Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 8a3b9bf435b7330f472468ed114d9f929f2aab3d / test / call_test.cc
  1. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  2. 7f57788 Removes trial to enable BBR congestion controller. by Sebastian Jansson · 6 years ago
  3. 1c931c4 Use VideoSourceInterface for owning test video sources by Niels Möller · 6 years ago
  4. 8eeccbe Delete Start and Stop methods from TestVideoCapturer. by Niels Möller · 6 years ago
  5. f7f13e0 Add end-to-end test for ColorSpace information by Johannes Kron · 6 years ago
  6. de8e6e6 Refactor bitrate configuration in CallTest by Niels Möller · 6 years ago
  7. cb7eddb Add tests for cpu overuse scaling. by Åsa Persson · 6 years ago
  8. c2ebe21 Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" by Jiawei Ou · 6 years ago
  9. 59844ce Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`." by Qingsi Wang · 6 years ago
  10. be14217 Use the factory instead of using the builtin code path in `VideoCodecInitializer`. by Jiawei Ou · 6 years ago
  11. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  12. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  13. cbcbc22 Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Niels Möller · 6 years ago
  14. 377b26e Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Sebastian Jansson · 6 years ago
  15. efb94d5 Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."" by Oleh Prypin · 6 years ago
  16. 7961dc2 Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Niels Moller · 6 years ago
  17. 529d0d9 Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config. by Niels Möller · 6 years ago
  18. 569397f Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor." by philipel · 6 years ago
  19. 6f68324 Revert "Added field trial WebRTC-GenericDescriptor for the new generic descriptor." by Lu Liu · 6 years ago
  20. 3f4a4fa Added field trial WebRTC-GenericDescriptor for the new generic descriptor. by philipel · 6 years ago
  21. cb7e1d2 Use SdpVideoFormat in VideoReceiveStream::Decoder by Niels Möller · 6 years ago
  22. b0588e6 Reland "Move FakeCodec to separate target and behave like real encoder." by Ilya Nikolaevskiy · 6 years ago
  23. 8d92e8d Revert "Reland "Move FakeCodec to separate target and behave like real encoder."" by Ilya Nikolaevskiy · 6 years ago
  24. f2a8287 Reland "Move FakeCodec to separate target and behave like real encoder." by Ilya Nikolaevskiy · 6 years ago
  25. 7d13a6e Revert "Move FakeCodec to separate target and behave like real encoder." by Ilya Nikolaevskiy · 6 years ago
  26. 223eba5 Move FakeCodec to separate target and behave like real encoder. by Ilya Nikolaevskiy · 6 years ago
  27. 4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
  28. 46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
  29. 50eb4c4 Adds BBR field trial to CallTest. by Sebastian Jansson · 6 years ago
  30. 7258224 Replaces call config create in tests with modify. by Sebastian Jansson · 6 years ago
  31. 3bd2c79 Moving functionality from VideoQualityTest to CallTest 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 · 6 years ago
  35. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  36. 431abd9 Replace rtc::Optional with absl::optional in test and rtc_tools by Danil Chapovalov · 6 years ago
  37. dfce03a Allows injection of network controller factory into peer connection factory. by Sebastian Jansson · 6 years ago
  38. 0a8f435 Move VideoEncoderConfig from call/ to api/. by Niels Möller · 6 years ago
  39. 49fcc10 Merge DegradationPreference enums. by Taylor Brandstetter · 6 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. fb82fcc Move creating encoder to VideoStreamEncoder. by Niels Möller · 7 years ago
  43. 259a497 Reland "Reland "Move rtp-specific config out of EncoderSettings."" by Niels Möller · 7 years ago
  44. 6c2c13a Revert "Reland "Move rtp-specific config out of EncoderSettings."" by Niels Möller · 7 years ago
  45. 2784a03 Add audio_ prefix to audio-related members of CallTest. by Niels Möller · 7 years ago
  46. 04dd176 Reland "Move rtp-specific config out of EncoderSettings." by Niels Möller · 7 years ago
  47. 92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 7 years ago
  48. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 7 years ago
  49. 465a5d9 Refactor payload types constants in CallTest by Ilya Nikolaevskiy · 7 years ago
  50. 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
  51. 03e6ec9 Reland "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
  52. 081136f Revert "Reland "Add multiplex case to webrtc_perf_tests"" by Taylor Brandstetter · 7 years ago
  53. 7c5bc1c Reland "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
  54. 5aac372 Revert "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
  55. d90a7e8 Add multiplex case to webrtc_perf_tests by Emircan Uysaler · 7 years ago
  56. 6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 7 years ago
  57. 8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 7 years ago
  58. 9a03dd8 Removed new calls on RtpTransportControllerSend. by Sebastian Jansson · 7 years ago
  59. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  60. 3f6804d Optional: Use nullopt and implicit construction in /test by Oskar Sundbom · 7 years ago
  61. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  62. 62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
  63. 255d1cd Implement dual stream full stack test and loopback tool by Ilya Nikolaevskiy · 7 years ago
  64. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  65. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  66. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  67. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  68. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  69. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  70. 7cd28b9 Set protected_by_flexfec flag properly in tests. by brandtr · 7 years ago
  71. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  72. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/test/call_test.cc]
  73. 73276ad - Removes voe_conference_test. by Fredrik Solenberg · 7 years ago
  74. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  75. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  76. 2bf9e73 Delete unneeded Start and Stop methods on FlexfecReceiveStream. by Niels Möller · 7 years ago
  77. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  78. 445f1a1 nit: Order CallTest's methods in the .cc according to their order in the .h file. by eladalon · 7 years ago
  79. c0d481a Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
  80. 863f03b Fix video_replay tool to respect RTX stream and fix default parameters. by ilnik · 7 years ago
  81. d2702ef Fix flaky test VideoSendStreamTest.SendsKeepAlive by sprang · 7 years ago
  82. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  83. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  84. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  85. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  86. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  87. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  88. 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
  89. 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
  90. 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
  91. 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
  92. 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
  93. 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
  94. 64e739a Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 8 years ago
  95. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
  96. 4fb651d Event log cleanup in tests. by philipel · 8 years ago
  97. d8ce1e1 Move SelectMediaType from RampUpTester to BaseTest. by nisse · 8 years ago
  98. 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
  99. 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
  100. 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