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gerrit-public.fairphone.software / platform / external / webrtc / c4f18fed26cfc39ea6610df439a1d67cf14da051 / call / rampup_tests.cc
  1. dd3987f Add _[no]red suffix to RampUpTests. by Edward Lemur · 7 years ago
  2. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  3. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  4. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  5. 05d9822 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss. by Taylor Brandstetter · 7 years ago
  6. 06319b7 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Mac. by Alex Loiko · 7 years ago
  7. 1405afe Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Linux due to flakiness. by lliuu · 7 years ago
  8. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  9. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  10. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/rampup_tests.cc]
  11. ca5706d Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) by nisse · 7 years ago
  12. 8e7eee0 Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) by nisse · 7 years ago
  13. 35713ea Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) by nisse · 7 years ago
  14. 3c39c01 Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) by nisse · 7 years ago
  15. 5c0f6c6 Use RtxReceiveStream. by nisse · 7 years ago
  16. 26e3abb Reverse |rtx_payload_types| map, and rename. by nisse · 7 years ago
  17. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  18. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  19. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  20. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  21. 4847ae6 Reland of Periodically update codec bit/frame rate settings. by sprang · 7 years ago
  22. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
  23. d8ce1e1 Move SelectMediaType from RampUpTester to BaseTest. by nisse · 8 years ago
  24. 76d9c9c Reland of Enable trendline experiment and bayesian bitrate estimator experiment by default. by stefan · 8 years ago
  25. 029f7cc Revert of Enable trendline experiment and bayesian bitrate estimator experiment by default. (patchset #6 id:100001 of https://codereview.webrtc.org/2777333003/ ) by lliuu · 8 years ago
  26. 27925de Enable trendline experiment and bayesian bitrate estimator experiment by default. by stefan · 8 years ago
  27. 7a3615b Revert of Enable the bayesian bitrate estimator by default. (patchset #5 id:80001 of https://codereview.webrtc.org/2749803002/ ) by lliuu · 8 years ago
  28. c53a17f Enable the bayesian bitrate estimator by default. by stefan · 8 years ago
  29. 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
  30. 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
  31. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 8 years ago
  32. ff2ebf5 Clean up perf metrics and report ramp-up stats for fewer tests. by stefan · 8 years ago
  33. 45b5fe5 Don't report perf metrics for packet loss ramp-up tests. by stefan · 8 years ago
  34. 0f8b403 Introduce a new constructor to PlatformThread. by tommi · 8 years ago
  35. 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
  36. 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
  37. a518a39 Fixes a bug where a video stream can get stuck in the suspended state. by stefan · 8 years ago
  38. 5a2c506 Set the start bitrate to the delay-based BWE. by stefan · 8 years ago
  39. 1474212 Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) by brandtr · 8 years ago
  40. e497495 Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) by kjellander · 8 years ago
  41. fe2bef3 Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. by brandtr · 8 years ago
  42. 38d8b3c Clean up ramp-up tests and make sure they all pass. by stefan · 8 years ago
  43. 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
  44. fbfb536 Explicitly enable RED over RTX in rampup tests. by brandtr · 8 years ago
  45. 10cbb46 Fixing config for Audio BWE. by minyue · 8 years ago
  46. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  47. b5f2c3f Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
  48. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  49. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  50. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  51. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  52. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  53. 649a21a Disable RampUpTest.UpDownUpThreeStreams. by philipel · 8 years ago
  54. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  55. 101f250 Implementing auto pausing of video streams. by mflodman · 8 years ago
  56. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  57. f4d8441 Disabled flaky tests by philipel · 9 years ago
  58. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  59. ff2a635 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 9 years ago
  60. d20e651 Fix test bug introduced in r11101. by Stefan Holmer · 9 years ago
  61. f1685c7 Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac by kjellander · 9 years ago
  62. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  63. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago[Renamed (78%) from webrtc/video/rampup_tests.cc]
  64. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  65. d1590b2 Lint clean video/ and add lint presubmit check. by mflodman · 9 years ago
  66. 8c38e8b Clean up PlatformThread. by Peter Boström · 9 years ago
  67. 12411ef Move ThreadWrapper to ProcessThread in base. by pbos · 9 years ago
  68. 43edf0f Require negotiation to send transport cc feedback over RTCP. by stefan · 9 years ago
  69. 0b9e29c Remove include dirs from modules/{media_file,pacing} by Henrik Kjellander · 9 years ago
  70. d153a37 Remove contention between RTCP packets and encoding. by Peter Boström · 9 years ago
  71. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  72. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  73. f116bd0 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  74. 723dff1 Poll stats more often to get more stable stats in ramp-up tests. by Stefan Holmer · 9 years ago
  75. f3a7c9d In rampup tests, set start time when starting poller thread. by Erik Språng · 9 years ago
  76. a050e98 Avoid race in RampUpTest by sprang · 9 years ago
  77. 092508a Fix bug in ramp-up tests stats where rtx was accounted for in the media ssrc. by stefan · 9 years ago
  78. 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
  79. 6b8d355 Reland "Wire up send-side bandwidth estimation." by Erik Språng · 9 years ago
  80. 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
  81. ef165ee Wire up send-side bandwidth estimation. by sprang · 9 years ago
  82. 91d6ede Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 9 years ago
  83. 68786d2 Wire up PacketTime to ReceiveStreams. by stefan · 9 years ago
  84. 4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 9 years ago
  85. caa498a Make sure RTCP is sent in tests when receiving packets even if REMB is delayed. by stefan · 9 years ago
  86. 11324b9 Wait for a longer time (5 seconds) before establishing the first bandwidth estimate. by Stefan Holmer · 9 years ago
  87. 468e62a Remove MimdRateControl and factories for RemoteBitrateEstimor. by Erik Språng · 9 years ago
  88. bd2522a Fail RTP parsing on excessive padding length. by pbos · 9 years ago
  89. ff4ea93 Only use paced packets for estimating bitrate probes. by Stefan Holmer · 9 years ago
  90. f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 10 years ago
  91. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
  92. e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 10 years ago
  93. 9526187 Default enable abs send time bwe for CallTest by Erik Språng · 10 years ago
  94. e590416 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 10 years ago
  95. 2b4ce3a Convert webrtc/video/ abort/assert to CHECK/DCHECK. by pbos@webrtc.org · 10 years ago
  96. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  97. 8f27fcc Revert 8028 "Support associated payload type when registering Rt..." by andrew@webrtc.org · 10 years ago
  98. 2a16964 Support associated payload type when registering Rtx payload type. by pbos@webrtc.org · 10 years ago
  99. 8817256 Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator. by stefan@webrtc.org · 10 years ago
  100. 742386a Enable payload-based padding by default and remove the API. by stefan@webrtc.org · 10 years ago