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  1. ef53a7f Reset IO thread checker when iOS audio unit stops by Brian Dai · 3 years, 3 months ago
  2. bf95da8 Update WebRTC code version (2021-01-23T04:04:31). by webrtc-version-updater · 3 years, 3 months ago
  3. 8c007ff Restrict usage of resolution bitrate limits to singlecast by Sergey Silkin · 3 years, 3 months ago
  4. 461b1d9 Restart CPU overuse detection when encoder settings has changed. by Jakob Ivarsson · 3 years, 3 months ago
  5. 2803a2d Make audio device mocks publicly visible by Steve Anton · 3 years, 3 months ago
  6. 8df643b Introduce FinalRefCountedObject template class by Danil Chapovalov · 3 years, 3 months ago
  7. cbacec5 Monitor the "concealed samples" stat for the audio during negotiation. by Harald Alvestrand · 3 years, 4 months ago
  8. 11215fe Require scalability mode to initialize av1 encoder. by Danil Chapovalov · 3 years, 3 months ago
  9. 2ed56fe Update WebRTC code version (2021-01-22T04:03:26). by webrtc-version-updater · 3 years, 3 months ago
  10. d2dd732 Introduce network emulated endpoint optional name for better logging by Artem Titov · 3 years, 3 months ago
  11. e4fd1ba Delete mutable rtc::CopyOnWriteBuffer::data by Danil Chapovalov · 3 years, 3 months ago
  12. 6031b74 Implement a Neon optimized function to find the argmax element in an array. by Ivo Creusen · 3 years, 3 months ago
  13. 03eed7c Fixes issue triggered by WebRTC-VP9-PerformanceFlags trial. by Erik Språng · 3 years, 3 months ago
  14. a7e34d3 Add resolution_bitrate_limits to EncoderInfo field trial. by Åsa Persson · 3 years, 3 months ago
  15. 026ad9a Update WebRTC code version (2021-01-21T04:03:14). by webrtc-version-updater · 3 years, 3 months ago
  16. 49b20f9 Fix race with SctpTransport destruction and usrsctp timer thread. by Taylor Brandstetter · 3 years, 3 months ago
  17. c2ae4c8 Allow separate dump sets for the data dumper in APM by Per Åhgren · 3 years, 3 months ago
  18. 0be1846 Fix enabling DependencyDescriptor for VP9 with spatial layers by Danil Chapovalov · 3 years, 3 months ago
  19. 1657baf Add `.cache` to .gitignore. by Rasmus Brandt · 3 years, 3 months ago
  20. 4f281f1 Cleanup FakeRtcEventLog from thread awareness by Danil Chapovalov · 3 years, 3 months ago
  21. 812c73c Another ilbc cross correlation fix by Ivo Creusen · 3 years, 3 months ago
  22. 5eb43b4 Prefix HAVE_SCTP macro with WEBRTC_. by Mirko Bonadei · 3 years, 3 months ago
  23. 6dcbcea Update WebRTC code version (2021-01-20T04:04:26). by webrtc-version-updater · 3 years, 3 months ago
  24. 33c0ab4 Call MediaChannel::OnPacketReceived on the network thread. by Tomas Gunnarsson · 3 years, 3 months ago
  25. 1cbf21e ChannelStatistics RTT test case around remote SSRC change. by Tim Na · 3 years, 3 months ago
  26. a24d35e AlignmentAdjuster: take reduced layers into account for default downscaling. by Åsa Persson · 3 years, 3 months ago
  27. 5c3ff6b Switch to enable the HMM transparent mode classifier by Gustaf Ullberg · 3 years, 3 months ago
  28. 801c999 Signal extmap-allow-mixed by default on session level by Emil Lundmark · 3 years, 3 months ago
  29. 1e75df2 Remove lock from UlpfecReceiverImpl and replace with a sequence checker. by Tomas Gunnarsson · 3 years, 3 months ago
  30. 5eb527c Replace sigslot usages with callback list library. by Lahiru Ginnaliya Gamathige · 3 years, 3 months ago
  31. a722d2a Add DeliverPacketAsync method to PacketReceiver. by Tomas Gunnarsson · 3 years, 3 months ago
  32. 6cdb67f Document expected thread safety of the TaskQueue interface by Danil Chapovalov · 3 years, 3 months ago
  33. 676d61f Update WebRTC code version (2021-01-19T04:01:32). by webrtc-version-updater · 3 years, 3 months ago
  34. 29bd863 Add field trial for allowing cropped resolution when limiting max layers. by Åsa Persson · 3 years, 3 months ago
  35. 5cf0ef0 Stricter compile-time thread annotations in JsepTransportController by Niels Möller · 3 years, 3 months ago
  36. 25b8235 Remove unused function VideoDecoder::PrefersLateDecoding. by philipel · 3 years, 4 months ago
  37. 42eef86 Remove unused code in APM by Alessio Bazzica · 3 years, 3 months ago
  38. 111a371 Delete unused.h include from api as unused by Danil Chapovalov · 3 years, 3 months ago
  39. 3e9cb2c Move deprecated code to their own build targets. by Niels Möller · 3 years, 4 months ago
  40. 844c759 fix variable naming in ReportSdpFormatReceived by Philipp Hancke · 3 years, 3 months ago
  41. ee95c1c Roll chromium_revision da24822732..18311e2720 (844357:844473) by chromium-webrtc-autoroll · 3 years, 3 months ago
  42. 77ceff9 payload type mapper: use media constants by Philipp Hancke · 3 years, 3 months ago
  43. 4bab23f Update pc/ to use C++ lambdas instead of rtc::Bind by Niels Möller · 3 years, 3 months ago
  44. d51000f Delete RTC_WARN_UNUSED_RESULT as no longer used by Danil Chapovalov · 3 years, 3 months ago
  45. cc6ae44 Reland "Improve structuring of test for audio glitches." by Harald Alvestrand · 3 years, 3 months ago
  46. c20e333 Update SCTP test to use C++ lambdas instead of rtc::Bind by Niels Möller · 3 years, 4 months ago
  47. 588526c Remove deprecated //rtc_base:async_resolver. by Mirko Bonadei · 3 years, 3 months ago
  48. e091fd2 Remove lock from RtpStreamReceiverController. by Tomas Gunnarsson · 3 years, 3 months ago
  49. 8467cf2 Reduce redundant flags for audio stream playout state. by Tomas Gunnarsson · 3 years, 3 months ago
  50. 12971a2 Update WebRTC code version (2021-01-18T04:03:42). by webrtc-version-updater · 3 years, 3 months ago
  51. 09729d2 Update WebRTC code version (2021-01-17T04:01:53). by webrtc-version-updater · 3 years, 3 months ago
  52. 9c8dd87 Fixing WebRTC/Chromium FYI build. by Mirko Bonadei · 3 years, 3 months ago
  53. be93b78 Move iOS bundle data for tests inside rtc_include_test (take 2). by Mirko Bonadei · 3 years, 3 months ago
  54. 2397b6e SimulcastEncoderAdapter: Add field trial for EncoderInfo settings. by Åsa Persson · 3 years, 3 months ago
  55. fd9500e In criket::BaseChannel replace AsyncInvoker with task queue functions by Danil Chapovalov · 3 years, 3 months ago
  56. 8ed6185 Move iOS bundle data for tests inside rtc_include_test. by Mirko Bonadei · 3 years, 3 months ago
  57. d705bbe Roll chromium_revision 3c2d1e3ba1..da24822732 (844134:844357) by chromium-webrtc-autoroll · 3 years, 3 months ago
  58. f4d1ecf Update WebRTC code version (2021-01-16T04:03:13). by webrtc-version-updater · 3 years, 3 months ago
  59. 61ede7a Roll chromium_revision 72fe6d7aab..3c2d1e3ba1 (844008:844134) by chromium-webrtc-autoroll · 3 years, 3 months ago
  60. d723da1 Reland "Default enable delay adaptation during DTX." by Jakob Ivarsson · 3 years, 3 months ago
  61. 098da17 Reland "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code" by Danil Chapovalov · 3 years, 4 months ago
  62. ba91dbc In SVC controllers add support for frames dropped by encoder by Danil Chapovalov · 3 years, 3 months ago
  63. e5f4c6b Reland "Refactor rtc_base build targets." by Mirko Bonadei · 3 years, 3 months ago
  64. 79d9c37 Revert "Default enable delay adaptation during DTX." by Jakob Ivarsson · 3 years, 3 months ago
  65. 59bdcbe Default enable delay adaptation during DTX. by Jakob Ivarsson · 3 years, 3 months ago
  66. 5ab6a8c Refactors SimulcastEncoder Adapter. by Erik Språng · 3 years, 4 months ago
  67. 1528e2b Set AV1E_SET_ERROR_RESILIENT_MODE on T1 and T2 enhanced layers by Sergio Garcia Murillo · 3 years, 4 months ago
  68. 809a261 Roll chromium_revision 42ab9dc8c8..72fe6d7aab (843550:844008) by chromium-webrtc-autoroll · 3 years, 3 months ago
  69. a86cef7 Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding by Danil Chapovalov · 3 years, 4 months ago
  70. 87e9f6e Update p2p/ to use C++ lambdas instead of rtc::Bind by Niels Möller · 3 years, 4 months ago
  71. 3ae09f5 Revert "Improve structuring of test for audio glitches." by Alex Loiko · 3 years, 3 months ago
  72. 98db5d1 Revert "Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll" by Alex Loiko · 3 years, 3 months ago
  73. b45d3aa Update android jni code to use C++ lambdas instead of rtc::Bind by Niels Möller · 3 years, 4 months ago
  74. d76dcbd Simplify FakeRtcEventLog, delete rtc::Bind usage by Niels Möller · 3 years, 4 months ago
  75. f9ee0e0 Add cross trafic emulation api by Andrey Logvin · 3 years, 4 months ago
  76. a7ccacc Update WebRTC code version (2021-01-15T04:04:08). by webrtc-version-updater · 3 years, 4 months ago
  77. 6c8738c mb: Fully remove references to 'masters' in favor of 'builder_groups'. by Mirko Bonadei · 3 years, 4 months ago
  78. 7acc2d9 Revert "Refactor rtc_base build targets." by Mirko Bonadei · 3 years, 4 months ago
  79. 884118d Delete unused functions in ModuleRtpRtcpImpl by Danil Chapovalov · 3 years, 4 months ago
  80. 23f60eb Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll by Austin Orion · 3 years, 4 months ago
  81. c17bca7 SetOfferedRtpHeaderExtensions: fix error code. by Markus Handell · 3 years, 4 months ago
  82. 1669103 Roll chromium_revision 189823ba75..42ab9dc8c8 (843026:843550) by chromium-webrtc-autoroll · 3 years, 4 months ago
  83. f77aa81 Update AudioDeviceBuffer to use C++ lambdas instead of rtc::Bind by Niels Möller · 3 years, 4 months ago
  84. 42c0d70 Include packetization in video codec string by Emil Lundmark · 3 years, 4 months ago
  85. 4319b16 Revert "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code" by Danil Chapovalov · 3 years, 4 months ago
  86. 8c2250e Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code by Danil Chapovalov · 3 years, 4 months ago
  87. 1921708 SetNegotiatedHeaderExtensions_w: Set list synchronously. by Markus Handell · 3 years, 4 months ago
  88. c12f625 Adds VideoDecoder::GetDecoderInfo() by Erik Språng · 3 years, 4 months ago
  89. a0bb2ef Delete unused VideoType enum values by Niels Möller · 3 years, 4 months ago
  90. e61a40e Fix typo in audio processing header. by Hua, Chunbo · 3 years, 4 months ago
  91. fdbaeda Improve structuring of test for audio glitches. by Harald Alvestrand · 3 years, 4 months ago
  92. dbcaff0 Fix AudioProcessing::Config::ToString() implementation. by Hua, Chunbo · 3 years, 4 months ago
  93. d73426d6 Add new empty build targets rtp_rtcp_legacy and video_legacy. by Niels Möller · 3 years, 4 months ago
  94. b24e720 Fix inconsistencies in network BUILD.gn file by Andrey Logvin · 3 years, 4 months ago
  95. db79204 Change PeerConnectionE2EQualityTest to use lambdas instead of rtc::Bind by Niels Möller · 3 years, 4 months ago
  96. 76714a6 AGC2 minor code clean up by Alessio Bazzica · 3 years, 4 months ago
  97. ece6712 Add av1 to lower range IDs. by Jerome Jiang · 3 years, 4 months ago
  98. 507eacf Reland "ChannelStatistics used for RTP stats in VoipStatistics." by Tim Na · 3 years, 4 months ago
  99. 2297272 Roll chromium_revision c0feffab8f..189823ba75 (842900:843026) by chromium-webrtc-autoroll · 3 years, 4 months ago
  100. 8606b9c Replace all uses of the word 'master' with 'builder_group' in //tools/mb by Mirko Bonadei · 3 years, 4 months ago