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gerrit-public.fairphone.software / platform / external / webrtc / 22660f34a1fd7150f624f2212d7fb33ad7680d19 / call / call_perf_tests.cc
  1. 16cb8f5 Reland "Replace usage of old SetRates/SetRateAllocation methods" by Erik Språng · 6 years ago
  2. 7ddef1a Revert "Replace usage of old SetRates/SetRateAllocation methods" by Minyue Li · 6 years ago
  3. 7ac0d5f Replace usage of old SetRates/SetRateAllocation methods by Erik Språng · 6 years ago
  4. ded1e4f Disable flaky call_perf tests for iOS devices by Artem Titarenko · 6 years ago
  5. 41f9f2c ClangTidy fixes for call/ by Benjamin Wright · 6 years ago
  6. d8d3248 Reland "Delete test/constants.h" by Elad Alon · 6 years ago
  7. 0500b52 Reduce webrtc_perf_tests duration on buildbots by Ilya Nikolaevskiy · 6 years ago
  8. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  9. af228ee Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN. by Alex Loiko · 6 years ago
  10. de8e6e6 Refactor bitrate configuration in CallTest by Niels Möller · 6 years ago
  11. c2ebe21 Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" by Jiawei Ou · 6 years ago
  12. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  13. 59844ce Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`." by Qingsi Wang · 6 years ago
  14. be14217 Use the factory instead of using the builtin code path in `VideoCodecInitializer`. by Jiawei Ou · 6 years ago
  15. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  16. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  17. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  18. cbcbc22 Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Niels Möller · 6 years ago
  19. 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
  20. 377b26e Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Sebastian Jansson · 6 years ago
  21. efb94d5 Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."" by Oleh Prypin · 6 years ago
  22. 7961dc2 Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Niels Moller · 6 years ago
  23. 529d0d9 Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config. by Niels Möller · 6 years ago
  24. 8c1bf95 Reland "Add initial support for RtpEncodingParameters max_framerate." by Åsa Persson · 6 years ago
  25. 631cafa Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe by Artem Titov · 6 years ago
  26. 4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
  27. 46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
  28. 948b7e3 Revert "Add initial support for RtpEncodingParameters max_framerate." by Mirko Bonadei · 6 years ago
  29. ced5cfd Add initial support for RtpEncodingParameters max_framerate. by Åsa Persson · 6 years ago
  30. 9a75061 Add test CallPerfTest.PlaysOutAudioAndVideoInSyncWithoutClockDrift by Niels Möller · 6 years ago
  31. 7258224 Replaces call config create in tests with modify. by Sebastian Jansson · 6 years ago
  32. f33905d Makes some CallTest members private. by Sebastian Jansson · 6 years ago
  33. 8e6602f Separates send and receive event log in CallTest. by Sebastian Jansson · 6 years ago
  34. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
  35. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  36. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  37. 0a8f435 Move VideoEncoderConfig from call/ to api/. by Niels Möller · 7 years ago
  38. ef75ebe Directly include VideoBitrateAllocation in call/ targets by Erik Språng · 7 years ago
  39. 566124a Move BitrateAllocation to api/ and rename it VideoBitrateAllocation by Erik Språng · 7 years ago
  40. 4db138e Reland "Move creating encoder to VideoStreamEncoder." by Niels Möller · 7 years ago
  41. 0d650b4 Revert "Move creating encoder to VideoStreamEncoder." by Niels Moller · 7 years ago
  42. 0676f22 Probe on video encoder reconfiguration test. by philipel · 7 years ago
  43. fb82fcc Move creating encoder to VideoStreamEncoder. by Niels Möller · 7 years ago
  44. 2784a03 Add audio_ prefix to audio-related members of CallTest. by Niels Möller · 7 years ago
  45. 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
  46. 6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 7 years ago
  47. 8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 7 years ago
  48. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  49. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  50. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  51. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  52. 85904f4 Disabling a couple flaky CallPerfTests on Mac. by Taylor Brandstetter · 7 years ago
  53. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  54. fedc00c Optional: Use nullopt and implicit construction in /call by Oskar Sundbom · 7 years ago
  55. 62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
  56. 7f331fa Add metric name for MinVideoAndAudioBitRate. by Edward Lemur · 7 years ago
  57. 947f3fe Fix reporting of perf results on PlaysOutAudioAndVideoInSync* tests by Edward Lemur · 7 years ago
  58. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  59. 2f06168 Make PrintResultList receive a vector of doubles instead of a string. by Edward Lemur · 7 years ago
  60. d0e196b Adding two tests: by Alex Narest · 7 years ago
  61. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  62. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  63. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  64. 5aea38c Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}. by Alex Loiko · 7 years ago
  65. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  66. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  67. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  68. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  69. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/call_perf_tests.cc]
  70. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  71. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  72. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  73. d9827b5 Removed an unused variable from CallPerfTest::TestAudioVideoSync() by eladalon · 7 years ago
  74. cc3d442 Rename ViEEncoder to VideoStreamEncoder by mflodman · 7 years ago
  75. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  76. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  77. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  78. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  79. c467520 Delete helper class MediaTypePacketReceiver. by nisse · 8 years ago
  80. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  81. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  82. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
  83. 4fb651d Event log cleanup in tests. by philipel · 8 years ago
  84. 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 · 8 years ago
  85. 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 · 8 years ago
  86. 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 · 8 years ago
  87. 0ffdcc5 Delete unneeded includes of deprecated system_wrappers include files. by nisse · 8 years ago
  88. e5ad5ca Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) by nisse · 8 years ago
  89. 3a3bd50 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) by lliuu · 8 years ago
  90. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 8 years ago
  91. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  92. 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 · 8 years ago
  93. 72acf25 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 8 years ago
  94. 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 · 8 years ago
  95. 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 · 8 years ago
  96. 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 · 8 years ago
  97. a514584 Add the ability to read/write to WAV files in FakeAudioDevice by oprypin · 8 years ago
  98. 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 · 8 years ago
  99. 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 · 8 years ago
  100. 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 · 8 years ago