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webrtc / src / dcac9fe3d1646465e6dbc2c6bf1d465e79ba5414
  1. dcac9fe Add may_contain_cursor property to DesktopFrame to avoid double capture by Austin Orion · 3 years, 11 months ago
  2. 688235d Exclude WS_EX_TOOLWINDOWs for WgcCapturerWin. by Austin Orion · 3 years, 11 months ago
  3. 516e284 Remove DataChannelType and deprecated option disable_sctp_data_channels by Florent Castelli · 3 years, 11 months ago
  4. eb9c3f2 Handle OnPacketSent on the network thread via MediaChannel. by Tomas Gunnarsson · 3 years, 11 months ago
  5. edb7ea2 Refactors Vp9UncompressedHeaderParser. by Erik Språng · 3 years, 11 months ago
  6. bfd9ba8 Fix unsafe variable access in RTCStatsCollector by Tomas Gunnarsson · 3 years, 11 months ago
  7. f703ed1 Ban std::shared_ptr in style guide by Harald Alvestrand · 3 years, 11 months ago
  8. 25e7352 Add support for setting the initial state to the pending task flag. by Tomas Gunnarsson · 3 years, 11 months ago
  9. e984aa2 Add thread accessors to Call. by Tomas Gunnarsson · 3 years, 11 months ago
  10. bddebc8 Fix an example in SequenceChecker documentation by Harald Alvestrand · 3 years, 11 months ago
  11. b849311 Update last received keyframe packet timestamp on all packets with the same RTP timestamp. by philipel · 3 years, 11 months ago
  12. 0d3c09a webrtc::Mutex: Introduce mutex_race_check.h. by Markus Handell · 3 years, 11 months ago
  13. d29c689 Expose adaptive_ptime from Android SDK. by Yura Yaroshevich · 3 years, 11 months ago
  14. d71b38e Update WebRTC code version (2021-04-19T04:03:03). by webrtc-version-updater · 3 years, 11 months ago
  15. d46a174 Expose adaptive_ptime from iOS SDK. by Yura Yaroshevich · 3 years, 11 months ago
  16. 7fa8d46 Slight code clarification in RemoveStoppedTransceivers. by Tomas Gunnarsson · 3 years, 11 months ago
  17. 0ee5bcf Update WebRTC code version (2021-04-18T04:03:49). by webrtc-version-updater · 4 years ago
  18. e632402 Remove rtp data channel related code from media_channel.* by Tomas Gunnarsson · 4 years ago
  19. 18ac30c Update WebRTC code version (2021-04-17T04:04:03). by webrtc-version-updater · 4 years ago
  20. 983b620 Remove third_party/xstream from DEPS by Bjorn Terelius · 4 years ago
  21. 78aa5cd dcsctp: Ensure packet size doesn't exceed MTU by Victor Boivie · 4 years ago
  22. 7af57c6 Remove RTP data implementation by Harald Alvestrand · 4 years ago
  23. f981cb3 Add video/g3doc/stats.md to the doc site menu by Artem Titov · 4 years ago
  24. 15e078c Fix unsignalled ssrc race in WebRtcVideoChannel. by Henrik Boström · 4 years ago
  25. 882d007 Add documentation for video/stats. by Åsa Persson · 4 years ago
  26. 0131a4d Delete StreamAdapterInterface by Niels Möller · 4 years ago
  27. b291da8 Add conceptual docs for modules/video_coding by Rasmus Brandt · 4 years ago
  28. dd36198 Revert "Expose AV1 encoder&decoder from Android SDK." by Björn Terelius · 4 years ago
  29. 220a252 Delete unused class MessageBufferReader by Niels Möller · 4 years ago
  30. 6c127a1 Add Stable Writable Connection Ping Interval parameter to RTCConfiguration. by Derek Bailey · 4 years ago
  31. 74b1bbe Remove unused a gn variable related to gtk by Byoungchan Lee · 4 years ago
  32. a43528c Update WebRTC code version (2021-04-16T04:04:52). by webrtc-version-updater · 4 years ago
  33. 3ceb16e [Android] Set use_raw_android_executable explicitly for test() template. by Peter Kotwicz · 4 years ago
  34. 0f57e0b Make libjingle_peerconnection_metrics_default_jni available in Linux builds. by Mirko Bonadei · 4 years ago
  35. 9fea310 Fix crash in WindowCapturerWinGdi::CaptureFrame. by Austin Orion · 4 years ago
  36. a80c3e5 sctp: Reorganize build targets by Florent Castelli · 4 years ago
  37. 6c7c495 doc: fix ice metadata + spelling by Philipp Hancke · 4 years ago
  38. fedd502 Expose AV1 encoder&decoder from Android SDK. by Yura Yaroshevich · 4 years ago
  39. 572f50f Delete left-over references to AsyncInvoker by Niels Möller · 4 years ago
  40. affd219 Delete AsyncInvoker usage from SimulatedPacketTransport by Niels Möller · 4 years ago
  41. bc959b6 Remove enable_rtp_data_channel by Harald Alvestrand · 4 years ago
  42. fa8a946 Remove obsolete DCHECK in remote_audio_source.cc. by Henrik Boström · 4 years ago
  43. 17490b5 Fix regression in UsrSctpReliabilityTest by Niels Möller · 4 years ago
  44. 403e328 Fix build with rtc_libvpx_build_vp9=false by Byoungchan Lee · 4 years ago
  45. 980c460 AGC2: retuning and large refactoring by Alessio Bazzica · 4 years ago
  46. d28434b Configure GN to use python3 to exec_script. by Mirko Bonadei · 4 years ago
  47. dad500a Remove PacketBuffers internal mutex. by philipel · 4 years ago
  48. 61982a7 AGC2 lightweight noise floor estimator by Alessio Bazzica · 4 years ago
  49. 3ab7a55 Reformat pacer doc and add it into sitemap by Artem Titov · 4 years ago
  50. 9aec8c2 Use default rtp parameters to init wrappers in iOS by Yura Yaroshevich · 4 years ago
  51. 89f3dd5 Make RTC_LOG_THREAD_BLOCK_COUNT less spammy for known call counts by Tomas Gunnarsson · 4 years ago
  52. 5744b7f Fix formatting in sitemap.md by Artem Titov · 4 years ago
  53. 08d30a2 Add documentation for video/adaptation by Evan Shrubsole · 4 years ago
  54. 24bc419 Revert "Fix RTP header extension encryption" by Björn Terelius · 4 years ago
  55. dea5721 Adding g3doc for AudioProcessingModule (APM) by Per Åhgren · 4 years ago
  56. 9861f96 dcsctp: Add operators on TimeMs and DurationMs by Victor Boivie · 4 years ago
  57. 8181b4f Add conceptual documentation for NetEq. by Jakob Ivarsson · 4 years ago
  58. a743303 Fix RTP header extension encryption by Lennart Grahl · 4 years ago
  59. 84ba164 Change from sakal@webrtc.org to xalep@webrtc.org in OWNERS files. by Mirko Bonadei · 4 years ago
  60. c54f672 dcsctp: Fix post-review comments for DataTracker by Victor Boivie · 4 years ago
  61. 0498519 Add g3doc for audio coding module. by Minyue Li · 4 years ago
  62. 1fad94f Remove ErleUncertainty by Gustaf Ullberg · 4 years ago
  63. 77d73a6 Document SctpTransport by Harald Alvestrand · 4 years ago
  64. 1d2d169 Update WebRTC code version (2021-04-14T04:04:15). by webrtc-version-updater · 4 years ago
  65. e871e02 Add telemetry to measure usage, perf, and errors in Desktop Capturers. by Austin Orion · 4 years ago
  66. efcfa4b Roll chromium_revision 0bde1c5411..1a13f11499 (871876:872016) by chromium-webrtc-autoroll · 4 years ago
  67. 250fbb3 dcsctp: Make Sequence Number API more consistent by Victor Boivie · 4 years ago
  68. ce423ce Track last packet receive times in RtpVideoStreamReceiver instead of the PacketBuffer. by philipel · 4 years ago
  69. cd83ae2 Speed up FrameCombiner::Combine by 3x by Steve Anton · 4 years ago
  70. 32347b5 Add readme for pacing module by Erik Språng · 4 years ago
  71. 09c7f1e Add architecture section about PeerConnection test framework by Artem Titov · 4 years ago
  72. 79cbe69 Removes incorrect test expectation. by Erik Språng · 4 years ago
  73. 3db3a06 Adding g3doc for AudioDeviceModule (ADM) - part of the AudioEngine by henrika · 4 years ago
  74. df1edc9 API description: PeerConnection description by Harald Alvestrand · 4 years ago
  75. 11b3089 Roll chromium_revision 74f869d04b..0bde1c5411 (871745:871876) by chromium-webrtc-autoroll · 4 years ago
  76. 1fded2f dcsctp: Fix build dependencies by Florent Castelli · 4 years ago
  77. e082984 Add death test for WrappingAsyncResolver by Harald Alvestrand · 4 years ago
  78. a168bb9 Add index.md documentation page for PC level test framework by Artem Titov · 4 years ago
  79. 696cea0 Refactor some RtpSender-level tests into RtpRtcp-level tests by Erik Språng · 4 years ago
  80. 5fe0b37 Roll chromium_revision 7e70585ca5..74f869d04b (871605:871745) by chromium-webrtc-autoroll · 4 years ago
  81. c8cf0a6 Remove MDNS message implementation by Harald Alvestrand · 4 years ago
  82. eff79cf Roll chromium_revision 2dffe06711..7e70585ca5 (871492:871605) by chromium-webrtc-autoroll · 4 years ago
  83. 067dce7 Fix processing of dropped frame for runtime added participant by Andrey Logvin · 4 years ago
  84. dc53ce6 Revert "Add addr in error msg if stun sock sent with error" by Mirko Bonadei · 4 years ago
  85. 8093935 Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492) by Mirko Bonadei · 4 years ago
  86. 5051693 [Battery]: TaskQueuePacedSender not started by default. by Etienne Pierre-doray · 4 years ago
  87. 9ff75a6 Add addr in error msg if stun sock sent with error by Yura Yaroshevich · 4 years ago
  88. 3928e8f dcsctp: Disable packet fuzzers by Victor Boivie · 4 years ago
  89. 0aa1a19 Add module overview of ICE by Jonas Oreland · 4 years ago
  90. 9de39f6 Add titovartem@webrtc.org as owner for /g3doc by Artem Titov · 4 years ago
  91. 4af6f2b Move threading documentation for API into g3doc structure by Harald Alvestrand · 4 years ago
  92. a3575cb Remove tautological 'unsigned expr < 0' comparisons by Anton Bikineev · 4 years ago
  93. 22379fc sctp: Rename SctpTransport to UsrSctpTransport by Florent Castelli · 4 years ago
  94. 606bd6d dcsctp: Use correct field width for PPID by Victor Boivie · 4 years ago
  95. 9d60936 dcsctp: Fix relative dependency paths in timer/ by Victor Boivie · 4 years ago
  96. 1003219 srtp: compare key length to srtp policy key length by Philipp Hancke · 4 years ago
  97. 5691053 IceStatesReachCompletionWithRemoteHostname: disable on Linux. by Markus Handell · 4 years ago
  98. 9071957 Remove unused members in tests. by Åsa Persson · 4 years ago
  99. 55de292 Use relative paths for //net/dcsctp/public:socket. by Mirko Bonadei · 4 years ago
  100. 1cdeb0a addIceCandidate with callback into Android's SDK. by Yura Yaroshevich · 4 years ago