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  1. bff6489 AV1: Disable several intra coding tools. by Fyodor Kyslov · 3 years, 2 months ago
  2. 995c5c8 Roll chromium_revision e4fd023c85..d935055b21 (862756:862883) by chromium-webrtc-autoroll · 3 years, 2 months ago
  3. db5d728 Add refined handling of the internal scaling of the audio in APM by Per Åhgren · 3 years, 2 months ago
  4. b315951 Remove incorrect DCHECKs from LibaomAv1Encoder::SetRates. by philipel · 3 years, 2 months ago
  5. fdd6099 Rework transient suppressor configuration in audioproc_f by Gustaf Ullberg · 3 years, 2 months ago
  6. 685be14 Disable flaky AddMediaToConnectedBundleDoesNotRestartIce on tsan by Rasmus Brandt · 3 years, 2 months ago
  7. e657d87 Allow port 53 as a TURN port. by Harald Alvestrand · 3 years, 2 months ago
  8. c88bdad Roll chromium_revision c3fb27225e..e4fd023c85 (861941:862756) by chromium-webrtc-autoroll · 3 years, 2 months ago
  9. 6ca955a Reland "Fix problem with ipv4 over ipv6 on Android" by Jonas Oreland · 3 years, 2 months ago
  10. 7087b83 Test that SCTP succeeds with one MTU and fails with a lower MTU by Harald Alvestrand · 3 years, 2 months ago
  11. 0e42cf7 Reland "Parse encoded frame QP if not provided by encoder" by Sergey Silkin · 3 years, 2 months ago
  12. b6bc357 turn: add logging for long usernames by Philipp Hancke · 3 years, 2 months ago
  13. 6097b0f Delete use of AsyncInvoker from PeerConnectionIntegrationWrapper by Niels Möller · 3 years, 2 months ago
  14. 13118a7 Update WebRTC code version (2021-03-15T04:05:00). by webrtc-version-updater · 3 years, 2 months ago
  15. 55bc077 Add one frame (10 ms) of silence in APM output after unmuting by Per Åhgren · 3 years, 2 months ago
  16. 1e60490 Revert "Fix problem with ipv4 over ipv6 on Android" by Taylor Brandstetter · 3 years, 2 months ago
  17. bc1c93d Add remote-outbound stats for audio streams by Alessio Bazzica · 3 years, 2 months ago
  18. c80f955 Avoid log spam when decoder implementation changes by Erik Språng · 3 years, 2 months ago
  19. 5eda59c Replace legacy RtpRtcp::GetRemoteStat function with GetLatestReportBlockData by Danil Chapovalov · 3 years, 3 months ago
  20. fd87944 Removed WebRTC-NetworkCondition-EncoderSwitch field trial. by philipel · 3 years, 2 months ago
  21. 7c7885c Remove NTP timestamp from PacketBuffer::Packet. by philipel · 3 years, 2 months ago
  22. 662b306 Replace blocking invokes with PostTask in AndroidNetworkMonitor by Niels Möller · 3 years, 2 months ago
  23. 048e9c2 In full svc controller reuse unused frame configuration by Danil Chapovalov · 3 years, 2 months ago
  24. 8da67f6 In ksvc controller reuse unused frame configuration by Danil Chapovalov · 3 years, 2 months ago
  25. 8647340 Introduce WebRTC documentation structure and how-to by Artem Titov · 3 years, 2 months ago
  26. cf70793 Update WebRTC code version (2021-03-12T04:03:49). by webrtc-version-updater · 3 years, 2 months ago
  27. 8973655 measure ice candidate poolsize setting for different bundle policys by Philipp Hancke · 3 years, 2 months ago
  28. 727d2af Revert "Parse encoded frame QP if not provided by encoder" by Sergey Silkin · 3 years, 2 months ago
  29. 2d9f53c Expose addIceCandidate with completion handler. by Yura Yaroshevich · 3 years, 2 months ago
  30. 31c5c9d Revert "Reland "Enable quality scaling when allowed"" by Ilya Nikolaevskiy · 3 years, 2 months ago
  31. 0021fe7 Reland "Enable quality scaling when allowed" by Sergey Silkin · 3 years, 2 months ago
  32. 7bf29bc Roll chromium_revision fc9c86fd36..c3fb27225e (861807:861941) by chromium-webrtc-autoroll · 3 years, 2 months ago
  33. b7227a5 Fix handling of partial match for GetVpnUnderlyingAdapterType by Jonas Oreland · 3 years, 2 months ago
  34. fd1e9d1 [Stats] Add minimum RTCReceivedRtpStreamStats with jitter and packetsLost by Di Wu · 3 years, 2 months ago
  35. 8639673 Parse encoded frame QP if not provided by encoder by Sergey Silkin · 3 years, 2 months ago
  36. 8097935 Revert "Reduce complexity in the APM pipeline when the output is not used" by Ilya Nikolaevskiy · 3 years, 2 months ago
  37. be140b4 Change ObjCNetworkMonitor::OnPathUpdate to use PostTask by Niels Möller · 3 years, 2 months ago
  38. aa6adff Reduce complexity in the APM pipeline when the output is not used by Per Åhgren · 3 years, 2 months ago
  39. 54dbc3b Revert "[Battery]: Delay start of TaskQueuePacedSender." by Artem Titov · 3 years, 2 months ago
  40. 3135772 Changed setActive of RTCAudio Session, and it's working by Abby Yeh · 3 years, 2 months ago
  41. fbb2dcf Roll chromium_revision 43d5378f7f..fc9c86fd36 (861531:861807) by chromium-webrtc-autoroll · 3 years, 2 months ago
  42. b2e71b8 Reland "Fix race between destroying SctpTransport and receiving notification on timer thread." by Taylor Brandstetter · 3 years, 2 months ago
  43. 34fdc92 Add audioproc_f support for testing the runtime settings of whether the output is used by Per Åhgren · 3 years, 2 months ago
  44. 3d792e9 Add controls for MTU size of virtual socket server by Harald Alvestrand · 3 years, 2 months ago
  45. bccfd26 Allow webrtc mac cross compile by Andrey Logvin · 3 years, 2 months ago
  46. 048adc7 Add missing remote-outbound stats to RTCPReceiver::NTP by Alessio Bazzica · 3 years, 2 months ago
  47. da2fd2a Fix problem with ipv4 over ipv6 on Android by Jonas Oreland · 3 years, 2 months ago
  48. 89cb65e [Battery]: Delay start of TaskQueuePacedSender. by Etienne Pierre-Doray · 3 years, 2 months ago
  49. 79011ef Remove `ModuleRtpRtcpImpl2::LastReceivedNTP` by Alessio Bazzica · 3 years, 2 months ago
  50. ee8cd20 Add a mutex free implementation of webrtc::ReceiveStatistics by Per Kjellander · 3 years, 2 months ago
  51. bb22066 Roll chromium_revision e1b9354ff4..43d5378f7f (854007:861531) by chromium-webrtc-autoroll · 3 years, 2 months ago
  52. 213dc2c Temporarily disable Opus decode test. by Jakob Ivarsson · 3 years, 2 months ago
  53. 8bfa275 Fix nullability of completion handlers in iOS SDK. by Yura Yaroshevich · 3 years, 2 months ago
  54. 8912719 Revert "Roll chromium_revision e1b9354ff4..74090df66c (854007:861144)" by Ilya Nikolaevskiy · 3 years, 2 months ago
  55. 6adb8d9 Revert "Roll chromium_revision 74090df66c..f86a579769 (861144:861254)" by Ilya Nikolaevskiy · 3 years, 2 months ago
  56. 734ae52 Revert "Roll chromium_revision f86a579769..dc3e6d8b69 (861254:861387)" by Ilya Nikolaevskiy · 3 years, 2 months ago
  57. 6a55e73 Stop inheriting from has_slots in DtlsTransport. by Mirko Bonadei · 3 years, 2 months ago
  58. 68ef4e5 Update WebRTC code version (2021-03-10T04:02:37). by webrtc-version-updater · 3 years, 2 months ago
  59. 8114016 Roll chromium_revision f86a579769..dc3e6d8b69 (861254:861387) by chromium-webrtc-autoroll · 3 years, 2 months ago
  60. 14a626a Roll chromium_revision 74090df66c..f86a579769 (861144:861254) by chromium-webrtc-autoroll · 3 years, 2 months ago
  61. 5265b93 Add build-id to libjingle_peerconnection_so.so by Raman Budny · 3 years, 2 months ago
  62. cb7ff13 Roll chromium_revision e1b9354ff4..74090df66c (854007:861144) by Artem Titov · 3 years, 2 months ago
  63. dc08aea Fix chromium roll into WebRTC by Artem Titov · 3 years, 2 months ago
  64. c84a887 Allow port 80 for TURN servers by Harald Alvestrand · 3 years, 2 months ago
  65. 668dbf6 [Stats] Populate "frames" stats for video source. by Di Wu · 3 years, 2 months ago
  66. 5ab9b32 Update WebRTC code version (2021-03-09T04:03:43). by webrtc-version-updater · 3 years, 2 months ago
  67. 92d1270 Expose PeerConnection.restartIce in iOS SDK. by Yura Yaroshevich · 3 years, 2 months ago
  68. d672535 Expose parameterless setLocalDescription() in iOS SDK. by Yura Yaroshevich · 3 years, 2 months ago
  69. f00cd53 Do more actions on SDP fuzzing. by Harald Alvestrand · 3 years, 2 months ago
  70. a113d23 srtp: use srtp_crypto_policy_set_from_profile_for_* from libsrtp by Philipp Hancke · 3 years, 2 months ago
  71. c81665c Change AndroidNetworkMonitor::NotifyConnectionTypeChanged to use Invoke by Niels Möller · 3 years, 2 months ago
  72. 82a9412 Reland "Add a fuzzer test that tries to connect a PeerConnection." by Harald Alvestrand · 3 years, 2 months ago
  73. be66d95 srtp: document rationale for srtp overhead calculation by Philipp Hancke · 3 years, 2 months ago
  74. 456a264 Update WebRTC code version (2021-03-08T04:01:44). by webrtc-version-updater · 3 years, 2 months ago
  75. aedb0f0 Update WebRTC code version (2021-03-07T04:02:41). by webrtc-version-updater · 3 years, 2 months ago
  76. eb449a9 Revert "Reland "Enable quality scaling when allowed"" by Guido Urdaneta · 3 years, 2 months ago
  77. 77f97ec Update WebRTC code version (2021-03-06T04:02:56). by webrtc-version-updater · 3 years, 2 months ago
  78. 964a88f Prevent possible out-of-bounds access by Sergey Silkin · 3 years, 2 months ago
  79. 4a4273b VP9 ResolutionBitrateLimits: If bitrates are configured, use intersection. by Åsa Persson · 3 years, 2 months ago
  80. a86b29b Add VP9-specific default resolution bitrate limits by Sergey Silkin · 3 years, 2 months ago
  81. 1a89bc8 build: improve rtc_include_tests documentation by Philipp Hancke · 3 years, 2 months ago
  82. 1413e2d Update WebRTC code version (2021-03-05T04:03:29). by webrtc-version-updater · 3 years, 2 months ago
  83. 8a38b1c Revert "Fix race between destroying SctpTransport and receiving notification on timer thread." by Taylor · 3 years, 2 months ago
  84. 83be84b Reland "Enable quality scaling when allowed" by Sergey Silkin · 3 years, 2 months ago
  85. d140c8f Added missing nullable annotations to iOS SDK. by Yura Yaroshevich · 3 years, 2 months ago
  86. 854d59f Temporarily disable remaining Opus bit exactness tests. by Jakob Ivarsson · 3 years, 2 months ago
  87. 3208bf1 Allow uninstantiated parametrized tests in data_channel_integration_tests.cc by Bjorn Terelius · 3 years, 2 months ago
  88. 609b524 Revert "Enable quality scaling when allowed" by Björn Terelius · 3 years, 2 months ago
  89. ae44fde Revert "Add a fuzzer test that tries to connect a PeerConnection." by Evan Shrubsole · 3 years, 2 months ago
  90. d744c5a Update WebRTC code version (2021-03-04T04:02:19). by webrtc-version-updater · 3 years, 2 months ago
  91. 8cfb287 Add AV1 encoder&decoder wrappers for iOS SDK. by Yura Yaroshevich · 3 years, 2 months ago
  92. 662d4c3 AV1 test: change ssim threshold by Jerome Jiang · 3 years, 2 months ago
  93. 83e6ece Comment out uninstantiated parametrized PC full stack test by Bjorn Terelius · 3 years, 2 months ago
  94. e7a5581 Temporarily disable some Opus bit exactness tests. by Jakob Ivarsson · 3 years, 2 months ago
  95. b6b782d Fix potential unsafe access to VCMTimestampMap::data by Johannes Kron · 3 years, 2 months ago
  96. 752cbab Enable quality scaling when allowed by Sergey Silkin · 3 years, 2 months ago
  97. db0b4a8 Do not crash if codec is not available by Sergey Silkin · 3 years, 2 months ago
  98. 652ada5 Enabling a safe fall-back functionality for overruns in the runtime settings by Per Åhgren · 3 years, 2 months ago
  99. 99bcf60 Check MID for illegal token characters. by Harald Alvestrand · 3 years, 2 months ago
  100. c67b77e Add a fuzzer test that tries to connect a PeerConnection. by Harald Alvestrand · 3 years, 2 months ago