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gerrit-public.fairphone.software / platform / external / webrtc / 4030c65c30995c8b0920f9e01454cae5c792fc49 / test / call_test.cc
  1. 3f6804d Optional: Use nullopt and implicit construction in /test by Oskar Sundbom · 7 years ago
  2. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  3. 62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
  4. 255d1cd Implement dual stream full stack test and loopback tool by Ilya Nikolaevskiy · 7 years ago
  5. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  6. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  7. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  8. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  9. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  10. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  11. 7cd28b9 Set protected_by_flexfec flag properly in tests. by brandtr · 7 years ago
  12. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  13. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/test/call_test.cc]
  14. 73276ad - Removes voe_conference_test. by Fredrik Solenberg · 7 years ago
  15. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  16. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  17. 2bf9e73 Delete unneeded Start and Stop methods on FlexfecReceiveStream. by Niels Möller · 7 years ago
  18. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  19. 445f1a1 nit: Order CallTest's methods in the .cc according to their order in the .h file. by eladalon · 7 years ago
  20. c0d481a Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
  21. 863f03b Fix video_replay tool to respect RTX stream and fix default parameters. by ilnik · 7 years ago
  22. d2702ef Fix flaky test VideoSendStreamTest.SendsKeepAlive by sprang · 7 years ago
  23. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  24. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  25. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  26. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  27. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  28. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  29. 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 · 7 years ago
  30. 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 · 7 years ago
  31. 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 · 7 years ago
  32. 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 · 7 years ago
  33. 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 · 7 years ago
  34. 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 · 7 years ago
  35. 64e739a Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 7 years ago
  36. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 7 years ago
  37. 4fb651d Event log cleanup in tests. by philipel · 7 years ago
  38. d8ce1e1 Move SelectMediaType from RampUpTester to BaseTest. by nisse · 7 years ago
  39. 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
  40. 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
  41. 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
  42. 8a25652 Reduce flakiness in EndToEnd probing tests. by philipel · 7 years ago
  43. 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
  44. a1ab8ba We need to specify the decoder map explicitly nowadays by kwiberg · 7 years ago
  45. 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
  46. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 7 years ago
  47. 92220ff Low-bandwidth audio testing by oprypin · 7 years ago
  48. 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
  49. 72acf25 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 7 years ago
  50. e828c96 Probing EndToEndTests. by philipel · 7 years ago
  51. a514584 Add the ability to read/write to WAV files in FakeAudioDevice by oprypin · 8 years ago
  52. a014cc5 Reland of "Added large room scenario to full-stack tests" by ilnik · 8 years ago
  53. bfb1245 Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ ) by ilnik · 8 years ago
  54. d8bd1b1 Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test. by ilnik · 8 years ago
  55. 5ef2bc1 Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) by philipel · 8 years ago
  56. b80bdca Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) by philipel · 8 years ago
  57. a518a39 Fixes a bug where a video stream can get stuck in the suspended state. by stefan · 8 years ago
  58. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  59. b77c716 Enable send-side BWE by default for video in call tests. by stefan · 8 years ago
  60. ac61b74 Refactor FakeAudioDevice to have separate methods for starting recording and playout. by perkj · 8 years ago
  61. fb45c6c Inform jitter buffer about FlexFEC protection. by brandtr · 8 years ago
  62. fa5a368 Let FlexfecReceiveStreamImpl send RTCP RRs. by brandtr · 8 years ago
  63. 3d200bd Remove FlexfecConfig and replace with specific struct in VideoSendStream. by brandtr · 8 years ago
  64. 8313a6f Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config. by brandtr · 8 years ago
  65. b29e652 Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )" by brandtr · 8 years ago
  66. 1cfbd60 Generalize FlexfecReceiveStream::Config. by brandtr · 8 years ago
  67. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  68. e2b1501 Start probes only after network is connected. by Sergey Ulanov · 8 years ago
  69. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  70. 906c5dc Revert of Start probes only after network is connected. (patchset #9 id:240001 of https://codereview.webrtc.org/2458863002/ ) by honghaiz · 8 years ago
  71. 5c99c76 Start probes only after network is connected. by sergeyu · 8 years ago
  72. 841de6a Add FlexFEC to CallTest. by brandtr · 8 years ago
  73. 803d97f Let ViEEncoder express resolution requests as Sinkwants. by perkj · 8 years ago
  74. e566ac7 Remove voe::Channel::StopReceive() and associated logic. by solenberg · 8 years ago
  75. 68e6bdd Remove use of VoECodec in video/call tests. by solenberg · 8 years ago
  76. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  77. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  78. 55d932b Add logging statements to places where the frame might be dropped in WebRTC pipeline. by sakal · 8 years ago
  79. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  80. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  81. a49cbd3 Replace VideoCapturerInput with VideoSinkInterface. by perkj · 8 years ago
  82. 9fdbda6 Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ ) by perkj · 8 years ago
  83. 95a226f Replace VideoCapturerInput with VideoSinkInterface. by perkj · 8 years ago
  84. 88499ec Moving/renaming webrtc/common.h. by solenberg · 8 years ago
  85. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  86. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  87. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  88. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  89. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  90. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 8 years ago
  91. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  92. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 8 years ago
  93. ba7dc72 Add rotation to EncodedImage and make sure it is passed through encoders. by Per · 8 years ago
  94. 4a206a9 Remove webrtc::ScopedVector by kwiberg · 8 years ago
  95. 9c6a0c7 Added A/V sync tests with drifting clocks. by danilchap · 9 years ago
  96. 04cb763 Add tests for verifying transport feedback for audio and video. by Stefan Holmer · 9 years ago
  97. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  98. 9fea80f Add audio streams to CallTest and a first A/V call test. by Stefan Holmer · 9 years ago
  99. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago
  100. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago