · 8 days ago
  67. ee5e7e5 sdp munging: detect audio nack and changes to opus FEC and DTX by Philipp Hancke · 9 days ago
  68. b101a7e doc: SRTP_AES128_CM_HMAC_SHA1_32 is disabled by default by Philipp Hancke · 9 days ago
  69. f45f111 Sequence checker shouldn't guard itself by Artem Titov · 9 days ago
  70. 4df984d dtls-in-stun: Add callback for disabling of piggybacking by Jonas Oreland · 9 days ago
  71. 1816152 stats: do not expose ufrag before the first setRemoteDescription by Philipp Hancke · 10 days ago
  72. 2167733 Format fuzzer documentation by Björn Terelius · 10 days ago
  73. 3af24f3 Ensure one can build audioproc_f with apm_debug_dump=true by Lionel Koenig Gélas · 10 days ago
  74. a5da882 Update fuzzer documentation by Björn Terelius · 10 days ago
  75. e733ebc dtls1.3 - patch 6 by Jonas Oreland · 10 days ago
  76. 9789a44 Catch also IllegalStateException when creating hw encoder by Liad Rubin · 10 days ago
  77. 4a3dffe Add SurfaceEglRenderer ctor with VideoFrameDrawer and propagate to super class by Liad Rubin · 10 days ago
  78. 924b73c Checkout base/ only if fuzzing is enabled by Byoungchan Lee · 10 days ago
  79. fb8032e Move non-PT-assigning codec collection out of VoiceMediaEngine by Harald Alvestrand · 10 days ago
  80. 5568c98 Update WebRTC code version (2025-03-06T04:08:40). by webrtc-version-updater · 10 days ago
  81. 34bfdcd Roll chromium_revision d4f6168ad3..a1723b7af1 (1428571:1428676) by chromium-webrtc-autoroll · 11 days ago
  82. 0e8160a Roll chromium_revision 0b0f98e256..d4f6168ad3 (1428408:1428571) by chromium-webrtc-autoroll · 11 days ago
  83. 31f317e Roll chromium_revision a82e0ea13a..0b0f98e256 (1428241:1428408) by chromium-webrtc-autoroll · 11 days ago
  84. f1a3142 IWYU pc/test/ by Philipp Hancke · 11 days ago
  85. 72a1d7d Deprecate the cricket::Codec access functions in MediaEngine by Harald Alvestrand · 11 days ago
  86. d9e96ef Roll chromium_revision 547cac853e..a82e0ea13a (1428124:1428241) by chromium-webrtc-autoroll · 11 days ago
  87. cba7209 Fix -Wthread-safety-analysis warning on sequence_checker.h by Zequan Wu · 11 days ago
  88. 3a7525e dtls-in-stun: Improve test coverage by Jonas Oreland · 11 days ago
  89. 683a980 Implement FrameGeneratorCapturer::RequestRefreshFrame. by Jeremy Leconte · 11 days ago
  90. 55cb716 Use a vector of scalability modes in VideoQualityMetricsReporter. by Jeremy Leconte · 11 days ago
  91. eec79a0 dtls-in-stun: Only read IceConfig.dtls_handshake_in_stun in 1 place. by Jonas Oreland · 11 days ago
  92. b3373d8 Update WebRTC code version (2025-03-05T04:05:48). by webrtc-version-updater · 12 days ago
  93. f4687eaf4 Roll chromium_revision 887a261e75..547cac853e (1427905:1428124) by chromium-webrtc-autoroll · 12 days ago
  94. 74c5a21 Support negative SenderCaptureTimeOffset in TransformableVideoFrameInterface by Guido Urdaneta · 12 days ago
  95. a6b939a Roll chromium_revision 3678ad0b60..887a261e75 (1427739:1427905) by chromium-webrtc-autoroll · 12 days ago
  96. 243485a IWYU p2p/client and p2p/stun_prober by Philipp Hancke · 12 days ago
  97. 7d11c9f Roll chromium_revision 767b207241..3678ad0b60 (1426198:1427739) by chromium-webrtc-autoroll · 12 days ago
  98. fa3b0a5 Add DEPS needed for `ninja <...> all` by Björn Terelius · 12 days ago
  99. f0964b5 Roll chromium_revision 36f59af467..767b207241 (1424618:1426198) by Björn Terelius · 12 days ago
  100. a49e78a Make WebRTC-Video-H26xPacketBuffer enabled-by-default (delete flag). by Henrik Boström · 12 days ago