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gerrit-public.fairphone.software / platform / external / webrtc / 83d5e86163c4a60a90ea90645d2ec185e57150fc / call / rampup_tests.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. 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
  3. de8e6e6 Refactor bitrate configuration in CallTest by Niels Möller · 6 years ago
  4. cd2e105 Reenable test RampUpTest.AudioTransportSequenceNumber by Niels Möller · 6 years ago
  5. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  6. 2dfa998 Reland "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  7. c538fc7 Revert "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  8. 5ccdc13 Prefix flag macros with WEBRTC_. by Mirko Bonadei · 6 years ago
  9. f5e767d Don't send max allocation probe unless allocation changed. by Sebastian Jansson · 6 years ago
  10. cb7e1d2 Use SdpVideoFormat in VideoReceiveStream::Decoder by Niels Möller · 6 years ago
  11. bb095aa Allow send bitrate < start bitrate in RampUpTest. by Niels Möller · 6 years ago
  12. b0588e6 Reland "Move FakeCodec to separate target and behave like real encoder." by Ilya Nikolaevskiy · 6 years ago
  13. 631cafa Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe by Artem Titov · 6 years ago
  14. 4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
  15. 7258224 Replaces call config create in tests with modify. by Sebastian Jansson · 6 years ago
  16. cd8f382 Removes unnecessary destructor checks in tests. by Sebastian Jansson · 6 years ago
  17. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  18. 259a497 Reland "Reland "Move rtp-specific config out of EncoderSettings."" by Niels Möller · 7 years ago
  19. 0a713b6 replace stringstream in call/ by Jonas Olsson · 7 years ago
  20. 6c2c13a Revert "Reland "Move rtp-specific config out of EncoderSettings."" by Niels Möller · 7 years ago
  21. 04dd176 Reland "Move rtp-specific config out of EncoderSettings." by Niels Möller · 7 years ago
  22. 92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 7 years ago
  23. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 7 years ago
  24. 12fb170 Added some margin to ramp down target in perf test. by Sebastian Jansson · 7 years ago
  25. ba6b503 Disable RampUpTest.UpDownUpTransportSequenceNumberRtx on Mac. by Taylor Brandstetter · 7 years ago
  26. ab86e7f Disable a couple flaky RampUpTest tests on Mac. by Taylor Brandstetter · 7 years ago
  27. dd3987f Add _[no]red suffix to RampUpTests. by Edward Lemur · 7 years ago
  28. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  29. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  30. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  31. 05d9822 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss. by Taylor Brandstetter · 7 years ago
  32. 06319b7 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Mac. by Alex Loiko · 7 years ago
  33. 1405afe Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Linux due to flakiness. by lliuu · 7 years ago
  34. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  35. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  36. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/rampup_tests.cc]
  37. ca5706d Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) by nisse · 7 years ago
  38. 8e7eee0 Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) by nisse · 7 years ago
  39. 35713ea Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) by nisse · 7 years ago
  40. 3c39c01 Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) by nisse · 7 years ago
  41. 5c0f6c6 Use RtxReceiveStream. by nisse · 7 years ago
  42. 26e3abb Reverse |rtx_payload_types| map, and rename. by nisse · 7 years ago
  43. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  44. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  45. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  46. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  47. 4847ae6 Reland of Periodically update codec bit/frame rate settings. by sprang · 7 years ago
  48. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
  49. d8ce1e1 Move SelectMediaType from RampUpTester to BaseTest. by nisse · 8 years ago
  50. 76d9c9c Reland of Enable trendline experiment and bayesian bitrate estimator experiment by default. by stefan · 8 years ago
  51. 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
  52. 27925de Enable trendline experiment and bayesian bitrate estimator experiment by default. by stefan · 8 years ago
  53. 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
  54. c53a17f Enable the bayesian bitrate estimator by default. by stefan · 8 years ago
  55. 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
  56. 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
  57. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 8 years ago
  58. ff2ebf5 Clean up perf metrics and report ramp-up stats for fewer tests. by stefan · 8 years ago
  59. 45b5fe5 Don't report perf metrics for packet loss ramp-up tests. by stefan · 8 years ago
  60. 0f8b403 Introduce a new constructor to PlatformThread. by tommi · 8 years ago
  61. 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
  62. 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
  63. a518a39 Fixes a bug where a video stream can get stuck in the suspended state. by stefan · 8 years ago
  64. 5a2c506 Set the start bitrate to the delay-based BWE. by stefan · 8 years ago
  65. 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
  66. 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
  67. fe2bef3 Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. by brandtr · 8 years ago
  68. 38d8b3c Clean up ramp-up tests and make sure they all pass. by stefan · 8 years ago
  69. 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
  70. fbfb536 Explicitly enable RED over RTX in rampup tests. by brandtr · 8 years ago
  71. 10cbb46 Fixing config for Audio BWE. by minyue · 8 years ago
  72. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  73. b5f2c3f Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
  74. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  75. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  76. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  77. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  78. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  79. 649a21a Disable RampUpTest.UpDownUpThreeStreams. by philipel · 8 years ago
  80. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  81. 101f250 Implementing auto pausing of video streams. by mflodman · 8 years ago
  82. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  83. f4d8441 Disabled flaky tests by philipel · 9 years ago
  84. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  85. ff2a635 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 9 years ago
  86. d20e651 Fix test bug introduced in r11101. by Stefan Holmer · 9 years ago
  87. f1685c7 Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac by kjellander · 9 years ago
  88. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  89. 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]
  90. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  91. d1590b2 Lint clean video/ and add lint presubmit check. by mflodman · 9 years ago
  92. 8c38e8b Clean up PlatformThread. by Peter Boström · 9 years ago
  93. 12411ef Move ThreadWrapper to ProcessThread in base. by pbos · 9 years ago
  94. 43edf0f Require negotiation to send transport cc feedback over RTCP. by stefan · 9 years ago
  95. 0b9e29c Remove include dirs from modules/{media_file,pacing} by Henrik Kjellander · 9 years ago
  96. d153a37 Remove contention between RTCP packets and encoding. by Peter Boström · 9 years ago
  97. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  98. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  99. f116bd0 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  100. 723dff1 Poll stats more often to get more stable stats in ramp-up tests. by Stefan Holmer · 9 years ago