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gerrit-public.fairphone.software / platform / external / webrtc / ad88c886d7f4cee4c76be740da207d812a086efa
  1. ad88c88 Add API for returning a webrtc::DtlsTransport for a MID on a PC by Harald Alvestrand · 6 years ago
  2. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  3. 0f9c38e Add new names of perf bots that will be migrated to LUCI by Oleh Prypin · 6 years ago
  4. 3312092 Keep bitrate constraints. by Sergey Silkin · 6 years ago
  5. ff088a1 Reland "Run robolectric tests for Android on several Android API versions" by Artem Titarenko · 6 years ago
  6. d5696fb Add video support to LoopbackMediaTransport by Niels Möller · 6 years ago
  7. bb9f4c1 Delete ssrc book-keeping in NetEq by Niels Möller · 6 years ago
  8. 48fcf94 React to changes in either width or height in iOS Metal renderer. by Anders Carlsson · 6 years ago
  9. 071edf3 Add missing files to AAR. by Sami Kalliomäki · 6 years ago
  10. 648a7ce Delete method EncodedFrame::GetBitstream, part 1 by Niels Möller · 6 years ago
  11. 0cc11b4 Android: Bump stack trace logging severity from debug to warning by Magnus Jedvert · 6 years ago
  12. 68478b8 Added user-defined predicate to filter video codec implementations. by Yura Yaroshevich · 6 years ago
  13. 7f7e973 Roll chromium_revision 82a8b043ef..28d6168850 (611537:611644) by chromium-webrtc-autoroll · 6 years ago
  14. de10eea AEC3: Fix ENR in the dominant nearend detection by Gustaf Ullberg · 6 years ago
  15. cf69d22 AEC3: Optimizing the Update method of the FilterAnalyzer class. by Jesús de Vicente Peña · 6 years ago
  16. 154a262 Don't clear self.videoFrame when setting up OpenGL in the EAGL video view. by Aaron Golden · 6 years ago
  17. 2ba65c3 Fix webrtc-internal ios json config by Artem Titarenko · 6 years ago
  18. ce1b140 Adding WinUWP compilation support to WebRTC. by Robin Raymond · 6 years ago
  19. 3bc24bc Roll chromium_revision b04e513f82..82a8b043ef (611432:611537) by chromium-webrtc-autoroll · 6 years ago
  20. 5ec2c94 Reland "Delay call to Destroy until after SignalDone has finished firing." by Zach Stein · 6 years ago
  21. 0d007d7 Roll chromium_revision aed902039c..b04e513f82 (611312:611432) by chromium-webrtc-autoroll · 6 years ago
  22. 72d2ddd Fix raddr on srflx and relay candidates by Jeroen de Borst · 6 years ago
  23. 889b587 Revert "Delay call to Destroy until after SignalDone has finished firing." by Qingsi Wang · 6 years ago
  24. ab20b3f Roll chromium_revision 81c26a093b..aed902039c (611047:611312) by chromium-webrtc-autoroll · 6 years ago
  25. c7b8da4 Delay call to Destroy until after SignalDone has finished firing. by Zach Stein · 6 years ago
  26. 14f252a AEC3: Add metrics for API call jitter by Per Åhgren · 6 years ago
  27. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  28. c7f1a0a Fix webrtc-internal configs to run perf tests on separate bots by Artem Titarenko · 6 years ago
  29. 8b55602 Batch RTC event log output if using the new wire format. by Bjorn Terelius · 6 years ago
  30. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  31. 59cfd35 Address vptr race condition while PeerConnection is destructed. by Yves Gerey · 6 years ago
  32. 09d6588 Change HdrMetadataExtension to ColorSpaceExtension by Johannes Kron · 6 years ago
  33. 53382cb Move RtcpStatistics from common_types.h to a new header file by Niels Möller · 6 years ago
  34. 6b3d181 Remove unused BWE field trial strings. by Bjorn Terelius · 6 years ago
  35. 65c921c Add setters to ColorSpace class by Johannes Kron · 6 years ago
  36. 196c5ba Specific pacing configuration. by Christoffer Rodbro · 6 years ago
  37. ba2840c Various VP9 high fps fixes by Ilya Nikolaevskiy · 6 years ago
  38. af52b68 Populate VideoSendTime extension network2 field when configured by Danil Chapovalov · 6 years ago
  39. 31a4331 Roll chromium_revision 208bb982f7..81c26a093b (610939:611047) by chromium-webrtc-autoroll · 6 years ago
  40. c5095e5 Fix wrong forward declaration namespace. by Mirko Bonadei · 6 years ago
  41. 4e58444 Roll chromium_revision 00f78b5b14..208bb982f7 (610831:610939) by chromium-webrtc-autoroll · 6 years ago
  42. 74cdf78 add cstring include need for strncmp by Michel Promonet · 6 years ago
  43. e38a5a1 Small cleanup to mediasession_unittest.cc by Steve Anton · 6 years ago
  44. a6e034a Rebase std::is_trivially_* with absl::is_trivially_* by Jiawei Ou · 6 years ago
  45. 622eeda Bump variable sizes in response to fuzzer bug by Jonas Olsson · 6 years ago
  46. b24c00f Add AudioProcessingCaptureStats and a level estimator replacement by Sam Zackrisson · 6 years ago
  47. 2918d4e Roll chromium_revision 7579fcbc1c..00f78b5b14 (610728:610831) by chromium-webrtc-autoroll · 6 years ago
  48. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  49. b253303 Add magjed as owner of rtc_tools. by Patrik Höglund · 6 years ago
  50. 856cf22 In ReceiveStatistics use monotonic clock instead of ntp clock by Danil Chapovalov · 6 years ago
  51. 22027b9 Add a new Task Queue for WinUWP. by Robin Raymond · 6 years ago
  52. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  53. 8ce0d2b In ReceiveStatistic require callbacks during construction by Danil Chapovalov · 6 years ago
  54. 4c0cc5b Reland Profile 2 to default profiles by Emircan Uysaler · 6 years ago
  55. f1c194d Roll chromium_revision d298cced6c..7579fcbc1c (610627:610728) by chromium-webrtc-autoroll · 6 years ago
  56. 05aee74 Roll chromium_revision f9be7d3d66..d298cced6c (610432:610627) by Oleh Prypin · 6 years ago
  57. 9289eda Revert "Replace the IceConnectionState implementation." by Alex Loiko · 6 years ago
  58. 4f00075 Remove use of CodecSpecificInfo.codec_name by Erik Språng · 6 years ago
  59. 1e87b4f Replace the IceConnectionState implementation. by Jonas Olsson · 6 years ago
  60. 57f3ad0 Adds stable bandwidth estimate to GoogCC. by Sebastian Jansson · 6 years ago
  61. 88ce4ef Don't buffer encoded frames. by Sergey Silkin · 6 years ago
  62. 885cf60 Moves ProbeBitrateEstimator from DelayBasedBwe. by Sebastian Jansson · 6 years ago
  63. e3abb81 Decouple //rtc_base:rtc_base_tests_utils from gunit. by Mirko Bonadei · 6 years ago
  64. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  65. b357e54 Add field trial config to disable pacer emergency stops. by Christoffer Rodbro · 6 years ago
  66. 6d254bc Delete unused method NetEq::PacketBufferStatistics by Niels Möller · 6 years ago
  67. 5f2ffee Clean up deprecated APM stats by Sam Zackrisson · 6 years ago
  68. f40150d Removing ANA enabling field trials. by Minyue Li · 6 years ago
  69. 2c977b4 Remove RSID from stream configs in new event log format. by Bjorn Terelius · 6 years ago
  70. 14dfe7f [GN] Fix dependency rebasing in BUILD.gn files. by Yves Gerey · 6 years ago
  71. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  72. 3890c1a Roll chromium_revision 1500c78c93..f9be7d3d66 (610314:610432) by chromium-webrtc-autoroll · 6 years ago
  73. 3955a50 Metal: Don't render into an empty view. by Peter Hanspers · 6 years ago
  74. 777cf26 AEC3: Clockdrift detection by Gustaf Ullberg · 6 years ago
  75. f259078 Use cropping aligning in video quality analysis tool by Magnus Jedvert · 6 years ago
  76. ebb50c2 Fix setting max reordering threshold in ReceiveStatistics by Danil Chapovalov · 6 years ago
  77. 286df00 Add tool for aligning cropped region of video files by Magnus Jedvert · 6 years ago
  78. 8e66863 Remove cricket::UdpTransport. by Mirko Bonadei · 6 years ago
  79. 94c9420 Remove cricket::BundleFilter. by Mirko Bonadei · 6 years ago
  80. eccfc47 Cleanup AimdRateController and remove RateControlRegion enum. by Bjorn Terelius · 6 years ago
  81. 42d2e4b Increase test timeouts in TCPChannelClientTest by Artem Titarenko · 6 years ago
  82. 00dfe93 Remove superfluous constructor from dltsTransport by Harald Alvestrand · 6 years ago
  83. 44727b4 Cleanup rtcp StreamStatistician::OnRtpPacket by Danil Chapovalov · 6 years ago
  84. af228ee Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN. by Alex Loiko · 6 years ago
  85. 5486bcd Remove SetChannelParameters function from API classes. by philipel · 6 years ago
  86. ecd6205 Disable GoogCcNetworkControllerTest.DetectsHighRateInSafeResetTrial by Alex Loiko · 6 years ago
  87. 8ac05cc Adds trial to use link capacity estimate in Opus encoder. by Sebastian Jansson · 6 years ago
  88. 2ff3f49 Move webrtc::CreatePeerConnectionFactory definition next to decl. by Mirko Bonadei · 6 years ago
  89. d51b355 Delete unused NetEq Rtcp stats. by Niels Möller · 6 years ago
  90. 7c36c71 Roll chromium_revision 6931f4c0d0..1500c78c93 (610209:610314) by chromium-webrtc-autoroll · 6 years ago
  91. 8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
  92. 4a2dd7a Roll chromium_revision 5825fead7b..6931f4c0d0 (610108:610209) by chromium-webrtc-autoroll · 6 years ago
  93. 540ef28 Adds OnReceivedUplinkAllocation method to AudioEncoder. by Sebastian Jansson · 6 years ago
  94. 6736df1 Moves BitrateAllocationUpdate to api. by Sebastian Jansson · 6 years ago
  95. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  96. e4cccae Removed ability to set CryptoOptions through PeerConnectionFactory from bindings. by Benjamin Wright · 6 years ago
  97. a526ae6 Roll chromium_revision 92f8c5b2a2..5825fead7b (609994:610108) by chromium-webrtc-autoroll · 6 years ago
  98. 5eae1d9 Remove legacy SetTargetTransferRateObserver by Piotr (Peter) Slatala · 6 years ago
  99. 37227be Add check for media transport and bundle policy by Piotr (Peter) Slatala · 6 years ago
  100. 47dfdca Create 'MaybeCreateMediaTransport' function by Piotr (Peter) Slatala · 6 years ago