1. 73cf80a Fixes incorrect feedback to EncoderBitrateAdjuster [perf note] by Erik Språng · 4 years ago
  2. ef036cd [Stats] Cleanup obsolete stats - isRemote & deleted by Di Wu · 4 years ago
  3. 56db9ff VideoStreamEncoder: Don't map kNative video frame buffers. by Henrik Boström · 4 years ago
  4. 5cf8c2c Fix unspecified time origin for `lastPacketReceivedTimestamp` by Alessio Bazzica · 4 years ago
  5. 9054aa8 Update WebRTC code version (2021-03-24T04:02:05). by webrtc-version-updater · 4 years ago
  6. f7b1b95 Add `RTCRemoteOutboundRtpStreamStats` for audio streams by Alessio Bazzica · 4 years ago
  7. 26abdaf AV1: Use Default TX type for encoding by Fyodor Kyslov · 4 years ago
  8. 2f71b61 Make sure "remote-inbound-rtp.jitter" and "packetsLost" is exposed to JS by Henrik Boström · 4 years ago
  9. ca18809 Move RtpFrameObject and EncodedFrame out of video_coding namespace. by philipel · 4 years ago
  10. 93ee168 Allow empty video layer allocation extension by Jonas Oreland · 4 years ago
  11. fa4db49 Make GL errors thrown by checkNoGLES2Error inherit GLException. by Sami Kalliomäki · 4 years ago
  12. 7cbe887 Change default adaptive ptime min bitrate to 16kbps. by Jakob Ivarsson · 4 years ago
  13. f0adf38 Fix timestamps for the remote outbound audio stream stats by Alessio Bazzica · 4 years ago
  14. 3889de1 Support native scaling of VideoFrameBuffers in LibvpxVp8Encoder. by Henrik Boström · 4 years ago
  15. 6a67150 Move RtpFrameReferenceFinder out of video_coding namespace. by philipel · 4 years ago
  16. 2ba32f3 Delete AsyncInvoker usage in TurnServer by Niels Möller · 4 years ago
  17. a9ba450 stats: add address as alias for ip by Philipp Hancke · 4 years ago
  18. e2ac591 Update WebRTC code version (2021-03-23T04:03:37). by webrtc-version-updater · 4 years ago
  19. 1cdbabd Update WgcCaptureSession to handle portrait oriented screen capture. by Austin Orion · 4 years ago
  20. e0059dc Roll chromium_revision b7f0a0c111..c0436807ae (865116:865247) by chromium-webrtc-autoroll · 4 years ago
  21. c366d51 Fix unit for inbound RTP stat `lastPacketReceivedTimestamp` (s -> ms) by Alessio Bazzica · 4 years ago
  22. c303f82 Add new owners for sdk/android. by Sami Kalliomäki · 4 years ago
  23. 464bcd4 Revert "Reland "[Battery]: Delay start of TaskQueuePacedSender."" by Etienne Pierre-Doray · 4 years ago
  24. efad89c Roll chromium_revision 23141e38f1..b7f0a0c111 (864556:865116) by chromium-webrtc-autoroll · 4 years ago
  25. bd9e4a9 Support native scaling of VideoFrameBuffers in LibvpxVp9Encoder. by Henrik Boström · 4 years ago
  26. 2ff25db Update apply-iwyu tool to report compile errors by Harald Alvestrand · 4 years ago
  27. ffb7603 Delete TurnPort usage of AsyncInvoker by Niels Möller · 4 years ago
  28. eb28298 Update rsid and mid spec links from draft to release version by Danil Chapovalov · 4 years ago
  29. 47350c2 Reland "Triggering CI." by Mirko Bonadei · 4 years ago
  30. f412976 Provide a default implementation of NV12BufferInterface::CropAndScale. by Henrik Boström · 4 years ago
  31. 50d79ba Revert "Triggering CI." by Mirko Bonadei · 4 years ago
  32. 0a104c4 Delete obsolete method EncodedImage::Retain() by Niels Möller · 4 years ago
  33. 6f7e205 Delete AsyncInvoker usage from StunProber by Niels Möller · 4 years ago
  34. ebd2010 Check if ifa_addr field is null. by Björn Terelius · 4 years ago
  35. 76b51e2 Improve thread annotations for TurnServer by Niels Möller · 4 years ago
  36. 2f5f5fa standalone ice transport: dont use component 0 by Philipp Hancke · 4 years ago
  37. c732576 Triggering CI. by Mirko Bonadei · 4 years ago
  38. 8bf1cd1 Rename (packets|bytes)_dropped to (packets|bytes)_discarded_no_receiver by Artem Titov · 4 years ago
  39. eecc4f5 Fix: when SamplesStatsCounter is empty it's not propagated to the Histogram perf output by Artem Titov · 4 years ago
  40. 2bab0ef Update WebRTC code version (2021-03-20T04:03:21). by webrtc-version-updater · 4 years ago
  41. 490c150 Delete unowned buffer in EncodedImage. by Niels Möller · 4 years ago
  42. 2b250734 Delete FakeIceTransport usage of AsyncInvoker by Niels Möller · 4 years ago
  43. ef7d61e Update WebRTC code version (2021-03-19T04:04:06). by webrtc-version-updater · 4 years ago
  44. ba3e6c2 Roll chromium_revision 5f1d8e0c95..23141e38f1 (864439:864556) by chromium-webrtc-autoroll · 4 years ago
  45. 18b0947 Roll chromium_revision 6bb9b62b86..5f1d8e0c95 (864335:864439) by chromium-webrtc-autoroll · 4 years ago
  46. 3328761 Roll chromium_revision 74fb21b370..6bb9b62b86 (864218:864335) by chromium-webrtc-autoroll · 4 years ago
  47. d6d2a29 Roll chromium_revision a1e978b5ab..74fb21b370 (864105:864218) by chromium-webrtc-autoroll · 4 years ago
  48. c780605 Make num_encoded_channels_ atomic by Gustaf Ullberg · 4 years ago
  49. 049e611 Add missing EXPECT_CALL for `RTCStatsCollectorTest` tests by Alessio Bazzica · 4 years ago
  50. 0848994 Replace AsyncInvoker with PostDelayedTask, in DtmfSender by Niels Möller · 4 years ago
  51. 9243088 Add thread annotations to FakeIceTransport by Niels Möller · 4 years ago
  52. 0357477 Delete use of AsyncInvoker from FakePacketTransport by Niels Möller · 4 years ago
  53. cbadb8b Expose offerExtmapAllowMixed in iOS SDK. by Yura Yaroshevich · 4 years ago
  54. 426d679 Update WebRTC code version (2021-03-18T04:03:50). by webrtc-version-updater · 4 years ago
  55. b9a6c03 Roll chromium_revision 345f2fb2f5..a1e978b5ab (863976:864105) by chromium-webrtc-autoroll · 4 years ago
  56. 5a23b3d Roll chromium_revision ca2293bd17..345f2fb2f5 (863854:863976) by chromium-webrtc-autoroll · 4 years ago
  57. f8776cb Revert "AV1: Use Default TX type for encoding" by Fyodor Kyslov · 4 years ago
  58. db01a82 Roll chromium_revision 5371070da9..ca2293bd17 (863745:863854) by chromium-webrtc-autoroll · 4 years ago
  59. b0dc518 AV1: Use Default TX type for encoding by Fyodor Kyslov · 4 years ago
  60. dd4d5e3 Reland "[Battery]: Delay start of TaskQueuePacedSender." by Etienne Pierre-doray · 4 years ago
  61. 32af25b Disable more flaky PeerConnectionIntegrationTests on Windows by Rasmus Brandt · 4 years ago
  62. cf93670 sctp: Finish sending partial messages before sending stream reset events by Florent Castelli · 4 years ago
  63. 92a768a Roll chromium_revision 47b94319df..5371070da9 (863625:863745) by chromium-webrtc-autoroll · 4 years ago
  64. 4173614 Expose enableImplicitRollback in iOS SDK. by Yura Yaroshevich · 4 years ago
  65. 1827483 Update WebRTC code version (2021-03-17T04:02:27). by webrtc-version-updater · 4 years ago
  66. 2ba6435 Roll chromium_revision 7f72620d8b..47b94319df (863490:863625) by chromium-webrtc-autoroll · 4 years ago
  67. 861a0d1 Roll chromium_revision 67d2a9f799..7f72620d8b (863354:863490) by chromium-webrtc-autoroll · 4 years ago
  68. f19aec8 Updates ulpfec reader to accept padding on media packets. by Erik Språng · 4 years ago
  69. 87dbe9a Roll chromium_revision 432f33c810..67d2a9f799 (863160:863354) by chromium-webrtc-autoroll · 4 years ago
  70. d19e3b9 Reland "Reland "Enable quality scaling when allowed"" by Sergey Silkin · 4 years ago
  71. e37fa19 Delete unused class DummyDtmfObserver by Niels Möller · 4 years ago
  72. ab63350 Delete RtpRtcp::RemoteRTCPStat in favor of GetLatestReportBlockData by Danil Chapovalov · 4 years ago
  73. 19775cb Reland "Reduce complexity in the APM pipeline when the output is not used" by Per Åhgren · 4 years ago
  74. 15179a9 Allowing reduced computations in the noise suppressor when the output is not used by Per Åhgren · 4 years ago
  75. 8ee1ec8 Allowing reduced computations in the AEC3 when the output is not used by Per Åhgren · 4 years ago
  76. 3e774f6 Make AndroidNetworkMonitor::Start() create a new task safety flag by Niels Möller · 4 years ago
  77. a776f51 Avoid two consecutive version updates. by Mirko Bonadei · 4 years ago
  78. 9d1e070 Increase wait-for-lost-packet from 10 to 100 msec in MTU test by Harald Alvestrand · 4 years ago
  79. 596ba4c Roll chromium_revision 0b0b620d02..432f33c810 (863050:863160) by chromium-webrtc-autoroll · 4 years ago
  80. 785e23b Drop # of video tracks in renegotiate-many-videos to 8 by Harald Alvestrand · 4 years ago
  81. 0855302 Update WebRTC code version (2021-03-16T04:03:07). by webrtc-version-updater · 4 years ago
  82. f172706 Roll chromium_revision d935055b21..0b0b620d02 (862883:863050) by chromium-webrtc-autoroll · 4 years ago
  83. bff6489 AV1: Disable several intra coding tools. by Fyodor Kyslov · 4 years ago
  84. 995c5c8 Roll chromium_revision e4fd023c85..d935055b21 (862756:862883) by chromium-webrtc-autoroll · 4 years ago
  85. db5d728 Add refined handling of the internal scaling of the audio in APM by Per Åhgren · 4 years ago
  86. b315951 Remove incorrect DCHECKs from LibaomAv1Encoder::SetRates. by philipel · 4 years ago
  87. fdd6099 Rework transient suppressor configuration in audioproc_f by Gustaf Ullberg · 4 years ago
  88. 685be14 Disable flaky AddMediaToConnectedBundleDoesNotRestartIce on tsan by Rasmus Brandt · 4 years ago
  89. e657d87 Allow port 53 as a TURN port. by Harald Alvestrand · 4 years ago
  90. c88bdad Roll chromium_revision c3fb27225e..e4fd023c85 (861941:862756) by chromium-webrtc-autoroll · 4 years ago
  91. 6ca955a Reland "Fix problem with ipv4 over ipv6 on Android" by Jonas Oreland · 4 years ago
  92. 7087b83 Test that SCTP succeeds with one MTU and fails with a lower MTU by Harald Alvestrand · 4 years ago
  93. 0e42cf7 Reland "Parse encoded frame QP if not provided by encoder" by Sergey Silkin · 4 years ago
  94. b6bc357 turn: add logging for long usernames by Philipp Hancke · 4 years ago
  95. 6097b0f Delete use of AsyncInvoker from PeerConnectionIntegrationWrapper by Niels Möller · 4 years ago
  96. 13118a7 Update WebRTC code version (2021-03-15T04:05:00). by webrtc-version-updater · 4 years ago
  97. 55bc077 Add one frame (10 ms) of silence in APM output after unmuting by Per Åhgren · 4 years ago
  98. 1e60490 Revert "Fix problem with ipv4 over ipv6 on Android" by Taylor Brandstetter · 4 years ago
  99. bc1c93d Add remote-outbound stats for audio streams by Alessio Bazzica · 4 years ago
  100. c80f955 Avoid log spam when decoder implementation changes by Erik Språng · 4 years ago