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  1. 74154e6 Save delays in history for 2 seconds instead of fixed 100 packets. by Jakob Ivarsson · 3 hours ago master
  2. 4e615d5 Wire the stable target bitrate from GoogCC to the BitrateAllocator by Florent Castelli · 4 hours ago
  3. 3dd1985 Delete unused function MediaTypeFromString by Niels Möller · 7 hours ago
  4. b88b44e Don't include duplicated and incomplete frames in stats. by Johannes Kron · 8 hours ago
  5. d47941e Reland "Simplification and refactoring of the AudioBuffer code" by Per Åhgren · 9 hours ago
  6. a2dae38 Revert "Reland "Delete mac_utils.h and mac_utils.cc"" by Niels Moller · 10 hours ago lkgr
  7. 05f8f1d Add helper classes to send and receive abs-capture-time extensions. by Chen Xing · 10 hours ago
  8. 9fd2908 Remove unused framerate parameter from simulcast bitrate allocator. by Jonas Olsson · 11 hours ago
  9. df57833 Reland "Delete mac_utils.h and mac_utils.cc" by Niels Möller · 11 hours ago
  10. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 12 hours ago
  11. b689af4 Changes to enable use of DatagramTransport as a data channel transport. by Bjorn A Mellem · 25 hours ago
  12. f254e9e Revert "Simplification and refactoring of the AudioBuffer code" by Steve Anton · 26 hours ago
  13. f5815fa Remove WebRTC-Pacer-LegacyPacketReferencing flag and most usage by Erik Språng · 27 hours ago
  14. 1c602e3 Process 8 kHz audio as 16 kHz internally of the audio processing module by Per Åhgren · 29 hours ago
  15. 81c0cf2 Simplification and refactoring of the AudioBuffer code by Per Åhgren · 30 hours ago
  16. f69bd5f Delete AudioDeviceWindowsCore::WideToUTF8, replaced with rtc::ToUtf8 by Niels Möller · 30 hours ago
  17. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 31 hours ago
  18. 21e99da Add implemented-but-missing members to RTCMediaStreamTrackStats::Members by Henrik Boström · 33 hours ago
  19. 1c2f637 Simplify the VideoFrameDumpingDecoder API. by Markus Handell · 34 hours ago
  20. 54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 34 hours ago
  21. e8ef87b Include menus & dialogs in frames captured by WindowCapturerWin by Bryan Ferguson · 36 hours ago
  22. 364b267 Replace DatagramDtlsAdaptor with DatagramRtpTransport. by Bjorn A Mellem · 2 days ago
  23. a310b38 Roll chromium_revision 5a34954f26..318f298cba (688384:688507) by chromium-webrtc-autoroll · 2 days ago
  24. 2dac4e4 Remove rtc_use_lto GN arg. by Mirko Bonadei · 2 days ago
  25. 5ceb4ac Delete some unused AudioCodingModule methods by Niels Möller · 2 days ago
  26. 728a0ee Reland "Introduce ability to test echo in PC level test framework" by Artem Titov · 2 days ago
  27. a854921 Enable custom metrics gathering from stats API in PC framework. by Artem Titov · 2 days ago
  28. e21f3f5 Revert "Delete mac_utils.h and mac_utils.cc" by Niels Moller · 2 days ago
  29. 6b117a5 Make the callbacks to PollStats for RampUp* tests more regular. by Tommi · 2 days ago
  30. ada8e17 Delete mac_utils.h and mac_utils.cc by Niels Möller · 2 days ago
  31. 928146f Removing all external access to the integer sample data in AudioBuffer by Per Åhgren · 2 days ago
  32. 93d4c10 Declare references as constant in the metal renderers. by Kári Tristan Helgason · 2 days ago
  33. 2579f0c RTCError as return type for PeerConnectionInterface::SetConfiguration by Niels Möller · 3 days ago
  34. 7627fdd Sanitize the address field of peer-reflexive remote candidates. by Qingsi Wang · 3 days ago
  35. 7c78e42 Roll chromium_revision 21d23ea529..5a34954f26 (688221:688384) by chromium-webrtc-autoroll · 3 days ago
  36. 587991c Remove jeroendb@webrtc.org from OWNERS by Steve Anton · 3 days ago
  37. a0b52b5 Remove zhihuang@webrtc.org from OWNERS by Steve Anton · 3 days ago
  38. 9bdb1b1 Roll chromium_revision afb0a631b9..21d23ea529 (688061:688221) by chromium-webrtc-autoroll · 3 days ago
  39. 1ba5dec Reland "Set the usage pattern bits for adding remote ICE candidates from SDP." by Qingsi Wang · 3 days ago
  40. 0d1996f Removes empty p2p/base/transport.h by Sebastian Jansson · 3 days ago
  41. fdf3880 Make "WebRTC-BweAllocProbingOnlyInAlr/Enabled/" default and remove key. by Konrad Hofbauer · 3 days ago
  42. e3a10e1 Remove usage of RtpRtcp::SetSSRC() in video/ by Erik Språng · 3 days ago
  43. 185243b Remove most of PacedSenderUnittest by Erik Språng · 3 days ago
  44. a2bc362 Roll chromium_revision c7f14188a3..afb0a631b9 (687843:688061) by Artem Titarenko · 3 days ago
  45. fce0b72 NetEq fuzzer: reduce max input size slightly to avoid timeout by Henrik Lundin · 3 days ago
  46. 62c174c Reland of Correct conversion between float and fixed formats by Per Åhgren · 3 days ago
  47. 5870503 Revert "Introduce ability to test echo in PC level test framework" by Sami Kalliomäki · 3 days ago
  48. 93f5189 Remove some usage of RtpRtcp::SetSSRC() by Erik Språng · 3 days ago
  49. cd277b8 AEC3: Fix computation of audio buffer delay by Gustaf Ullberg · 3 days ago
  50. 17f9ee5 Enable `VideoReceiveStreamTestWithFakeDecoder.RenderedFrameUpdatesGetSources` for iOS. by Chen Xing · 3 days ago
  51. 77acb01 Introduce ability to test echo in PC level test framework by Artem Titov · 3 days ago
  52. 3ab8eb5 Add steps logging into PC test framework by Artem Titov · 3 days ago
  53. 83773b5 Delete deprecated RtpRtcp::CreateRtpRtcp factory by Danil Chapovalov · 3 days ago
  54. 3c7abdc Roll chromium_revision d5a13ccb8e..c7f14188a3 (687732:687843) by Artem Titarenko · 3 days ago
  55. 637f110 Remove rtcbot. by Mirko Bonadei · 3 days ago
  56. 6b43086 Reland "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Boström · 3 days ago
  57. d05edec Extract most of PacedSender into PacedSendingController. by Erik Språng · 3 days ago
  58. 9755f97 Update constributing source before publishing frame by Jonas Oreland · 3 days ago
  59. fa8f4ee Only combine media transport and ICE states if used for media. by Bjorn A Mellem · 6 days ago
  60. 015c3cb Remove deprecated constructors of RtpSource by Johannes Kron · 6 days ago
  61. 0e1a558 Allowing 40ms audio frame length. by Ying Wang · 6 days ago
  62. 0ee4311 Roll chromium_revision c7f850c75e..d5a13ccb8e (687596:687732) by chromium-webrtc-autoroll · 6 days ago
  63. f5e5d25 BalancedDegradationSettings: add option to configure a min framerate diff. by Åsa Persson · 6 days ago
  64. df625f4 Revert "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Andreassson · 6 days ago
  65. 6094953 Add helper functions to convert between integer milliseconds and fixed-point seconds. by Chen Xing · 6 days ago
  66. 2b9fa09 [GetStats] Expose video codec implementation in standardized metrics. by Henrik Boström · 6 days ago
  67. cc96db6 Simplify stats poller stop in PC level framework by Artem Titov · 6 days ago
  68. 6950b30 Fix thread naming in Call constructor by Erik Språng · 6 days ago
  69. bbeb109 Reporting audio device underrun counter by Alex Narest · 6 days ago
  70. 9b29d69 Make ANA frame length controller more robust to encoder frame lengths. by Minyue Li · 6 days ago
  71. 533c225 Revert "Correct conversion between float and fixed formats" by Henrik Andreassson · 6 days ago
  72. 07a6652 Roll chromium_revision f54998af9c..c7f850c75e (687496:687596) by chromium-webrtc-autoroll · 6 days ago
  73. 98bbd88 Roll chromium_revision 7a2da7b921..f54998af9c (686822:687496) by chromium-webrtc-autoroll · 7 days ago
  74. e5defb1 Sanitize the selected candidate pair in the public API. by Qingsi Wang · 7 days ago
  75. ffc525b Fix a bug/typo in WebRtcSpl_FilterAR which updates the wrong state vector by Jiawei Ou · 7 days ago
  76. 67e43c8 Correct conversion between float and fixed formats by Per Åhgren · 7 days ago
  77. a135127 Remove all AudioBuffer code that is not related to storing audio data by Per Åhgren · 7 days ago
  78. 6e4791f Add check for IsCurrent() for SendTask in SingleThreadedTaskQueueForTesting. by Tommi · 7 days ago
  79. 65feec5 Reenable UlpfecWithNack integration tests by Danil Chapovalov · 7 days ago
  80. 1b247f1 BalancedDegradationSettings: add option to configure min bitrate. by Åsa Persson · 8 days ago
  81. 3aa0d76 Use struct parser for AlrDetector config. by Sebastian Jansson · 8 days ago
  82. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 8 days ago
  83. 09ba219 Roll chromium_revision fa752aeae4..7a2da7b921 (686692:686822) by chromium-webrtc-autoroll · 8 days ago
  84. c759f83 Avoid copying of vectors in RtpPacketInfos. by Minyue Li · 8 days ago
  85. c14b233 Disable the most flaky tests on iOS. by Sami Kalliomäki · 8 days ago
  86. 7daf550 Add new FrameRateEstimator utility class for more precis FPS estimation. by Erik Språng · 8 days ago
  87. 0ee8008 Use struct parser for rate control trial. by Sebastian Jansson · 8 days ago
  88. ad9113f Adds sequence numbers to feedback generator output. by Sebastian Jansson · 8 days ago
  89. 0c38a86 BalancedDegradationSettings: add option to configure no fps limit. by Åsa Persson · 8 days ago
  90. 704c8c4 Re-enable AudioDeviceTest in combination with sanitizers. by Yves Gerey · 8 days ago
  91. 78c56cb Delete deprecated version of ReceiveStatistics::Create by Niels Möller · 8 days ago
  92. 1e04a9b Roll chromium_revision bcb9240637..fa752aeae4 (686583:686692) by chromium-webrtc-autoroll · 9 days ago
  93. fb6edd3 Handle case of empty connection in pair change event by Alex Drake · 9 days ago
  94. bb19942 Roll chromium_revision 6652dd41e1..bcb9240637 (686436:686583) by chromium-webrtc-autoroll · 9 days ago
  95. 68c2a56 Propagating Network Type in Candidate for JNI by Alex Drake · 9 days ago
  96. 608e6ba Add AudioDecoderIsacT::Config to include sampling rate and BWInfo object by Jiawei Ou · 9 days ago
  97. 05497f2 Pull a DataChannelTransportInterface out of MediaTransportInterface. by Bjorn A Mellem · 9 days ago
  98. d419808 Revert "Set the usage pattern bits for adding remote ICE candidates from SDP." by Qingsi Wang · 9 days ago
  99. 7c6f74a Set the usage pattern bits for adding remote ICE candidates from SDP. by Qingsi Wang · 9 days ago
  100. 1a03784 Roll chromium_revision 3ae2445b34..6652dd41e1 (686310:686436) by chromium-webrtc-autoroll · 9 days ago