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webrtc/src/HEAD
  1. 4fdd50a Use accessor functions instead of direct access to ASN1_STRING fields by xfding · 8 hours ago lkgr main master
  2. 225793c Update WebRTC code version (2026-10-17T04:07:04). by webrtc-version-updater · 10 hours ago
  3. f7126d6 Check libyuv return values in VideoFrameBuffer conversions by Erik Språng · 23 hours ago
  4. 3cdeaaa Enforce kMaxFrameSidePixels on decoded video frames by Erik Språng · 23 hours ago
  5. ad255b4 Handle failed I420 conversion in VideoFrameSampler::Create by Erik Språng · 25 hours ago
  6. ff1c351 Fix flaky ScaleToMaintainsAspectRatio. by Henrik Boström · 26 hours ago
  7. ec0c50d Mark Buffer->string_view conversion explicit by Danil Chapovalov · 26 hours ago
  8. 524dd52 Add fallback for WGC window capturer by Palak Agarwal · 27 hours ago
  9. 65b585a dcsctp: Fast retransmit while in fast recovery by Victor Boivie · 28 hours ago
  10. b308828 Fix flaky ScaleToIsOrientationAgnostic parameterized test. by Henrik Boström · 28 hours ago
  11. bef233b Cleanup StunByteStringAttribute constructors by Danil Chapovalov · 28 hours ago
  12. 541ce6d Implement adaptation in FakePeriodicVideoSource (apply sink wants). by Henrik Boström · 29 hours ago
  13. b67a9ce Revert "Allow combining AV1 SVC frames without dependency descriptor." by Erik Språng · 31 hours ago
  14. b1fb16d Lazily allocate SctpDataChannel::ObserverAdapter by Tommi · 31 hours ago
  15. e741d2d dcsctp: Rate limit duplicate FORWARD-TSN chunks by Victor Boivie · 31 hours ago
  16. 04db1a1 ScreamV2: Add parameter for minimum max_data_in_flight in MSS. by Per K · 31 hours ago
  17. 9b1c953 Deflake RTCStatsIntegrationTest.GetStatsFromCaller. by Henrik Boström · 31 hours ago
  18. 43e7485 Add gMock guidance to the pre-reviewer skill by Tommi · 32 hours ago
  19. 7b923bf Avoid integer-overflow in WebRtcSpl_FilterMAFastQ12 by Joachim Reiersen · 32 hours ago
  20. c81ada6 Update WebRTC code version (2026-10-16T04:06:39). by webrtc-version-updater · 34 hours ago
  21. 9a49bcc Make GetPacketizationFormat a member of RTPSenderVideo by kwasniow · 36 hours ago
  22. 2f22461 Roll back transports without a separate blocking call by Tommi · 2 days ago
  23. ee18f34 Remove redundant CopyOnWriteBuffer constructor by Danil Chapovalov · 2 days ago
  24. b6cbf8b Start gathering and discard the candidate pool without blocking by Tommi · 2 days ago
  25. 52da778 Allow combining AV1 SVC frames without dependency descriptor. by Erik Språng · 2 days ago
  26. be2b87b dtls-in-stun: improve test coverage of Connection by Philipp Hancke · 2 days ago
  27. c5f0fa1 Run audio_decoder_unittest and webrtc_opus_fec_test in audio_coding_unittests by Jeremy Leconte · 2 days ago
  28. 81075eb Add qualityLimitationReason test for "maintain-framerate". by Henrik Boström · 2 days ago
  29. 71180d8 Fix qualityLimitationReason bug in "maintain-resolution" mode. by Henrik Boström · 2 days ago
  30. 45872db Create FakePeerConnectionForStats channels with CreateChannel() by Tommi · 2 days ago
  31. 9890fd1 Replace ReferenceBufferTracker with ReferenceBufferManager by Erik Språng · 2 days ago
  32. 0820fe1 Add Sframe to RtpPacketizer::Create by kwasniow · 2 days ago
  33. c08eb4e Use standard EWMA with g = 1/16 for SCReAMv2 l4s_alpha by Per K · 2 days ago
  34. be0b336 DualPi2: Set default target_delay=15ms and step_threshold=10ms by Per K · 2 days ago
  35. c3a079c Update WebRTC code version (2026-10-15T04:06:02). by webrtc-version-updater · 2 days ago
  36. 36df283 Use the matching relay port when using TURN servers as STUN servers by kwasniow · 2 days ago
  37. 602d5fc Let FakeMediaEngine take custom fake engines by Tommi · 3 days ago
  38. d53e2bf Move the Plan B legacy stats tests to their own file by Tommi · 3 days ago
  39. 1c42989 Split the HasStream parameterization of the legacy stats tests by Tommi · 3 days ago
  40. 6870f3a Keep unchanged video decoders across SetDecoders() by Tommi · 3 days ago
  41. 1926895 Look up DTLS transports without blocking by Tommi · 3 days ago
  42. 1aeb005 Revert "Run audio_decoder_unittest, neteq tests and webrtc_opus_fec_test in audio_coding_unittests" by Jeremy Leconte · 3 days ago
  43. 157bdfb Avoid reusing released decoders after SetDecoders() by Tommi · 3 days ago
  44. e848b24 Run audio_decoder_unittest, neteq tests and webrtc_opus_fec_test in audio_coding_unittests by Jeremy Leconte · 3 days ago
  45. 38985cc Removing undefined left shifts in AudioProcessing by Jeremy Leconte · 3 days ago
  46. 55344d2 dcsctp: Limit when chunks are retransmitted by Victor Boivie · 3 days ago
  47. 1670ec1 Avoid CopyOnWriteBuffer copy in dcsctp_socket_network_test by Fredrik Solenberg · 3 days ago
  48. c6bb4fb Set the local certificate without blocking by Tommi · 3 days ago
  49. 546d2fb Update SCReAMv2 implementation_diff.md against draft -01 by Per K · 3 days ago
  50. 421d35b Remove hta@webrtc.org from WATCHLISTS by Tommi · 3 days ago
  51. 8b2e2d5 Move audio_engine_tests and rtc_p2p_unittests to there respective build file by Jeremy Leconte · 3 days ago
  52. f618cb3 SCReAM: Do not bump l4s_alpha to 0.25 after long uncongested period by Per K · 3 days ago
  53. 9ac655a Plumb receive side ECN marking to RtcTransport by Kaustav Ghosh · 3 days ago
  54. eba0caf Update JsepTransportController tests to use a separate network thread by Tommi · 3 days ago
  55. 0d3c711 Update WebRTC code version (2026-10-14T04:06:57). by webrtc-version-updater · 3 days ago
  56. ac9591c Add RtpPacketizerSframe for per-frame Sframe packetization by kwasniow · 4 days ago
  57. d786236 Reland "Add AES-256-CTR-HMAC-SHA512 Sframe cipher suites" by kwasniow · 4 days ago
  58. 926d586 Fix inverted check in EventLogger::Stop() by Tommi · 4 days ago
  59. 999ac32 Have a unique test file per rtc_cc_test by Jeremy Leconte · 4 days ago
  60. 0941d25 SCReAM: Scale reference window increase linearly for small RTTs by Per K · 4 days ago
  61. c9b8314 Discard the ICE candidate pool after gathering for a local answer by Tommi · 4 days ago
  62. 5859cc6 Add SCReAMv2 g3doc overview and update sitemap by Per K · 4 days ago
  63. ba84c3b Stop reading input_api.tbr in PRESUBMIT.py by Henrique Ferreiro · 4 days ago
  64. 7e6b034 Switch owners email address for jleconte by Jeremy Leconte · 4 days ago
  65. 4f4904d Clean up temporary directories and files leaked by tests into out/ by Tommi · 4 days ago
  66. 51fe478 Check video receive codecs against decoder formats by Tommi · 4 days ago
  67. ec78315 Cleanup usage and deprecate CopyOnWriteBuffer(ptr, size) constructor by Danil Chapovalov · 4 days ago
  68. 9905ba5 Avoid some unnecessary copies in audio channel setup by Fredrik Solenberg · 4 days ago
  69. 3efe1a3 Spanify and thus harden forward error correction code by Danil Chapovalov · 4 days ago
  70. 689d015 Restrict 32-bit STUN integrity to GOOG_PING and ignore post-MI attrs by Tommi · 4 days ago
  71. 6521715 Replace rtc_test by rtc_test_suite by Jeremy Leconte · 4 days ago
  72. 7dee88c Set CQP via cq_level with bounded QP range in AV1 v2 encoder. by Erik Språng · 4 days ago
  73. 9639bc0 Remove unused parameters from BasicIceController by kwasniow · 4 days ago
  74. f86da14 Update WebRTC code version (2026-10-13T04:06:05). by webrtc-version-updater · 4 days ago
  75. b321f63 Reland "Enable compound prediction for 2+ reference frames in LibaomAv1EncoderV2" by Erik Språng · 5 days ago
  76. acc4489 Remove RTPSender's deprecated setters by Tommi · 5 days ago
  77. 8bf6fb4 dcsctp: Do not retransmit expired chunks by Victor Boivie · 5 days ago
  78. c743fa6 Group the Plan B-only legacy stats tests at the end of the file by Tommi · 5 days ago
  79. 5e0a750 Reduce log spam in LibaomAv1EncoderV2::GetSvcParams by Erik Språng · 5 days ago
  80. 6cfac08 SCReAM: Bound drain timer and scale burst backoff by Per K · 5 days ago
  81. 5d25352 Never compile _bin targets on bots. by Jeremy Leconte · 5 days ago
  82. da5e1be Revert "Enable compound prediction for 2+ reference frames in LibaomAv1EncoderV2" by Erik Språng · 5 days ago
  83. 4363692 Move legacy stats test helpers to a shared test library by Tommi · 5 days ago
  84. 97c21f2 Set the MID at construction of the RTP module by Tommi · 5 days ago
  85. 7bcf102 Fix sanitizer issue with StdoutMetricsExporter by Jeremy Leconte · 5 days ago
  86. 7f4ac19 Optionally use a socket-capable worker thread as the network thread by Tommi · 5 days ago
  87. f10949a Cleanup RtpSenderVideo tests by Danil Chapovalov · 5 days ago
  88. c503930 Remove debug_static_bot config by Jeremy Leconte · 5 days ago
  89. f5f0d78 Set AudioRtpReceiver volume without blocking by Tommi · 5 days ago
  90. 00fe7a7 Ignore stale transport-state snapshots in JsepTransportController by Tommi · 5 days ago
  91. 4e909fe Avoid some unnecessary copies in video/ code by Fredrik Solenberg · 5 days ago
  92. 08c3fae Enable compound prediction for 2+ reference frames in LibaomAv1EncoderV2 by Erik Språng · 5 days ago
  93. 8669b30 Only update last_captured_timestamp_ for frames passed to encoding by Tommi · 5 days ago
  94. 64e8dea Make GetAndResetCaptureProcessingTimeStats non-const. by Tore Rudberg · 5 days ago
  95. 5a6496a Set up senders/receivers before channels in stats tests by Tommi · 5 days ago
  96. 87624bd Remove unused stream plumbing from FakePeerConnectionForStats by Tommi · 5 days ago
  97. 6c0dde2 Fix crash when SLD/SRD observers call Close() by Tommi · 5 days ago
  98. 7354db4 Update WebRTC code version (2026-10-12T04:08:08). by webrtc-version-updater · 5 days ago
  99. aeea8ee Update WebRTC code version (2026-10-11T04:07:42). by webrtc-version-updater · 6 days ago
  100. bd53d17 Update WebRTC code version (2026-10-10T04:05:27). by webrtc-version-updater · 7 days ago