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  1. 2ab4764 Clean-up for calculation of upper bandwidth limit. by Christoffer Rodbro · 3 years, 1 month ago
  2. 81e13d3 Roll chromium_revision 2826799ea1..1b27d646a6 (885837:886225) by chromium-webrtc-autoroll · 3 years, 1 month ago
  3. 36005af Refactor and improve RtpSender packet history test. by Erik Språng · 3 years, 1 month ago
  4. 02c0295 Remove obsolete DCHECK in RtpPacket::CopyHeaderFrom by Danil Chapovalov · 3 years, 1 month ago
  5. 6396b48 Avoid modifying BWE internal state on reception of REMB feedback. by Christoffer Rodbro · 3 years, 1 month ago
  6. c09c581 dcsctp: Limit the size of generated SACK chunks by Victor Boivie · 3 years, 1 month ago
  7. e4adedc Update WebRTC code version (2021-05-25T04:03:57). by webrtc-version-updater · 3 years, 1 month ago
  8. 41a111d Switch to av_packet_alloc() by Ted Meyer · 3 years, 1 month ago
  9. 0f50678 Remove usage of TOOLKIT_GTK define. by Byoungchan Lee · 3 years, 2 months ago
  10. 816134a Reland "Fix race between enabled() and set_enabled() in VideoTrack." by Tommi · 3 years, 1 month ago
  11. 2f3c5e6 Skip WindowCapturerTest.Capture on macOS. by Mirko Bonadei · 3 years, 1 month ago
  12. ae0d117 Implement the mixer-to-client per CSRC audio level extension (RFC 6465). by Doudou Kisabaka · 3 years, 1 month ago
  13. 096ad02 Revert "Fix race between enabled() and set_enabled() in VideoTrack." by Evan Shrubsole · 3 years, 1 month ago
  14. 2fa4774 Revert "Deprecate microsecond timestamps in RTC event log." by Björn Terelius · 3 years, 1 month ago
  15. cf49789 Refactor some retransmission tests. by Erik Språng · 3 years, 1 month ago
  16. 006815e Add temp peer_connection_interface include rtc_base/event.h by Evan Shrubsole · 3 years, 1 month ago
  17. e6ee8fa Deprecate microsecond timestamps in RTC event log. by Björn Terelius · 3 years, 1 month ago
  18. 5ffefe9 Fix race between enabled() and set_enabled() in VideoTrack. by Tommi · 3 years, 1 month ago
  19. 13e5851 Update WebRTC code version (2021-05-24T04:02:02). by webrtc-version-updater · 3 years, 1 month ago
  20. f3d71c2 Roll chromium_revision 70eb2d0977..2826799ea1 (885736:885837) by chromium-webrtc-autoroll · 3 years, 1 month ago
  21. 93faab1 dcsctp: Implement Round Robin scheduler by Victor Boivie · 3 years, 1 month ago
  22. 70cd086 SEA: Only spawn multi-layered encoders if active layers > 1. by Henrik Boström · 3 years, 1 month ago
  23. bcadacd Update WebRTC code version (2021-05-23T04:03:13). by webrtc-version-updater · 3 years, 1 month ago
  24. 2440d34 dcsctp: Rename FCFSSendQueue to RRSendQueue by Victor Boivie · 3 years, 1 month ago
  25. 913c3af Update WebRTC code version (2021-05-22T04:04:00). by webrtc-version-updater · 3 years, 1 month ago
  26. 7ee9b6b Roll chromium_revision 9431bab2be..70eb2d0977 (885625:885736) by chromium-webrtc-autoroll · 3 years, 1 month ago
  27. 9f6808b Remove cricket::DtlsTransportState. by Mirko Bonadei · 3 years, 1 month ago
  28. e976f75 Roll chromium_revision e5dd2eb61e..9431bab2be (884954:885625) by chromium-webrtc-autoroll · 3 years, 1 month ago
  29. 20f9401 Remove GTEST_ALLOW_UNINSTANTIATED in datachannel tests by Harald Alvestrand · 3 years, 1 month ago
  30. 32ee3b8 dcsctp: Ensure RTO is always greater than RTT by Victor Boivie · 3 years, 1 month ago
  31. cab90db Delete `NtpOffsetMs` and `TimeMicrosToNtp` methods. by Paul Hallak · 3 years, 1 month ago
  32. 46fbefa Convert to NTP time using the real clock. by Paul Hallak · 3 years, 1 month ago
  33. a6b0d53 Delete the old flavor of RtcpTransceiverImpl::ReceivePacket by Paul Hallak · 3 years, 1 month ago
  34. fe3dd51 Use the injected clock in rtcp_transciever. by Paul Hallak · 3 years, 1 month ago
  35. 61a287a Add accessor for UTC start time in event log by Björn Terelius · 3 years, 1 month ago
  36. 00f6e75 Use webrtc::Clock to query for the NTP time and to convert timestamps to NTP. by Paul Hallak · 3 years, 1 month ago
  37. 1cb796f Add performance tracing for API calls (inside api proxies). by Tommi · 3 years, 1 month ago
  38. 47ed998 Use the clock to convert absolute capture timestamps to NTP times. by Paul Hallak · 3 years, 1 month ago
  39. edc347c Introduce (Un)SubscribeDtlsTransportState methods. by Mirko Bonadei · 3 years, 1 month ago
  40. 95f1e51 Do not attempt setting the absolute capture time extension if we don't by Paul Hallak · 3 years, 1 month ago
  41. 2491dbd Make Clock::ConvertTimestampToNtpTime pure virtual by Paul Hallak · 3 years, 1 month ago
  42. e93fe6c Enable Chromium to stop including api/proxy.h indirectly. by Markus Handell · 3 years, 1 month ago
  43. b59e904 Add the ability to convert a timestamp to NTP time. by Paul Hallak · 3 years, 1 month ago
  44. 0cff391 Start with a BeginLog event in event log encoder unittest by Björn Terelius · 3 years, 1 month ago
  45. b8dc7fa Make AgcManagerDirect clipping parameters configurable by Hanna Silen · 3 years, 1 month ago
  46. e2b9fc6 Move FecOverheadRate, BitrateCallbacks to rtp_sender_egress_unittest. by Erik Språng · 3 years, 1 month ago
  47. 63b3095 Make local to capturer clock offset a separate entry in PacketInfo. by Minyue Li · 3 years, 1 month ago
  48. cbde0cf Roll chromium_revision 6291fe6f0e..e5dd2eb61e (884821:884954) by chromium-webrtc-autoroll · 3 years, 1 month ago
  49. 0de1ed0 Have only two pure virtual methods for webrtc::Clock, by Paul Hallak · 3 years, 1 month ago
  50. 8ed1e93 Switch from check_targets to no_check_targets in .gn by Byoungchan Lee · 3 years, 1 month ago
  51. 193f4bf Replace legacy getStats with standard getStats in the iOS example by Jaehyun Ko · 3 years, 1 month ago
  52. bd346d7 Update WebRTC code version (2021-05-20T04:01:58). by webrtc-version-updater · 3 years, 1 month ago
  53. 069ed35 Roll chromium_revision 0321a6153c..6291fe6f0e (884706:884821) by chromium-webrtc-autoroll · 3 years, 1 month ago
  54. 4fb5684 Roll chromium_revision b8d2317c2e..0321a6153c (884575:884706) by chromium-webrtc-autoroll · 3 years, 1 month ago
  55. 6c96611 Roll chromium_revision 3bdbd47d23..b8d2317c2e (884422:884575) by chromium-webrtc-autoroll · 3 years, 1 month ago
  56. 86bd92f Add test for many non-dropped packets in unreliable mode. by Harald Alvestrand · 3 years, 1 month ago
  57. e7481a4 Add an UlpFec test to RtpRtcp unit tests. by Erik Språng · 3 years, 1 month ago
  58. 398def6 Improvements to AEC3 logging to simplify debugging by Gustaf Ullberg · 3 years, 1 month ago
  59. aeb8ce8 AEC3: Change adaptation speed of the matched filter after a delay is found by Gustaf Ullberg · 3 years, 1 month ago
  60. 402ceff sctp: Reduce logging level for common calls by Victor Boivie · 3 years, 1 month ago
  61. 3a45d32 dcsctp: Report duplicate TSNs by Victor Boivie · 3 years, 1 month ago
  62. 91fef02 Roll chromium_revision 108d11241e..3bdbd47d23 (884294:884422) by chromium-webrtc-autoroll · 3 years, 1 month ago
  63. f6be1b2 Simplify RtpSenderTest.SendFlexfecPackets and move to RtpRtcp-level. by Erik Språng · 3 years, 1 month ago
  64. 3de4067 Increase the bound for the value of the filter reset block interval by Per Åhgren · 3 years, 1 month ago
  65. 12c881b Make RtpHelper<>::sending_ atomic. by Tommi · 3 years, 1 month ago
  66. 38f1d4b [LibvpxVp8Encoder] Don't DCHECK crash if I420 is not equal to I420A. by Henrik Boström · 3 years, 1 month ago
  67. 91a892f Add ability to dump the coarse filter in the echo subtractor by Per Åhgren · 3 years, 1 month ago
  68. cf0ec28 Delete RtcpStatistic struct as no longer used by Danil Chapovalov · 3 years, 1 month ago
  69. 5f4efd5 Update WebRTC code version (2021-05-19T04:03:01). by webrtc-version-updater · 3 years, 1 month ago
  70. 723e92e Roll chromium_revision 19bcae5aaa..108d11241e (884108:884294) by chromium-webrtc-autoroll · 3 years, 1 month ago
  71. fab9256 Roll chromium_revision a0592593ad..19bcae5aaa (883988:884108) by chromium-webrtc-autoroll · 3 years, 1 month ago
  72. ee483f7 Roll chromium_revision ea49ecba06..a0592593ad (883877:883988) by chromium-webrtc-autoroll · 3 years, 1 month ago
  73. db28555 Improve test coverage for padding packet generation. by Erik Språng · 3 years, 1 month ago
  74. 567e847 Move Send(Generic|Raw)Video from rtp sender unittest to RtpRtcp-level. by Erik Språng · 3 years, 1 month ago
  75. ea7474e Remove redundant VideoSendStream::rtcp_stats field by Danil Chapovalov · 3 years, 1 month ago
  76. a399c82 Field trial to disable the transient suppressor by Gustaf Ullberg · 3 years, 1 month ago
  77. a77e16c Update BitBuffer methods to style guide by Björn Terelius · 3 years, 1 month ago
  78. 7c286c0 Roll chromium_revision 1c010044c4..ea49ecba06 (883756:883877) by chromium-webrtc-autoroll · 3 years, 1 month ago
  79. fe6595f Include all RTP packet infos from the mix list when updating the audio frame for mixing. by Doudou Kisabaka · 3 years, 1 month ago
  80. 92bd902 dcsctp: Restrict fuzzing input length by Victor Boivie · 3 years, 1 month ago
  81. 718acf6 peerconnection: add test for createOffer({voiceActivityDetection: true}) by Philipp Hancke · 3 years, 1 month ago
  82. 4e9442d Update WebRTC code version (2021-05-18T04:02:06). by webrtc-version-updater · 3 years, 1 month ago
  83. 2e929ea Roll chromium_revision 2620ef6e05..1c010044c4 (883646:883756) by chromium-webrtc-autoroll · 3 years, 1 month ago
  84. 8b94832 Roll chromium_revision fb62f861d3..2620ef6e05 (883525:883646) by chromium-webrtc-autoroll · 3 years, 1 month ago
  85. c2310b2 Set nativeObserver to 0 to avoid double release. by Tian Tan · 3 years, 1 month ago
  86. bc42984 Roll chromium_revision 5469e0bb11..fb62f861d3 (883387:883525) by chromium-webrtc-autoroll · 3 years, 1 month ago
  87. eadf457 Remove check following SetChannel. by Tommi · 3 years, 1 month ago
  88. cfa932f dcsctp: Bump rto_min to 220 ms by Victor Boivie · 3 years, 1 month ago
  89. 7ddadbc APM: dump `GainController1::AnalogGainController` in `Config::ToString` by Alessio Bazzica · 3 years, 1 month ago
  90. dbf13e3 AudioMixer: make the number of sources to mix configurable. by Doudou Kisabaka · 3 years, 1 month ago
  91. 726b0e8 Refactor RtpSenderTest.TrafficSmoothingW* tests by Erik Språng · 3 years, 1 month ago
  92. 4ccdf932 VideoRtpReceiver & AudioRtpReceiver threading fixes. by Tommi · 3 years, 1 month ago
  93. b27a9f9 Cleanup ReceiveStatistics collecting ReportBlock by Danil Chapovalov · 3 years, 1 month ago
  94. 4310375 Move SendPacketObserver tests to rtp_sender_egress_unittest. by Erik Språng · 3 years, 1 month ago
  95. 7a86aad Refactor RtpSenderTest.SendPadding. by Erik Språng · 3 years, 1 month ago
  96. 36b7d10a Delete unused test method in neteq that uses RtcpStatistics by Danil Chapovalov · 3 years, 1 month ago
  97. 4e60937 Add quality upscaling tests. by Åsa Persson · 3 years, 1 month ago
  98. fef609a Roll chromium_revision 60ef3818ac..5469e0bb11 (883287:883387) by chromium-webrtc-autoroll · 3 years, 1 month ago
  99. b247478 Update WebRTC code version (2021-05-17T04:03:41). by webrtc-version-updater · 3 years, 1 month ago
  100. 95aaf28 Refactors yet more rtp_sender_unitttests into rtp_sender_egress_unittest by Erik Språng · 3 years, 1 month ago