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gerrit-public.fairphone.software / platform / external / webrtc / b46c552dcd0aa32c5e1cc0cec62b1d9c866fe479 / call / bitrate_estimator_tests.cc
  1. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  2. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/bitrate_estimator_tests.cc]
  3. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  4. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  5. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  6. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  7. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  8. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
  9. 4fb651d Event log cleanup in tests. by philipel · 8 years ago
  10. 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 · 8 years ago
  11. 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 · 8 years ago
  12. 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 · 8 years ago
  13. 0ffdcc5 Delete unneeded includes of deprecated system_wrappers include files. by nisse · 8 years ago
  14. 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
  15. 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
  16. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 8 years ago
  17. 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 · 8 years ago
  18. 72acf25 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 8 years ago
  19. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  20. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  21. 5d78e8d Remove audio from BitrateEstimatorTest. by aleloi · 8 years ago
  22. 803d97f Let ViEEncoder express resolution requests as Sinkwants. by perkj · 8 years ago
  23. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  24. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  25. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  26. 55d932b Add logging statements to places where the frame might be dropped in WebRTC pipeline. by sakal · 8 years ago
  27. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  28. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  29. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  30. a49cbd3 Replace VideoCapturerInput with VideoSinkInterface. by perkj · 8 years ago
  31. 9fdbda6 Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ ) by perkj · 8 years ago
  32. 95a226f Replace VideoCapturerInput with VideoSinkInterface. by perkj · 8 years ago
  33. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  34. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  35. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  36. 7522a28 Removed old probe cluster logic and logic related to ssrcs from DelayBasedBwe. by philipel · 8 years ago
  37. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  38. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  39. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 9 years ago
  40. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  41. 1086ed6 Disable SwitchesToASTThenBackToTOFForVideo test completely. by deadbeef · 9 years ago
  42. 844f993 Disabling SwitchesToASTThenBackToTOFForVideo test for MSan bot. by deadbeef · 9 years ago
  43. 4aa438c Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo. by minyuel · 9 years ago
  44. b25345e Replace scoped_ptr with unique_ptr in webrtc/call/ by kwiberg · 9 years ago
  45. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  46. 8c66a00 Initialize VideoSendStream members in constructor. by Peter Boström · 9 years ago
  47. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  48. 9fea80f Add audio streams to CallTest and a first A/V call test. by Stefan Holmer · 9 years ago
  49. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago
  50. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  51. 7c704b8 Use webrtc/base/logging.h in stefan@'s ownership. by Peter Boström · 9 years ago
  52. 521af4e Remove duplicate decoders in BitrateEstimatorTest. by Peter Boström · 9 years ago
  53. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  54. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  55. 0ccae13 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  56. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  57. f116bd0 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  58. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  59. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  60. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  61. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago[Renamed from webrtc/video/bitrate_estimator_tests.cc]
  62. 91d6ede Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 9 years ago
  63. 4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 9 years ago
  64. 6bb1b6e Control combined_audio_video_bwe with config bool. by pbos · 10 years ago
  65. 8fc7fa7 Base A/V synchronization on sync_labels. by pbos · 10 years ago
  66. 468e62a Remove MimdRateControl and factories for RemoteBitrateEstimor. by Erik Språng · 10 years ago
  67. d7da120 Disable reduced-size RTCP in default config. by Peter Boström · 10 years ago
  68. f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 10 years ago
  69. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
  70. 2b4ce3a Convert webrtc/video/ abort/assert to CHECK/DCHECK. by pbos@webrtc.org · 10 years ago
  71. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  72. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  73. 776e6f2 Use external VideoDecoders in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  74. 38344ed Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
  75. bbe0a85 Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
  76. ab071da Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  77. 6f729e8 Disable video_engine_tests and webrtc_perf_tests on Android. by kjellander@webrtc.org · 10 years ago
  78. dde16f1 Fix some code styles. by pbos@webrtc.org · 10 years ago
  79. b941fe8 Fix data races related with traces in bitrate estimator test. by andresp@webrtc.org · 11 years ago
  80. bd249bc Remove GetDefaultConfigs() from Call. by pbos@webrtc.org · 11 years ago
  81. 994d0b7 Refactor Call-based tests. by pbos@webrtc.org · 11 years ago
  82. 6ae48c6 Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 11 years ago
  83. db60434 Re-enable the BitrateEstimatorTest cases for the Call API. by solenberg@webrtc.org · 11 years ago
  84. de1429e Add thread annotations to Call API. by pbos@webrtc.org · 11 years ago
  85. 5ca6a53 Remove TraceCallback use from Call. by pbos@webrtc.org · 11 years ago
  86. a5c8d2c Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 11 years ago
  87. b08db28 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 11 years ago
  88. f577ae9 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  89. ab24051 Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
  90. 5ab7567 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  91. 41e2615 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  92. 341e914 Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago