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gerrit-public.fairphone.software / platform / external / webrtc / b24c00f02d90261a97fac77c393304f2c170bcc1
  1. b24c00f Add AudioProcessingCaptureStats and a level estimator replacement by Sam Zackrisson · 6 years ago
  2. 2918d4e Roll chromium_revision 7579fcbc1c..00f78b5b14 (610728:610831) by chromium-webrtc-autoroll · 6 years ago
  3. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  4. b253303 Add magjed as owner of rtc_tools. by Patrik Höglund · 6 years ago
  5. 856cf22 In ReceiveStatistics use monotonic clock instead of ntp clock by Danil Chapovalov · 6 years ago
  6. 22027b9 Add a new Task Queue for WinUWP. by Robin Raymond · 6 years ago
  7. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  8. 8ce0d2b In ReceiveStatistic require callbacks during construction by Danil Chapovalov · 6 years ago
  9. 4c0cc5b Reland Profile 2 to default profiles by Emircan Uysaler · 6 years ago
  10. f1c194d Roll chromium_revision d298cced6c..7579fcbc1c (610627:610728) by chromium-webrtc-autoroll · 6 years ago
  11. 05aee74 Roll chromium_revision f9be7d3d66..d298cced6c (610432:610627) by Oleh Prypin · 6 years ago
  12. 9289eda Revert "Replace the IceConnectionState implementation." by Alex Loiko · 6 years ago
  13. 4f00075 Remove use of CodecSpecificInfo.codec_name by Erik Språng · 6 years ago
  14. 1e87b4f Replace the IceConnectionState implementation. by Jonas Olsson · 6 years ago
  15. 57f3ad0 Adds stable bandwidth estimate to GoogCC. by Sebastian Jansson · 6 years ago
  16. 88ce4ef Don't buffer encoded frames. by Sergey Silkin · 6 years ago
  17. 885cf60 Moves ProbeBitrateEstimator from DelayBasedBwe. by Sebastian Jansson · 6 years ago
  18. e3abb81 Decouple //rtc_base:rtc_base_tests_utils from gunit. by Mirko Bonadei · 6 years ago
  19. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  20. b357e54 Add field trial config to disable pacer emergency stops. by Christoffer Rodbro · 6 years ago
  21. 6d254bc Delete unused method NetEq::PacketBufferStatistics by Niels Möller · 6 years ago
  22. 5f2ffee Clean up deprecated APM stats by Sam Zackrisson · 6 years ago
  23. f40150d Removing ANA enabling field trials. by Minyue Li · 6 years ago
  24. 2c977b4 Remove RSID from stream configs in new event log format. by Bjorn Terelius · 6 years ago
  25. 14dfe7f [GN] Fix dependency rebasing in BUILD.gn files. by Yves Gerey · 6 years ago
  26. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  27. 3890c1a Roll chromium_revision 1500c78c93..f9be7d3d66 (610314:610432) by chromium-webrtc-autoroll · 6 years ago
  28. 3955a50 Metal: Don't render into an empty view. by Peter Hanspers · 6 years ago
  29. 777cf26 AEC3: Clockdrift detection by Gustaf Ullberg · 6 years ago
  30. f259078 Use cropping aligning in video quality analysis tool by Magnus Jedvert · 6 years ago
  31. ebb50c2 Fix setting max reordering threshold in ReceiveStatistics by Danil Chapovalov · 6 years ago
  32. 286df00 Add tool for aligning cropped region of video files by Magnus Jedvert · 6 years ago
  33. 8e66863 Remove cricket::UdpTransport. by Mirko Bonadei · 6 years ago
  34. 94c9420 Remove cricket::BundleFilter. by Mirko Bonadei · 6 years ago
  35. eccfc47 Cleanup AimdRateController and remove RateControlRegion enum. by Bjorn Terelius · 6 years ago
  36. 42d2e4b Increase test timeouts in TCPChannelClientTest by Artem Titarenko · 6 years ago
  37. 00dfe93 Remove superfluous constructor from dltsTransport by Harald Alvestrand · 6 years ago
  38. 44727b4 Cleanup rtcp StreamStatistician::OnRtpPacket by Danil Chapovalov · 6 years ago
  39. af228ee Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN. by Alex Loiko · 6 years ago
  40. 5486bcd Remove SetChannelParameters function from API classes. by philipel · 6 years ago
  41. ecd6205 Disable GoogCcNetworkControllerTest.DetectsHighRateInSafeResetTrial by Alex Loiko · 6 years ago
  42. 8ac05cc Adds trial to use link capacity estimate in Opus encoder. by Sebastian Jansson · 6 years ago
  43. 2ff3f49 Move webrtc::CreatePeerConnectionFactory definition next to decl. by Mirko Bonadei · 6 years ago
  44. d51b355 Delete unused NetEq Rtcp stats. by Niels Möller · 6 years ago
  45. 7c36c71 Roll chromium_revision 6931f4c0d0..1500c78c93 (610209:610314) by chromium-webrtc-autoroll · 6 years ago
  46. 8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
  47. 4a2dd7a Roll chromium_revision 5825fead7b..6931f4c0d0 (610108:610209) by chromium-webrtc-autoroll · 6 years ago
  48. 540ef28 Adds OnReceivedUplinkAllocation method to AudioEncoder. by Sebastian Jansson · 6 years ago
  49. 6736df1 Moves BitrateAllocationUpdate to api. by Sebastian Jansson · 6 years ago
  50. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  51. e4cccae Removed ability to set CryptoOptions through PeerConnectionFactory from bindings. by Benjamin Wright · 6 years ago
  52. a526ae6 Roll chromium_revision 92f8c5b2a2..5825fead7b (609994:610108) by chromium-webrtc-autoroll · 6 years ago
  53. 5eae1d9 Remove legacy SetTargetTransferRateObserver by Piotr (Peter) Slatala · 6 years ago
  54. 37227be Add check for media transport and bundle policy by Piotr (Peter) Slatala · 6 years ago
  55. 47dfdca Create 'MaybeCreateMediaTransport' function by Piotr (Peter) Slatala · 6 years ago
  56. 64bfcde Add sakal@ to OWNERS in android tests / aarproject directories. by Sami Kalliomäki · 6 years ago
  57. 4749e4e Move HdrMetadata to ColorSpace by Johannes Kron · 6 years ago
  58. ecf6315 AGC2 adaptive digital: remove unnecessary flag. by Alessio Bazzica · 6 years ago
  59. 8da7b35 AGC2 adaptive digital false by default by Alessio Bazzica · 6 years ago
  60. cfddbb7 Add ios bindings for PeerConnectionState. by Jonas Olsson · 6 years ago
  61. 49a7843 Don't restart streams in scenario tests. by Sebastian Jansson · 6 years ago
  62. 0e4dfcb Roll chromium_revision 16e6b25329..92f8c5b2a2 (609893:609994) by chromium-webrtc-autoroll · 6 years ago
  63. 59a01b0 Set Framerate in RTCVideoEncoderH264 by Qiang Chen · 6 years ago
  64. 2b5b0e9 Disabling ScreenDrawerTest.TwoScreenDrawerLocks by Alex Loiko · 6 years ago
  65. c4d5642 Revert "Default to dlopening the PipeWire." by Oleh Prypin · 6 years ago
  66. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  67. a13be01 Default to dlopening the PipeWire. by Tomas Popela · 6 years ago
  68. c68d282 Add test PeerConnectionIntegrationTest.MediaTransportBidirectionalAudio by Niels Möller · 6 years ago
  69. 89c94b9 Adds target bandwidth to BitrateAllocator. by Sebastian Jansson · 6 years ago
  70. 66eedce Roll chromium_revision 7d53bc243c..16e6b25329 (609559:609893) by chromium-webrtc-autoroll · 6 years ago
  71. bd04f4a Increase buffer level threshold in VP8/9 tests. by Sergey Silkin · 6 years ago
  72. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  73. 38332cd Add RTCP and simulcast support for RTCRtpReceiver::getParameters() by Florent Castelli · 6 years ago
  74. 4bc6045 Add output directory option for audioproc_f data dump files. by Alessio Bazzica · 6 years ago
  75. 388e4e9 Make RTC_LOG_FILE_LINE use its parameters by Jonas Olsson · 6 years ago
  76. c20b82a Remove unused variables in RtcEventAudioXStreamConfig::Copy() by Bjorn Terelius · 6 years ago
  77. 22b70ff Move VideoCodecType from common_types.h to api/video/video_codec_type.h by Niels Möller · 6 years ago
  78. 22ff1a4 Fix threshold in VideoCodecTestLibvpx.ChangeFramerateVP9. by Mirko Bonadei · 6 years ago
  79. 6817038 APM audioproc_f: flag for AGC2 adaptive level estimator. by Alessio Bazzica · 6 years ago
  80. 44974e1 AEC3: Adding a correction factor for the Erle estimation that depends on the portion of the filter that is currently in use. by Jesús de Vicente Peña · 6 years ago
  81. 985a1f3 Add const or GUARDED_BY on a few ChannelSend members by Niels Möller · 6 years ago
  82. 5f00995 Using unit classes in AimdRateControl. by Sebastian Jansson · 6 years ago
  83. 50b8426 Roll chromium_revision 2f3cca903d..7d53bc243c (609431:609559) by chromium-webrtc-autoroll · 6 years ago
  84. f85b6d2 Roll chromium_revision 9508bd7fec..2f3cca903d (609314:609431) by chromium-webrtc-autoroll · 6 years ago
  85. b6787bc Using data unit classes in DelayBasedBwe. by Sebastian Jansson · 6 years ago
  86. 2e0c655 [Sanitizers] Don't retry failed tests. by Yves Gerey · 6 years ago
  87. b22f077 Adds FieldTrialConstrained class. by Sebastian Jansson · 6 years ago
  88. 76f5750 Roll chromium_revision 3efc758c50..9508bd7fec (609210:609314) by chromium-webrtc-autoroll · 6 years ago
  89. 85340ce Move rtc::scoped_refptr to api/. by Mirko Bonadei · 6 years ago
  90. 52e69d7 Explicitly specify color space enum indices by Johannes Kron · 6 years ago
  91. 3a83748 New loss-based bandwidth control mechanism. by Christoffer Rodbro · 6 years ago
  92. 26e88b0 Replace RTC_DCHECK by RTC_DCHECK_RUN_ON for worker thread. by Niels Möller · 6 years ago
  93. 2058d52 Disabling test StunPortTest.TestPrepareAddressHostname on WIN. by Alex Loiko · 6 years ago
  94. eb13484 Remove ChannelSendState by Fredrik Solenberg · 6 years ago
  95. c3313a3 Make api:create_peerconnection_factory public. by Mirko Bonadei · 6 years ago
  96. c5e8be3 Remove ChannelReceiveState by Fredrik Solenberg · 6 years ago
  97. 72bba62 Adds shared base class for data units. by Sebastian Jansson · 6 years ago
  98. d474672 Make rtc_event_log protos publicly visible. by Mirko Bonadei · 6 years ago
  99. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  100. 3cf8f3e Adding empty api:create_peerconnection_factory. by Mirko Bonadei · 6 years ago