Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 0e96535be97916d8fcaa9873ffab3c636539f9d8 / call / bitrate_estimator_tests.cc
  1. 33f9d2b Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class by Artem Titov · 4 years, 8 months ago
  2. d15a028 Hide deprecated SingleThreadedTaskQueueForTest behind an accessor by Danil Chapovalov · 4 years, 9 months ago
  3. 82a3f0a Replace SingleThreadedTaskQueueForTesting::SendTask usage with ::webrtc::SendTask by Danil Chapovalov · 4 years, 9 months ago
  4. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 4 years, 10 months ago
  5. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 4 years, 11 months ago
  6. 81687b3 Use explicit TaskQueueFactory for FrameGeneratorCapturer in BitrateEstimatorTest. by Danil Chapovalov · 5 years ago
  7. 41f9f2c ClangTidy fixes for call/ by Benjamin Wright · 5 years ago
  8. 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
  9. 8eeccbe Delete Start and Stop methods from TestVideoCapturer. by Niels Möller · 6 years ago
  10. c2ebe21 Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" by Jiawei Ou · 6 years ago
  11. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  12. 59844ce Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`." by Qingsi Wang · 6 years ago
  13. be14217 Use the factory instead of using the builtin code path in `VideoCodecInitializer`. by Jiawei Ou · 6 years ago
  14. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  15. cbcbc22 Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Niels Möller · 6 years ago
  16. 377b26e Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Sebastian Jansson · 6 years ago
  17. efb94d5 Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."" by Oleh Prypin · 6 years ago
  18. 7961dc2 Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Niels Moller · 6 years ago
  19. 529d0d9 Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config. by Niels Möller · 6 years ago
  20. cb7e1d2 Use SdpVideoFormat in VideoReceiveStream::Decoder by Niels Möller · 6 years ago
  21. dd2eebe Deprecate two DirectTransport ctors and remove their direct usage. by Artem Titov · 6 years ago
  22. f33905d Makes some CallTest members private. by Sebastian Jansson · 6 years ago
  23. 8e6602f Separates send and receive event log in CallTest. by Sebastian Jansson · 6 years ago
  24. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  25. f8d8d6d Use range-based-for instead of std::for_each and std::mem_fun by Yusuke Suzuki · 6 years ago
  26. 49fcc10 Merge DegradationPreference enums. by Taylor Brandstetter · 6 years ago
  27. 4db138e Reland "Move creating encoder to VideoStreamEncoder." by Niels Möller · 6 years ago
  28. 0d650b4 Revert "Move creating encoder to VideoStreamEncoder." by Niels Moller · 6 years ago
  29. fb82fcc Move creating encoder to VideoStreamEncoder. by Niels Möller · 6 years ago
  30. 259a497 Reland "Reland "Move rtp-specific config out of EncoderSettings."" by Niels Möller · 6 years ago
  31. 6c2c13a Revert "Reland "Move rtp-specific config out of EncoderSettings."" by Niels Möller · 6 years ago
  32. 04dd176 Reland "Move rtp-specific config out of EncoderSettings." by Niels Möller · 6 years ago
  33. 92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 6 years ago
  34. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 6 years ago
  35. 03e6ec9 Reland "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 6 years ago
  36. 081136f Revert "Reland "Add multiplex case to webrtc_perf_tests"" by Taylor Brandstetter · 6 years ago
  37. 7c5bc1c Reland "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 6 years ago
  38. 5aac372 Revert "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 6 years ago
  39. d90a7e8 Add multiplex case to webrtc_perf_tests by Emircan Uysaler · 6 years ago
  40. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  41. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/bitrate_estimator_tests.cc]
  42. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  43. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  44. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  45. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  46. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  47. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 7 years ago
  48. 4fb651d Event log cleanup in tests. by philipel · 7 years ago
  49. c5d62e2 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) by sprang · 7 years ago
  50. f9ed235 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) by lliuu · 7 years ago
  51. 3ea3c77 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) by sprang · 7 years ago
  52. 0ffdcc5 Delete unneeded includes of deprecated system_wrappers include files. by nisse · 7 years ago
  53. e5ad5ca Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) by nisse · 7 years ago
  54. 3a3bd50 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) by lliuu · 7 years ago
  55. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 7 years ago
  56. 8b45b11 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) by skvlad · 7 years ago
  57. 72acf25 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 7 years ago
  58. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  59. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  60. 5d78e8d Remove audio from BitrateEstimatorTest. by aleloi · 8 years ago
  61. 803d97f Let ViEEncoder express resolution requests as Sinkwants. by perkj · 8 years ago
  62. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  63. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  64. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  65. 55d932b Add logging statements to places where the frame might be dropped in WebRTC pipeline. by sakal · 8 years ago
  66. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  67. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  68. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  69. a49cbd3 Replace VideoCapturerInput with VideoSinkInterface. by perkj · 8 years ago
  70. 9fdbda6 Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ ) by perkj · 8 years ago
  71. 95a226f Replace VideoCapturerInput with VideoSinkInterface. by perkj · 8 years ago
  72. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  73. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  74. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  75. 7522a28 Removed old probe cluster logic and logic related to ssrcs from DelayBasedBwe. by philipel · 8 years ago
  76. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  77. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  78. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 8 years ago
  79. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  80. 1086ed6 Disable SwitchesToASTThenBackToTOFForVideo test completely. by deadbeef · 8 years ago
  81. 844f993 Disabling SwitchesToASTThenBackToTOFForVideo test for MSan bot. by deadbeef · 8 years ago
  82. 4aa438c Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo. by minyuel · 8 years ago
  83. b25345e Replace scoped_ptr with unique_ptr in webrtc/call/ by kwiberg · 8 years ago
  84. 789ba92 Simplify CongestionController. by Stefan Holmer · 8 years ago
  85. 8c66a00 Initialize VideoSendStream members in constructor. by Peter Boström · 8 years ago
  86. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 8 years ago
  87. 9fea80f Add audio streams to CallTest and a first A/V call test. by Stefan Holmer · 9 years ago
  88. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago
  89. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  90. 7c704b8 Use webrtc/base/logging.h in stefan@'s ownership. by Peter Boström · 9 years ago
  91. 521af4e Remove duplicate decoders in BitrateEstimatorTest. by Peter Boström · 9 years ago
  92. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  93. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  94. 0ccae13 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  95. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  96. f116bd0 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  97. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  98. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  99. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  100. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago[Renamed from webrtc/video/bitrate_estimator_tests.cc]