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gerrit-public.fairphone.software / platform / external / webrtc / 34029e209d6d116c338e7d65d6980cc395184658 / test / call_test.h
  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. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  4. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  5. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  6. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/test/call_test.h]
  7. 73276ad - Removes voe_conference_test. by Fredrik Solenberg · 7 years ago
  8. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  9. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  10. c0d481a Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
  11. 863f03b Fix video_replay tool to respect RTX stream and fix default parameters. by ilnik · 7 years ago
  12. d2702ef Fix flaky test VideoSendStreamTest.SendsKeepAlive by sprang · 7 years ago
  13. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  14. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  15. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 7 years ago
  16. 4fb651d Event log cleanup in tests. by philipel · 7 years ago
  17. d8ce1e1 Move SelectMediaType from RampUpTester to BaseTest. by nisse · 7 years ago
  18. 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
  19. 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
  20. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 7 years ago
  21. 92220ff Low-bandwidth audio testing by oprypin · 7 years ago
  22. e828c96 Probing EndToEndTests. by philipel · 7 years ago
  23. a014cc5 Reland of "Added large room scenario to full-stack tests" by ilnik · 7 years ago
  24. 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 · 7 years ago
  25. d8bd1b1 Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test. by ilnik · 7 years ago
  26. fa5a368 Let FlexfecReceiveStreamImpl send RTCP RRs. by brandtr · 8 years ago
  27. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  28. 841de6a Add FlexFEC to CallTest. by brandtr · 8 years ago
  29. 68e6bdd Remove use of VoECodec in video/call tests. by solenberg · 8 years ago
  30. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  31. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  32. 55d932b Add logging statements to places where the frame might be dropped in WebRTC pipeline. by sakal · 8 years ago
  33. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  34. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  35. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  36. bfefb03 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 8 years ago
  37. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 8 years ago
  38. ba7dc72 Add rotation to EncodedImage and make sure it is passed through encoders. by Per · 8 years ago
  39. 4a206a9 Remove webrtc::ScopedVector by kwiberg · 8 years ago
  40. 9c6a0c7 Added A/V sync tests with drifting clocks. by danilchap · 8 years ago
  41. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  42. 9fea80f Add audio streams to CallTest and a first A/V call test. by Stefan Holmer · 9 years ago
  43. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago
  44. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  45. 1295297 Register header extensions in RtpRtcpObserver to avoid log spam. by Stefan Holmer · 9 years ago
  46. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  47. f116bd0 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  48. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  49. 2d56668 Unify Transport and newapi::Transport interfaces. by pbos · 9 years ago
  50. 4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 9 years ago
  51. e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 9 years ago
  52. 9526187 Default enable abs send time bwe for CallTest by Erik Språng · 9 years ago
  53. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 9 years ago
  54. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 9 years ago
  55. 8f27fcc Revert 8028 "Support associated payload type when registering Rt..." by andrew@webrtc.org · 10 years ago
  56. 2a16964 Support associated payload type when registering Rtx payload type. by pbos@webrtc.org · 10 years ago
  57. 776e6f2 Use external VideoDecoders in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  58. bbe0a85 Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
  59. 01581da Fix audio/video sync when FEC is enabled. by stefan@webrtc.org · 10 years ago
  60. 7ae9108 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. by andresp@webrtc.org · 10 years ago
  61. 91f1752 Support VP8 encoder settings in VideoSendStream. by pbos@webrtc.org · 10 years ago
  62. 2bb1bda Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  63. be9d2a4 Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  64. 994d0b7 Refactor Call-based tests. by pbos@webrtc.org · 10 years ago