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gerrit-public.fairphone.software / platform / external / webrtc / 34029e209d6d116c338e7d65d6980cc395184658 / call / call_perf_tests.cc
  1. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  2. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  3. 5aea38c Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}. by Alex Loiko · 7 years ago
  4. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  5. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  6. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  7. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  8. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/call_perf_tests.cc]
  9. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  10. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  11. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  12. d9827b5 Removed an unused variable from CallPerfTest::TestAudioVideoSync() by eladalon · 7 years ago
  13. cc3d442 Rename ViEEncoder to VideoStreamEncoder by mflodman · 7 years ago
  14. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  15. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  16. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  17. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  18. c467520 Delete helper class MediaTypePacketReceiver. by nisse · 7 years ago
  19. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  20. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  21. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 7 years ago
  22. 4fb651d Event log cleanup in tests. by philipel · 7 years ago
  23. 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
  24. 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
  25. 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
  26. 0ffdcc5 Delete unneeded includes of deprecated system_wrappers include files. by nisse · 7 years ago
  27. 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
  28. 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
  29. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 7 years ago
  30. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 7 years ago
  31. 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
  32. 72acf25 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 7 years ago
  33. baded15 Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2751063005/ ) by ilnik · 7 years ago
  34. 2a420ce Revert of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #11 id:300001 of https://codereview.webrtc.org/2750473002/ ) by ilnik · 7 years ago
  35. 2549ad4 Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2748643002/ ) by ilnik · 7 years ago
  36. a514584 Add the ability to read/write to WAV files in FakeAudioDevice by oprypin · 7 years ago
  37. 382a72a Revert of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #2 id:90001 of https://codereview.webrtc.org/2744003002/ ) by ilnik · 7 years ago
  38. 8c0a589 Reland of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2743993002/ ) by ilnik · 7 years ago
  39. 1cb27c2 Revert of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #2 id:70001 of https://codereview.webrtc.org/2745583006/ ) by ilnik · 7 years ago
  40. b007425 Reland of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper. by ilnik · 7 years ago
  41. e5d3a3e Fix quick perf test setting that was accidentally inverted. by sprang · 7 years ago
  42. c1b57a1 Test field trial group with startswith rather than equals. by sprang · 7 years ago
  43. fa5fdce Reland of Set scaling limit at 320 * 180 for all implementations. (patchset #1 id:1 of https://codereview.webrtc.org/2711913007/ ) by kthelgason · 7 years ago
  44. 5328b9e added WebRTC-QuickPerfTest to RampUpTests and CallPerfTests by ilnik · 7 years ago
  45. 84a3759 Change rtc::VideoSinkWants to have target and a max pixel count by sprang · 7 years ago
  46. f9b6e5e Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations. by Stefan Holmer · 7 years ago
  47. ac61b74 Refactor FakeAudioDevice to have separate methods for starting recording and playout. by perkj · 7 years ago
  48. fe50b4d Make class of static functions in rtp_to_ntp.h: - UpdateRtcpList - RtpToNtp by asapersson · 8 years ago
  49. ca87b62 Disable failing perf test on Android. by kthelgason · 8 years ago
  50. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  51. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  52. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  53. 08127a9 Reland #2 of Issue 2434073003: Extract bitrate allocation ... by Erik Språng · 8 years ago
  54. 841de6a Add FlexFEC to CallTest. by brandtr · 8 years ago
  55. 1369c83 Revert of Issue 2434073003: Extract bitrate allocation ... (patchset #4 id:60001 of https://codereview.webrtc.org/2488833004/ ) by sprang · 8 years ago
  56. 647bf43 Reland of Issue 2434073003: Extract bitrate allocation ... by sprang · 8 years ago
  57. bf6a45b Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency by charujain · 8 years ago
  58. 3dc929e Replace RTCPUtility RtcpParser with Test RtcpParser making code cleaner by danilchap · 8 years ago
  59. 803d97f Let ViEEncoder express resolution requests as Sinkwants. by perkj · 8 years ago
  60. e566ac7 Remove voe::Channel::StopReceive() and associated logic. by solenberg · 8 years ago
  61. 21d45d2 Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. by Per · 8 years ago
  62. 05a55b5 Revert of Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #2 id:20001 of https://codereview.webrtc.org/2455963004/ ) by emircan · 8 years ago
  63. 5f1b051 Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. by perkj · 8 years ago
  64. 68e6bdd Remove use of VoECodec in video/call tests. by solenberg · 8 years ago
  65. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  66. b5f2c3f Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
  67. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  68. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  69. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  70. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  71. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  72. 88499ec Moving/renaming webrtc/common.h. by solenberg · 8 years ago
  73. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 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. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  76. 371b43b Changes synchronization offset perfomance tracking by Danil Chapovalov · 8 years ago
  77. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  78. 46b89b9 Collapse most stdout spammy output of webrtc_perf_tests with PrintResultList by danilchap · 8 years ago
  79. 01d70a3 Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h. by asapersson · 8 years ago
  80. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 8 years ago
  81. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 8 years ago
  82. 7ade7b3 Delete class webrtc::VideoRenderer and its header file. by nisse · 8 years ago
  83. eb83a1a This is an initial cleanup step, aiming to delete the by nisse · 8 years ago
  84. f8cdd18 Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs" by asapersson · 8 years ago
  85. b25345e Replace scoped_ptr with unique_ptr in webrtc/call/ by kwiberg · 8 years ago
  86. 905f8e7 Make ReconfigureVideoEncoder void. by Peter Boström · 8 years ago
  87. ac287ee VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock. by danilchap · 8 years ago
  88. cde5d6b removed five redundant avsync tests to make webrtc_perf_test faster by Danil Chapovalov · 8 years ago
  89. 9c6a0c7 Added A/V sync tests with drifting clocks. by danilchap · 8 years ago
  90. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  91. 7b971e7 Remove extra_options from VideoCodec. by Peter Boström · 9 years ago
  92. ea8c0f6 Fix capture ntp time issue introduced with r11187. by Stefan Holmer · 9 years ago
  93. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  94. 9fea80f Add audio streams to CallTest and a first A/V call test. by Stefan Holmer · 9 years ago
  95. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago
  96. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  97. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  98. 3e6db23 audio_coding: remove "main" directory by kjellander · 9 years ago
  99. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  100. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago