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gerrit-public.fairphone.software / platform / external / webrtc / fdf74bdb5f8a20fdc2c5aa108e2f427e5da3141e
  1. fdf74bd Remove non implemented function from WebRtcVideoChannel. by philipel · 5 years ago
  2. 656590d Roll chromium_revision 778353d874..50acc956cd (674780:674882) by chromium-webrtc-autoroll · 5 years ago
  3. 1efb4a2 Add field trial for forcing partition resilience mode in libvpx. by Rasmus Brandt · 5 years ago
  4. 3fbf1e2 Reduce kMaxSimulcastStreams to 3 by Florent Castelli · 5 years ago
  5. 9d96209 Switch unpack_aecdump to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  6. 4580ca2 Reland: Add ability to set ssrcs of RtpSender at construction time by Erik Språng · 5 years ago
  7. 8e3cb53 Remove activity_metric tool. by Mirko Bonadei · 5 years ago
  8. 543179e Roll chromium_revision 79b588ee95..778353d874 (674667:674780) by chromium-webrtc-autoroll · 5 years ago
  9. 2250b05 Adding support for channel mixing between different channel layouts. by henrika · 5 years ago
  10. 3f2eeb8 Adding test on GetSpanSamples() for NetEq PacketBuffer. by Minyue Li · 5 years ago
  11. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  12. e8fbc5d Refactor WebRtcOpus_PacketHasFec. by Minyue Li · 5 years ago
  13. 67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 5 years ago
  14. c6c730b Roll chromium_revision 54ec0ffb89..79b588ee95 (674538:674667) by chromium-webrtc-autoroll · 5 years ago
  15. 697f861 Roll chromium_revision 13b7238371..54ec0ffb89 (674397:674538) by chromium-webrtc-autoroll · 5 years ago
  16. 2e60217 Add speculative checks to RtpPacketHistory by Erik Språng · 5 years ago
  17. 46dda83 Improve buffer level estimation with DTX and add CNG time stretching. by Jakob Ivarsson · 5 years ago
  18. 3d642f8 Rename ..BitrateThresholds to ..BitrateLimits. by Sergey Silkin · 5 years ago
  19. cecee99 Disable VP9 2nd profile test for ios arm64 by Artem Titov · 5 years ago
  20. 51f599b Make rtc_base/base_java public. by Sami Kalliomäki · 5 years ago
  21. 81e1bf0 Remove using DegradationPreference from scenario_config.h by Artem Titov · 5 years ago
  22. 6542826 Add new tests with lossy networks on PC test framework by Artem Titov · 5 years ago
  23. cd8a6e2 Add writing and parsing of the `abs-capture-time` RTP header extension. by Chen Xing · 5 years ago
  24. 53d45ba Make TaskQueueFactory required construction parameter for Call by Danil Chapovalov · 5 years ago
  25. 84ce3c0 Macro rename s/CS_DEBUG_CHECKS/RTC_CS_DEBUG_CHECKS. by Mirko Bonadei · 5 years ago
  26. a2b30d8 Add functions to read from/write to bitstream values with known max value by Danil Chapovalov · 5 years ago
  27. 9eee121 Switch py_quality_assessment to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  28. b60141b Save and serialize the receive RIDs in MediaContentDescription by Florent Castelli · 5 years ago
  29. e8ed830 WebRtcVideoChannel encoder fallback. by philipel · 5 years ago
  30. e420c6a Add missing include for memcpy/memcmp by Artem Titov · 5 years ago
  31. 6a2c1ba Roll chromium_revision ba17fd6b36..13b7238371 (674288:674397) by chromium-webrtc-autoroll · 5 years ago
  32. 8fa7151 Replace the implementation of `GetContributingSources()` on the audio side. by Chen Xing · 5 years ago
  33. 16661eb Fix: report video_bwe_stats as bytes per second, as specified in the unit by Artem Titov · 5 years ago
  34. 443b7ee Destroy existing encoder instance before creating a new one. by Sergey Silkin · 5 years ago
  35. 2c5af4f Add * and / operator into SamplesStatsCounter. by Artem Titov · 5 years ago
  36. 1d46f9c Add RtpPacket::IsExtensionReserved(). by Erik Språng · 5 years ago
  37. 6038926 Roll chromium_revision 35be8751a4..ba17fd6b36 (674036:674288) by chromium-webrtc-autoroll · 5 years ago
  38. 02d7d35 Revert "Add ability to set ssrcs of RtpSender at construction time" by Amit Hilbuch · 5 years ago
  39. c442197 Check the rid direction matches the direction in simulcast description by Florent Castelli · 5 years ago
  40. 238aab9 Fix bug in use_datagram_transport configuration. by Bjorn A Mellem · 5 years ago
  41. b073f1c Only set the RtcEventLog for media transport when it's used for media. by Bjorn A Mellem · 5 years ago
  42. 73bfc0e Roll chromium_revision 6f0434662d..35be8751a4 (673926:674036) by chromium-webrtc-autoroll · 5 years ago
  43. ed56cf4 Remove deprecated version of Vp8FrameBufferControllerFactory::Create by Elad Alon · 5 years ago
  44. e9d6e65 Add ability to set ssrcs of RtpSender at construction time by Erik Språng · 5 years ago
  45. 5ee6967 Don't reset encoder on max/min bitrate change. by Sergey Silkin · 5 years ago
  46. bc70b61 Switch rnn_vad_tool to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  47. f1a7bb1 Stop using unnecessary gclient vars by Oleh Prypin · 5 years ago
  48. 45befc5 Pass FecControllerOverride to Vp8FrameBufferControllerFactory::Create by Elad Alon · 5 years ago
  49. 14be799 Switch neteq tools to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  50. e731a2e Remove check on supported profile in favor of expilict disabling by Artem Titov · 5 years ago
  51. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  52. 419aae2 Remove android_tools deps by Yun Liu · 5 years ago
  53. ebdf9f8 Roll chromium_revision 097ffaa18d..6f0434662d (673789:673926) by chromium-webrtc-autoroll · 5 years ago
  54. 6fdfec1 Add overload to CreateIceTransport that takes additional dependencies by Steve Anton · 5 years ago
  55. 1cf9470 Roll chromium_revision 05067e74f0..097ffaa18d (673689:673789) by chromium-webrtc-autoroll · 5 years ago
  56. 5985a04 Add a field trial to control datagram transport use. by Bjorn A Mellem · 5 years ago
  57. 3e8ef94 Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. by Chen Xing · 5 years ago
  58. 62eb89d Fixing possible overflow in NetEq buffle level filter. by Minyue Li · 5 years ago
  59. 5983087 Forced vp8 sw encoder fallback: only use min bitrate config if codec type is vp8. by Åsa Persson · 5 years ago
  60. 5b2ce12 Roll chromium_revision ce6d12c81b..05067e74f0 (673457:673689) by chromium-webrtc-autoroll · 5 years ago
  61. ea95c37 Report freeze_time_ms from PC test framework by Artem Titov · 5 years ago
  62. 754c952 Don't do ToI420() for each frame while checking is it dummy by Artem Titov · 5 years ago
  63. 6d9f001 Fix regression in PC quality test. by Artem Titov · 5 years ago
  64. a6cb150 Use Default instead of GlobalTaskQueueFactory to create AudioDeviceBuffer for ios by Danil Chapovalov · 5 years ago
  65. 3e3a6e5 Remove obsolete deps by Artem Titov · 5 years ago
  66. 896f4b6 Use Default instead of GlobalTaskQueueFactory to create AudioDeviceBuffer for android by Danil Chapovalov · 5 years ago
  67. a63aede Make VideoStreamEncoderInterface::SetFecControllerOverride pure virtual by Elad Alon · 5 years ago
  68. 65764e4 Add missing overrides in VideoEncoder proxies/adapters by Elad Alon · 5 years ago
  69. 2ce1da5 Roll chromium_revision dfe5e91525..ce6d12c81b (673350:673457) by chromium-webrtc-autoroll · 5 years ago
  70. 1e00dbc Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly by Min Wang · 5 years ago
  71. 67daf71 Implement RtpVideoSender::SetFecAllowed() by Elad Alon · 5 years ago
  72. 099b02a Get rid of deprecated version of NackSender::SendNack [2/2] by Elad Alon · 5 years ago
  73. 7e00c67 Pass FecControllerOverride to Vp8FrameBufferController by Elad Alon · 5 years ago
  74. 22896d4 Roll chromium_revision 27c2f87cf5..dfe5e91525 (672459:673350) by chromium-webrtc-autoroll · 5 years ago
  75. 8f01c4e Define FecControllerOverride and plumb it down to VideoEncoder by Elad Alon · 5 years ago
  76. 52e5242 Add trait to Build/Parse DependencyDescriptor rtp header extension by Danil Chapovalov · 5 years ago
  77. 225842c Initialize signal processing function pointers statically by Karl Wiberg · 5 years ago
  78. a47ba41 Get rid of deprecated version of NackSender::SendNack [1/2] by Elad Alon · 5 years ago
  79. a094849 RateControlSettings: add option to set min pixels per frame for libvpx vp8. by Åsa Persson · 5 years ago
  80. 60bfb3d NetEQ: BackgroundNoise::Update returns true when the filter is updated by Alessio Bazzica · 5 years ago
  81. 825aad1 Delete almost all includes of platform_file.h by Niels Möller · 5 years ago
  82. 767efab Delete method ReadableWav::Eof, which was used incorrectly. by Niels Möller · 5 years ago
  83. 71809c6 WindowCapturerWin: properly check return value of GetClassName by Julien Isorce · 5 years ago
  84. 9407776 Temporarily suppress -Wdeprecated-declarations to update jsoncpp. by Mirko Bonadei · 5 years ago
  85. dd4dc7a Adds additional fields to NetworkStateEstimate. by Sebastian Jansson · 5 years ago
  86. 49167de Adds interface for remote network estimates to NetworkControllerInterface. by Sebastian Jansson · 5 years ago
  87. 2efae77 Add RTCStats for keyFramesEncoded, keyFramesDecoded. by Rasmus Brandt · 5 years ago
  88. 478cb46 Add GeneratePadding method to replace TimeToSendPadding by Erik Språng · 5 years ago
  89. c2f5686 Extend structures to store updated version of the dependency descriptor by Danil Chapovalov · 5 years ago
  90. a3f3ab9 Remove Simple Command Line Parser. by Mirko Bonadei · 5 years ago
  91. 4ba04b7 Delete RtcEventLogFactory factory as now unused by Danil Chapovalov · 5 years ago
  92. 36c8ef6 Cleanup Ulpfec receiver: remove 2 blocks RED packets support by Ilya Nikolaevskiy · 5 years ago
  93. a36c591 Reland "Reland "Change buffer level filter to store current level in number of samples."" by Jakob Ivarsson · 5 years ago
  94. b93af85 Revert "Reland "Change buffer level filter to store current level in number of samples."" by Jakob Ivarsson · 5 years ago
  95. 2d821c3 Allow VideoTimingExtension to be used with FEC by Ilya Nikolaevskiy · 5 years ago
  96. e4ac723 Delete deprecated version of PeerConnectionFactoryInterface::StartAecDump by Niels Möller · 5 years ago
  97. bca1485 Enable setting surface_ice_candidates_on_ice_transport_type_changed on the fly. by Qingsi Wang · 5 years ago
  98. 0ded32d Reland "Change buffer level filter to store current level in number of samples." by Jakob Ivarsson · 5 years ago
  99. 4d69516 Don't use angle-bracket #includes for WebRTC's own files by Oleh Prypin · 5 years ago
  100. c57b0ee Fix for NACK retransmission in Scenario tests. by Sebastian Jansson · 5 years ago