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gerrit-public.fairphone.software / platform / external / webrtc / ea1bb35e2761159f00089f707ed845e2e0b4e03b / video / video_send_stream_tests.cc
  1. ea1bb35 Cleanup networkroute.h by Steve Anton · 7 years ago
  2. 7258224 Replaces call config create in tests with modify. by Sebastian Jansson · 7 years ago
  3. f33905d Makes some CallTest members private. by Sebastian Jansson · 7 years ago
  4. 8e6602f Separates send and receive event log in CallTest. by Sebastian Jansson · 7 years ago
  5. 798ee75 Always disable RED when ULPFEC is disabled. by Kári Tristan Helgason · 7 years ago
  6. 5e898d6 Stop using VideoCodec.targetBitrate for vp8 screenshare config by Erik Språng · 7 years ago
  7. 5ab67a5 Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. by philipel · 7 years ago
  8. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
  9. cb96ad8 Add ParsedPayload::video_header() accessor. by philipel · 7 years ago
  10. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  11. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  12. 520ca4e Delete enum RtpVideoCodecTypes, replaced with VideoCodecType. by Niels Möller · 7 years ago
  13. be71a1e Replace VP9 screen sharing. by Sergey Silkin · 7 years ago
  14. 49fcc10 Merge DegradationPreference enums. by Taylor Brandstetter · 7 years ago
  15. 377ef24 Remove extra reference from GOF. by Sergey Silkin · 7 years ago
  16. 566124a Move BitrateAllocation to api/ and rename it VideoBitrateAllocation by Erik Språng · 7 years ago
  17. 4db138e Reland "Move creating encoder to VideoStreamEncoder." by Niels Möller · 7 years ago
  18. 0d650b4 Revert "Move creating encoder to VideoStreamEncoder." by Niels Moller · 7 years ago
  19. fb82fcc Move creating encoder to VideoStreamEncoder. by Niels Möller · 7 years ago
  20. 259a497 Reland "Reland "Move rtp-specific config out of EncoderSettings."" by Niels Möller · 7 years ago
  21. 571e6c9 Fix rate allocation between temporal layers in SVC. by Sergey Silkin · 7 years ago
  22. 6c2c13a Revert "Reland "Move rtp-specific config out of EncoderSettings."" by Niels Möller · 7 years ago
  23. 8668496 Adding layering configurator and rate allocator for VP9 SVC. by Sergey Silkin · 7 years ago
  24. 04dd176 Reland "Move rtp-specific config out of EncoderSettings." by Niels Möller · 7 years ago
  25. 92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 7 years ago
  26. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 7 years ago
  27. 82fad3d Remove TemporalLayersFactory and associated classes by Erik Språng · 7 years ago
  28. 4425b05 Add video send stream test to check switch to and from screenshare by Ilya Nikolaevskiy · 7 years ago
  29. efbb978 Fix flacky VideoSendStreamTest.SupportsVideoContentType by Ilya Nikolaevskiy · 7 years ago
  30. a796a7e Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." by Sergey Silkin · 7 years ago
  31. d3f3816 Testing multiple retransmission in video nack test. by Sebastian Jansson · 7 years ago
  32. e27e0ac Revert "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." by Philip Eliasson · 7 years ago
  33. d2ed0a4 Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers. by Sergey Silkin · 7 years ago
  34. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  35. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  36. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  37. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  38. 45d725d Support sending flexfec and simulcast together. by Danil Chapovalov · 7 years ago
  39. cc7125f Sets sending status for active RtpRtcp modules. by Seth Hampson · 8 years ago
  40. 56fa050 Improved accuracy of packet loss calculation in tests. by Sebastian Jansson · 8 years ago
  41. 3587b83 Make RTCP report interval configurable by Jiawei Ou · 8 years ago
  42. a45c8da Removed GetPacingFactor from PacedSender. by Sebastian Jansson · 8 years ago
  43. cabe383 Moved ALR experiment settings to new experiments folder. by Sebastian Jansson · 8 years ago
  44. 8e07c13 Optional: Use nullopt and implicit construction in /video by Oskar Sundbom · 8 years ago
  45. 9299642 Make ALR probing experiment default on. by Erik Språng · 8 years ago
  46. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 8 years ago
  47. 3f2634e Reland "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices." by Sergey Silkin · 8 years ago
  48. ae29428 Revert "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices." by Ivo Creusen · 8 years ago
  49. 47836b4 Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices. by Sergey Silkin · 8 years ago
  50. 7c8cca3 Add check for send-side bwe before applying alr settings by Erik Språng · 8 years ago
  51. d9f99c1 Replace Atomic32 with std::atomic in video/ by Yuwei Huang · 8 years ago
  52. f74d641 Simplify setting/unsetting REMB in RtcpSender by Danil Chapovalov · 8 years ago
  53. 51e21aa Simplify RtpRtcp interface for REMB by Danil Chapovalov · 8 years ago
  54. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 8 years ago
  55. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  56. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/video/video_send_stream_tests.cc]
  57. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 8 years ago
  58. 6d5b4d6 Piggybacking simulcast id and ALR experiment id into video content type extension. by ilnik · 8 years ago
  59. 5212700 Removing dependencies on stub headers within WebRTC. by mbonadei · 8 years ago
  60. 60e10c7 Add experiment to disable ulpfec. by stefan · 8 years ago
  61. 41cadbc Remove WebRTC-videocontenttypeextension field trial completely by ilnik · 8 years ago
  62. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 8 years ago
  63. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 8 years ago
  64. cc3d442 Rename ViEEncoder to VideoStreamEncoder by mflodman · 8 years ago
  65. 8a1d2a3 Remove NullReceiveStatistics by danilchap · 8 years ago
  66. 901b2df Simplify FakeReceiveStatistics in video send stream tests by danilchap · 8 years ago
  67. e5c4a81 Move RTP keep-alive config from VideoSendStream::Config to Call::Config by sprang · 8 years ago
  68. d2702ef Fix flaky test VideoSendStreamTest.SendsKeepAlive by sprang · 8 years ago
  69. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  70. 168794c Implement RTP keepalive in native stack. by sprang · 8 years ago
  71. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  72. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  73. 4847ae6 Reland of Periodically update codec bit/frame rate settings. by sprang · 8 years ago
  74. 1089499 Fix timing frames and FEC conflict by ilnik · 8 years ago
  75. 04f4d12 Implement timing frames. by ilnik · 8 years ago
  76. 90e3190 Update webrtc/test to new VideoFrameBuffer interface by Magnus Jedvert · 8 years ago
  77. 317005a Revert of Periodically update codec bit/frame rate settings. (patchset #2 id:160001 of https://codereview.webrtc.org/2924023002/ ) by sprang · 8 years ago
  78. cf705c5 Reland of Protect new header extension by field trial experiment to allow hardcoding it in SDP (patchset #1 id:1 of https://codereview.webrtc.org/2922723002/ ) by ilnik · 8 years ago
  79. cdafeda Reland of Periodically update codec bit/frame rate settings. (patchset #1 id:1 of https://codereview.webrtc.org/2923993002/ ) by sprang · 8 years ago
  80. 5390c48 Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ ) by sprang · 8 years ago
  81. 6431e21 Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended. by sprang · 8 years ago
  82. ed9b9ff Revert of Protect new header extension by field trial experiment to allow hardcoding it in SDP (patchset #3 id:40001 of https://codereview.webrtc.org/2922683002/ ) by ilnik · 8 years ago
  83. cafa1d6 Protect new header extension by field trial experiment to allow hardcoding it in SDP by ilnik · 8 years ago
  84. 7a3006b Fix packetization logic to leave space for extensions in the last packet by ilnik · 8 years ago
  85. ea3a798 Move webrtc/video_frame to common_video/include. by nisse · 8 years ago
  86. f93752a Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ ) by nisse · 8 years ago
  87. 61b22dd Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ ) by nisse · 8 years ago
  88. 3870a07 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854883002/ ) by nisse · 8 years ago
  89. 6e6a485 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2852303002/ ) by nisse · 8 years ago
  90. d71ebd7 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2845333002/ ) by nisse · 8 years ago
  91. aec49d2 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ ) by nisse · 8 years ago
  92. 713a3bb Delete deprecated and transitional stuff related to video frame refactoring. by nisse · 8 years ago
  93. 00d802b Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ ) by ilnik · 8 years ago
  94. 27c46e2 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ ) by ilnik · 8 years ago
  95. 774f6b4 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
  96. 29dbb19 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ ) by ilnik · 8 years ago
  97. 4fa0c4f Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
  98. 5721866 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ ) by ilnik · 8 years ago
  99. 64e739a Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 8 years ago
  100. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago