Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / de6fa1ef2946002c305861742561f94b8606426d / call / rampup_tests.h
  1. 9f5ae7b Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting by Danil Chapovalov · 5 years ago
  2. 44db436 Propagate task queue to create test::DirectTransport by TaskQueueBase interface by Danil Chapovalov · 5 years ago
  3. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
  4. 6b117a5 Make the callbacks to PollStats for RampUp* tests more regular. by Tommi · 5 years ago
  5. 891d393 Call Call::GetStats() from the correct thread in ProbingEndToEndTest. by Tommi · 5 years ago
  6. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
  7. 5e005f4 Fix RampUp tests to call Call::GetStats() from the right thread - and remove the need for a dedicated polling thread. by Tommi · 5 years ago
  8. de8e6e6 Refactor bitrate configuration in CallTest by Niels Möller · 6 years ago
  9. f5e767d Don't send max allocation probe unless allocation changed. by Sebastian Jansson · 6 years ago
  10. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  11. 631cafa Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe by Artem Titov · 6 years ago
  12. 46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
  13. 7258224 Replaces call config create in tests with modify. by Sebastian Jansson · 6 years ago
  14. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  15. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/rampup_tests.h]
  16. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  17. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  18. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  19. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  20. d8ce1e1 Move SelectMediaType from RampUpTester to BaseTest. by nisse · 8 years ago
  21. 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
  22. 45b5fe5 Don't report perf metrics for packet loss ramp-up tests. by stefan · 8 years ago
  23. 0f8b403 Introduce a new constructor to PlatformThread. by tommi · 8 years ago
  24. 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
  25. 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
  26. a518a39 Fixes a bug where a video stream can get stuck in the suspended state. by stefan · 8 years ago
  27. 5a2c506 Set the start bitrate to the delay-based BWE. by stefan · 8 years ago
  28. 38d8b3c Clean up ramp-up tests and make sure they all pass. by stefan · 8 years ago
  29. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  30. db752f9 Revert "Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )" by stefan · 8 years ago
  31. fbfb536 Explicitly enable RED over RTX in rampup tests. by brandtr · 8 years ago
  32. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  33. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  34. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  35. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  36. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  37. b25345e Replace scoped_ptr with unique_ptr in webrtc/call/ by kwiberg · 9 years ago
  38. ff2a635 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 9 years ago
  39. d20e651 Fix test bug introduced in r11101. by Stefan Holmer · 9 years ago
  40. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  41. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago[Renamed (82%) from webrtc/video/rampup_tests.h]
  42. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  43. 8c38e8b Clean up PlatformThread. by Peter Boström · 9 years ago
  44. 12411ef Move ThreadWrapper to ProcessThread in base. by pbos · 9 years ago
  45. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  46. f116bd0 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  47. 092508a Fix bug in ramp-up tests stats where rtx was accounted for in the media ssrc. by stefan · 9 years ago
  48. 4fbd145 Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side. by stefan · 9 years ago
  49. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago
  50. 6b8d355 Reland "Wire up send-side bandwidth estimation." by Erik Språng · 9 years ago
  51. c9bbeb0 Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ ) by Erik Språng · 9 years ago
  52. ef165ee Wire up send-side bandwidth estimation. by sprang · 9 years ago
  53. 68786d2 Wire up PacketTime to ReceiveStreams. by stefan · 9 years ago
  54. 11324b9 Wait for a longer time (5 seconds) before establishing the first bandwidth estimate. by Stefan Holmer · 9 years ago
  55. 468e62a Remove MimdRateControl and factories for RemoteBitrateEstimor. by Erik Språng · 9 years ago
  56. f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 9 years ago
  57. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 9 years ago
  58. e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 9 years ago
  59. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  60. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  61. 8f27fcc Revert 8028 "Support associated payload type when registering Rt..." by andrew@webrtc.org · 10 years ago
  62. 2a16964 Support associated payload type when registering Rtx payload type. by pbos@webrtc.org · 10 years ago
  63. 273a414 Report encoded frame size in VideoSendStream. by pbos@webrtc.org · 10 years ago
  64. 3d7da88 Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. by stefan@webrtc.org · 10 years ago