Google Git
Sign in
webrtc / src / d7842008efdd9d5cbfa4157dde78c13556c1636b
  1. d784200 Add utilities to facilitate correct usage of rtc::RefCounted classes. by Tomas Gunnarsson · 3 years, 11 months ago
  2. e6de5ae Remove virtual inheritance from RTCStatsCollector by Tomas Gunnarsson · 3 years, 11 months ago
  3. 0c3b909 Roll chromium_revision 1a13f11499..492c83d619 (872016:875565) by chromium-webrtc-autoroll · 3 years, 11 months ago
  4. f3a687a video_replay: add --start-timestamp and --stop-timestamp by Philipp Hancke · 3 years, 11 months ago
  5. 5663ce9 Avoid undefined behavior in a division operation. by Minyue Li · 3 years, 11 months ago
  6. 6674b98 Update WebRTC code version (2021-04-23T04:02:21). by webrtc-version-updater · 3 years, 11 months ago
  7. 0fd0d58 Implement FocusOnSelectedSource for WgcCapturerWin. by Austin Orion · 3 years, 11 months ago
  8. 88f4b33 usrsctp: Support sending and receiving empty messages by Florent Castelli · 3 years, 11 months ago
  9. 9bd2457 Delete SignalQueueDestroyed by Niels Möller · 3 years, 11 months ago
  10. 39e2385 Add conceptual documentation for Audio - Mixer by Alessio Bazzica · 3 years, 11 months ago
  11. feb6eb9 Create a test showing that maxRetransmits=0, ordered=false works by Harald Alvestrand · 3 years, 11 months ago
  12. 1366b0f AsyncResolver: avoid hanging the WorkerThread. by Markus Handell · 3 years, 11 months ago
  13. c5bac77 Add rendered_frames metric to DVQA. by Mirko Bonadei · 3 years, 11 months ago
  14. e7b752b Add fuzzer to validate libvpx vp9 encoder wrapper by Danil Chapovalov · 3 years, 11 months ago
  15. 898f091 Replace interfaces for sending RTCP with std::functions in ReceiveSideCongestionController by Per Kjellander · 3 years, 11 months ago
  16. 1585587 Uniform IPAddress::ToSensitiveString() behavior (debug vs release). by Mirko Bonadei · 3 years, 11 months ago
  17. 48171ec Remove more mentions of RTP datachannels by Harald Alvestrand · 3 years, 11 months ago
  18. 762f21c dcsctp: Add Send Queue by Victor Boivie · 3 years, 11 months ago
  19. 67b80ac Fix iOS chromium roll issue by Artem Titov · 3 years, 11 months ago
  20. 97c4458 PlatformThread: add support for detached threads. by Markus Handell · 3 years, 11 months ago
  21. 6ef4af9 Purge old FEC packets from receiver's queue before media sequence numbers wrap around by Harsh Maniar · 3 years, 11 months ago
  22. 20ee02c Add codec comparison function to SdpVideoFormat by Johannes Kron · 3 years, 11 months ago
  23. 86ee89f Simplify reference counting implementation of PendingTaskSafetyFlag. by Tommi · 3 years, 11 months ago
  24. e313c07 Fix iOS compilation for chromium roll by Artem Titov · 3 years, 11 months ago
  25. 63b01e1 Remove ReceiveDataParams::timestamp by Florent Castelli · 3 years, 11 months ago
  26. 49bec37 dcsctp: Log integers as unsigned by Victor Boivie · 3 years, 11 months ago
  27. 0e73602 dcsctp: Merge ReconfigResponseSN/ReconfigRequestSN by Victor Boivie · 3 years, 11 months ago
  28. 0b0afaa dcsctp: Add Chunk Validators by Victor Boivie · 3 years, 11 months ago
  29. 59d6e2a dcsctp: Add test for StrongAlias<bool> as bool by Victor Boivie · 3 years, 11 months ago
  30. 437d129 AEC3: Avoid overcompensating for render onsets during dominant nearend by Gustaf Ullberg · 3 years, 11 months ago
  31. 1153974 Fixed crash due wrong format specifier. by Yura Yaroshevich · 3 years, 11 months ago
  32. 319d76c Fix incorrect link in README.md by Byoungchan Lee · 3 years, 11 months ago
  33. b4ced39 dcsctp: Add OWNERS file by Florent Castelli · 3 years, 11 months ago
  34. c3fcee7 Move h264_profile_level_id and vp9_profile to api/video_codecs by Johannes Kron · 3 years, 11 months ago
  35. 8546666 Add threading assertions to TransceiverList by Harald Alvestrand · 3 years, 11 months ago
  36. dcac9fe Add may_contain_cursor property to DesktopFrame to avoid double capture by Austin Orion · 3 years, 11 months ago
  37. 688235d Exclude WS_EX_TOOLWINDOWs for WgcCapturerWin. by Austin Orion · 3 years, 11 months ago
  38. 516e284 Remove DataChannelType and deprecated option disable_sctp_data_channels by Florent Castelli · 3 years, 11 months ago
  39. eb9c3f2 Handle OnPacketSent on the network thread via MediaChannel. by Tomas Gunnarsson · 3 years, 11 months ago
  40. edb7ea2 Refactors Vp9UncompressedHeaderParser. by Erik Språng · 3 years, 11 months ago
  41. bfd9ba8 Fix unsafe variable access in RTCStatsCollector by Tomas Gunnarsson · 3 years, 11 months ago
  42. f703ed1 Ban std::shared_ptr in style guide by Harald Alvestrand · 3 years, 11 months ago
  43. 25e7352 Add support for setting the initial state to the pending task flag. by Tomas Gunnarsson · 3 years, 11 months ago
  44. e984aa2 Add thread accessors to Call. by Tomas Gunnarsson · 3 years, 11 months ago
  45. bddebc8 Fix an example in SequenceChecker documentation by Harald Alvestrand · 3 years, 11 months ago
  46. b849311 Update last received keyframe packet timestamp on all packets with the same RTP timestamp. by philipel · 3 years, 11 months ago
  47. 0d3c09a webrtc::Mutex: Introduce mutex_race_check.h. by Markus Handell · 3 years, 11 months ago
  48. d29c689 Expose adaptive_ptime from Android SDK. by Yura Yaroshevich · 3 years, 11 months ago
  49. d71b38e Update WebRTC code version (2021-04-19T04:03:03). by webrtc-version-updater · 3 years, 11 months ago
  50. d46a174 Expose adaptive_ptime from iOS SDK. by Yura Yaroshevich · 3 years, 11 months ago
  51. 7fa8d46 Slight code clarification in RemoveStoppedTransceivers. by Tomas Gunnarsson · 3 years, 11 months ago
  52. 0ee5bcf Update WebRTC code version (2021-04-18T04:03:49). by webrtc-version-updater · 4 years ago
  53. e632402 Remove rtp data channel related code from media_channel.* by Tomas Gunnarsson · 4 years ago
  54. 18ac30c Update WebRTC code version (2021-04-17T04:04:03). by webrtc-version-updater · 4 years ago
  55. 983b620 Remove third_party/xstream from DEPS by Bjorn Terelius · 4 years ago
  56. 78aa5cd dcsctp: Ensure packet size doesn't exceed MTU by Victor Boivie · 4 years ago
  57. 7af57c6 Remove RTP data implementation by Harald Alvestrand · 4 years ago
  58. f981cb3 Add video/g3doc/stats.md to the doc site menu by Artem Titov · 4 years ago
  59. 15e078c Fix unsignalled ssrc race in WebRtcVideoChannel. by Henrik Boström · 4 years ago
  60. 882d007 Add documentation for video/stats. by Åsa Persson · 4 years ago
  61. 0131a4d Delete StreamAdapterInterface by Niels Möller · 4 years ago
  62. b291da8 Add conceptual docs for modules/video_coding by Rasmus Brandt · 4 years ago
  63. dd36198 Revert "Expose AV1 encoder&decoder from Android SDK." by Björn Terelius · 4 years ago
  64. 220a252 Delete unused class MessageBufferReader by Niels Möller · 4 years ago
  65. 6c127a1 Add Stable Writable Connection Ping Interval parameter to RTCConfiguration. by Derek Bailey · 4 years ago
  66. 74b1bbe Remove unused a gn variable related to gtk by Byoungchan Lee · 4 years ago
  67. a43528c Update WebRTC code version (2021-04-16T04:04:52). by webrtc-version-updater · 4 years ago
  68. 3ceb16e [Android] Set use_raw_android_executable explicitly for test() template. by Peter Kotwicz · 4 years ago
  69. 0f57e0b Make libjingle_peerconnection_metrics_default_jni available in Linux builds. by Mirko Bonadei · 4 years ago
  70. 9fea310 Fix crash in WindowCapturerWinGdi::CaptureFrame. by Austin Orion · 4 years ago
  71. a80c3e5 sctp: Reorganize build targets by Florent Castelli · 4 years ago
  72. 6c7c495 doc: fix ice metadata + spelling by Philipp Hancke · 4 years ago
  73. fedd502 Expose AV1 encoder&decoder from Android SDK. by Yura Yaroshevich · 4 years ago
  74. 572f50f Delete left-over references to AsyncInvoker by Niels Möller · 4 years ago
  75. affd219 Delete AsyncInvoker usage from SimulatedPacketTransport by Niels Möller · 4 years ago
  76. bc959b6 Remove enable_rtp_data_channel by Harald Alvestrand · 4 years ago
  77. fa8a946 Remove obsolete DCHECK in remote_audio_source.cc. by Henrik Boström · 4 years ago
  78. 17490b5 Fix regression in UsrSctpReliabilityTest by Niels Möller · 4 years ago
  79. 403e328 Fix build with rtc_libvpx_build_vp9=false by Byoungchan Lee · 4 years ago
  80. 980c460 AGC2: retuning and large refactoring by Alessio Bazzica · 4 years ago
  81. d28434b Configure GN to use python3 to exec_script. by Mirko Bonadei · 4 years ago
  82. dad500a Remove PacketBuffers internal mutex. by philipel · 4 years ago
  83. 61982a7 AGC2 lightweight noise floor estimator by Alessio Bazzica · 4 years ago
  84. 3ab7a55 Reformat pacer doc and add it into sitemap by Artem Titov · 4 years ago
  85. 9aec8c2 Use default rtp parameters to init wrappers in iOS by Yura Yaroshevich · 4 years ago
  86. 89f3dd5 Make RTC_LOG_THREAD_BLOCK_COUNT less spammy for known call counts by Tomas Gunnarsson · 4 years ago
  87. 5744b7f Fix formatting in sitemap.md by Artem Titov · 4 years ago
  88. 08d30a2 Add documentation for video/adaptation by Evan Shrubsole · 4 years ago
  89. 24bc419 Revert "Fix RTP header extension encryption" by Björn Terelius · 4 years ago
  90. dea5721 Adding g3doc for AudioProcessingModule (APM) by Per Åhgren · 4 years ago
  91. 9861f96 dcsctp: Add operators on TimeMs and DurationMs by Victor Boivie · 4 years ago
  92. 8181b4f Add conceptual documentation for NetEq. by Jakob Ivarsson · 4 years ago
  93. a743303 Fix RTP header extension encryption by Lennart Grahl · 4 years ago
  94. 84ba164 Change from sakal@webrtc.org to xalep@webrtc.org in OWNERS files. by Mirko Bonadei · 4 years ago
  95. c54f672 dcsctp: Fix post-review comments for DataTracker by Victor Boivie · 4 years ago
  96. 0498519 Add g3doc for audio coding module. by Minyue Li · 4 years ago
  97. 1fad94f Remove ErleUncertainty by Gustaf Ullberg · 4 years ago
  98. 77d73a6 Document SctpTransport by Harald Alvestrand · 4 years ago
  99. 1d2d169 Update WebRTC code version (2021-04-14T04:04:15). by webrtc-version-updater · 4 years ago
  100. e871e02 Add telemetry to measure usage, perf, and errors in Desktop Capturers. by Austin Orion · 4 years ago