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gerrit-public.fairphone.software / platform / external / webrtc / cde46b72788f5e37cfae33824b54d9c7d8a14ef8
  1. cde46b7 Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 7 years ago
  2. 28dc285 Adding cbr support for Opus by soren · 7 years ago
  3. 388fe42 Make WARN_UNUSED_RESULT a no-op on gcc by kwiberg · 7 years ago
  4. 177b17e Move AndroidVideoTrackSourceObserver from API to src by magjed · 7 years ago
  5. 639d46a Delete system_wrappers logging facility. by nisse · 7 years ago
  6. be77920 Revert of CQ: Remove Linux ARM64 Debug trybot from default set. (patchset #1 id:1 of https://codereview.webrtc.org/2790263003/ ) by kjellander · 7 years ago
  7. 2418001 ACM: Change test output files from PCM to WAV by henrik.lundin · 7 years ago
  8. 4fcfdd8 Enable rtc_unittests on iOS simulator by kjellander · 7 years ago
  9. a280f7c Added integer parsing functions in base/string_to_number.h by ossu · 7 years ago
  10. b1e3fc4 Enable tools_unittests and rtc_stats_unittests on iOS Simulator by kjellander · 7 years ago
  11. e24991d Adds AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex unittest. by henrika · 7 years ago
  12. 978504e Move rtp header extension length check from Packet::FindExtension to ExtensionT::Parse by danilchap · 7 years ago
  13. ed6343d Roll chromium_revision 875e8893e9..3014f8b41e (462360:462374) by buildbot · 7 years ago
  14. 251eb27 Roll chromium_revision 75820eb165..875e8893e9 (460410:462360) by kjellander · 7 years ago
  15. f6a4f37 Reland of Fixed error for missing explict class initialization error on iOS WebRTC buildbots (patchset #1 id:1 of https://codereview.webrtc.org/2803933002/ ) by guidou · 7 years ago
  16. 854e507 Revert of Fixed error for missing explict class initialization error on iOS WebRTC buildbots (patchset #1 id:1 of https://codereview.webrtc.org/2799813002/ ) by guidou · 7 years ago
  17. 5ac18af Fixed error for missing explicit class initialization error on iOS buildbots by peah · 7 years ago
  18. 7343c8e DirectX capturer may crash after switching shared screen by zijiehe · 7 years ago
  19. cf02cf1 Major AEC3 render pipeline changes by peah · 7 years ago
  20. 4aceaf2 Android: Move Histogram from api to src. by sakal · 7 years ago
  21. c522e75 Use new RTCCameraVideoCapturer in AppRTCMobile. by sakal · 7 years ago
  22. 1ba21eb Add [c]begin() and [c]end() member functions to rtc::Buffer by kwiberg · 7 years ago
  23. dea682d This CL fixes the following: by alessiob · 7 years ago
  24. 129fc9c Enabling 'gn check' on //webrtc/tools. by mbonadei · 7 years ago
  25. c88b5d5 Reland of PyLint fixes for tools-webrtc and webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2737233003/ ) by kjellander · 7 years ago
  26. adf0635 Make GetConfig() part of the AudioProcessing interface by henrik.lundin · 7 years ago
  27. 368f5cf Replace use of system_wrappers/include/logging.h by base/logging.h. by nisse · 7 years ago
  28. 2299b0a Android: Remove VideoCapturerAndroid by magjed · 7 years ago
  29. d60d06a Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) by ilnik · 7 years ago
  30. d8ce1e1 Move SelectMediaType from RampUpTester to BaseTest. by nisse · 7 years ago
  31. 6eca98b Add histogram stats for number of cpu/quality adapt changes per minute for sent video streams: by asapersson · 7 years ago
  32. d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 7 years ago
  33. 465faf0 [iOS] Changed ptr to const ref for RTCConfiguration initialization by jtteh · 7 years ago
  34. ba13131 Trivial data() and mutable_data() implementations by yujo · 7 years ago
  35. baf9b58 README.md with deps, build, usage, troubleshooting by alessiob · 7 years ago
  36. c337258 Revert of Deliver video frames on Android, on the decode thread. (patchset #7 id:120001 of https://codereview.webrtc.org/2764573002/ ) by guidou · 7 years ago
  37. fbd4f85 Javascript audio player for the exported HTML file. by alessiob · 7 years ago
  38. fab6707 Add number of quality adapt changes to VideoSendStream stats. by asapersson · 7 years ago
  39. e3aa88b Deliver video frames on Android, on the decode thread. by tommi · 7 years ago
  40. aa7d935 Evaluation scores export library and CSS file for the exported HTML file by alessiob · 7 years ago
  41. 8edb839 Reland of Export library that generates an HTLM file with the scores organized in tables. (patchset #1 id:1 of https://codereview.webrtc.org/2791293002/ ) by alessiob · 7 years ago
  42. fab482b Simplify RTPSender::RegisterRtpHeaderExtension by danilchap · 7 years ago
  43. 55a0135 Make sure we observe enough frames before scaling. by kthelgason · 7 years ago
  44. 751c9dc Roll chromium_revision 581ff14023..75820eb165 (459789:460410) by buildbot · 7 years ago
  45. 5115645 CQ: Remove Linux ARM64 Debug trybot from default set. by Henrik Kjellander · 7 years ago
  46. 3153c6d Fixing DTX unittest for AudioEncoderOpus. by minyue · 7 years ago
  47. 4271224 CQ: Remove linux32_rel from default trybots by Henrik Kjellander · 7 years ago
  48. 0786c04 Fix crash in XServerPixelBuffer. by sergeyu · 7 years ago
  49. 4eeb537 [iOS] Added an initialization method to RTCConfiguration that takes a by jtteh · 7 years ago
  50. 9ab17d3 Forward capturer_id to shared desktopframe by zijiehe · 7 years ago
  51. c964d0b Fixing some case-sensitive codec name comparisons. by deadbeef · 7 years ago
  52. 716d7ac Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) by guidou · 7 years ago
  53. 880c85b Revert of Export library that generates an HTLM file with the scores organized in tables. (patchset #3 id:40001 of https://codereview.webrtc.org/2717973006/ ) by alessiob · 7 years ago
  54. 29e3330 Export library that generates an HTLM file with the scores organized in tables. by alessiob · 7 years ago
  55. c42f540 Move video_encoder.h and video_decoder.h to /api and create GN targets for them by ilnik · 7 years ago
  56. eb4662a Single simulation runner. by alessiob · 7 years ago
  57. 0deb594 POLQA evaluation score. by alessiob · 7 years ago
  58. c533df2 Audio level evaluation score. by alessiob · 7 years ago
  59. 54ad3df I added two factory classes to address an important comment I got in another CL from kjellander@webrtc.org. by alessiob · 7 years ago
  60. 8a1b3c9 Environmental noise generator implemented. by alessiob · 7 years ago
  61. 653063f Add functions to get/set rtp header extension by id. by danilchap · 7 years ago
  62. 23425f9 Add methods to register congestion controller observer after construction. by nisse · 7 years ago
  63. d197cd9 Revert of Add empty header to fix internal project. (patchset #1 id:1 of https://codereview.webrtc.org/2790493006/ ) by kthelgason · 7 years ago
  64. d1b0e0e Merge UpdateBandwidthEstimate with Update in AimdRateControl. by terelius · 7 years ago
  65. 97a7fb0 Delete obsolete file mock_congestion_controller.h. by nisse · 7 years ago
  66. 52a0ce6 Delete unused method OnProbeBitrate. by nisse · 7 years ago
  67. 3a407ee Making RtpSender tests cover BWE with overhead. by minyue · 7 years ago
  68. 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 · 7 years ago
  69. 76d9c9c Reland of Enable trendline experiment and bayesian bitrate estimator experiment by default. by stefan · 7 years ago
  70. bc436ed Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ ) by lliuu · 7 years ago
  71. 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 · 7 years ago
  72. 8a8b238 ScreenCapturerWinDirectx should have two DxgiDuplicatorController::Context instances, each for one DesktopFrame. So both DesktopFrame instances can be correctly updated. by zijiehe · 7 years ago
  73. 5f93709 Supporting 48kHz PCM file. by minyue · 7 years ago
  74. 16d5bae Add empty header to fix internal project. by kthelgason · 7 years ago
  75. 27925de Enable trendline experiment and bayesian bitrate estimator experiment by default. by stefan · 7 years ago
  76. 9c79ed9 Add loss-based BWE experiment which allows us to try different parameters. by Stefan Holmer · 7 years ago
  77. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 7 years ago
  78. ec6fbd2 Moves channel-dependent audio input processing to separate encoder task queue. by henrika · 7 years ago
  79. 36e9eb4 Do not report quality limited resolution stats when quality scaler is disabled. by asapersson · 7 years ago
  80. 3883ccb New RTCCameraVideoCapturer. by sakal · 7 years ago
  81. ee8b861 remove module-wide warning suppression. by kthelgason · 7 years ago
  82. e6a8009 Remove voe_auto_test cases for VoEFile. by solenberg · 7 years ago
  83. d00aad5 Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ ) by mbonadei · 7 years ago
  84. 124a6fc MacOS: Add metal renderer and view. by denicija · 7 years ago
  85. 515dff4 Revert of Adding PRESUBMIT check on google::protobuf (patchset #2 id:20001 of https://codereview.webrtc.org/2753823003/ ) by mbonadei · 7 years ago
  86. ff046c7 Remove ALL usage of CriticalSectionWrapper. by kthelgason · 7 years ago
  87. 5533bd3 Reland: Use native (optimized) functions for byte order conversion. (patchset #1 id:1 of https://codereview.webrtc.org/2755103002/ ) by jbauch · 7 years ago
  88. 65a8308 Another landmine for low_bandwidth_audio_test by Henrik Kjellander · 7 years ago
  89. 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 · 7 years ago
  90. 7a3615b Revert of Enable the bayesian bitrate estimator by default. (patchset #5 id:80001 of https://codereview.webrtc.org/2749803002/ ) by lliuu · 7 years ago
  91. c53a17f Enable the bayesian bitrate estimator by default. by stefan · 7 years ago
  92. 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 · 7 years ago
  93. 8a25652 Reduce flakiness in EndToEnd probing tests. by philipel · 7 years ago
  94. b13237b Fix deprecated methods in AppRTCMobile. by kthelgason · 7 years ago
  95. 18703f9 Disable flaky test EndToEndTest.TriggerMidCallProbing by aleloi · 7 years ago
  96. 6d305ba Add Windows, Mac, Android support to low bandwidth audio test by oprypin · 7 years ago
  97. 30cbd0b Landmine #2 for https://codereview.webrtc.org/2767383005 by oprypin · 7 years ago
  98. 0318463 gtest-parallel: Concatenate the log files in the passed, failed and interrupted dirs. by ehmaldonado · 7 years ago
  99. dd27055 Adding PRESUBMIT check on google::protobuf by mbonadei · 7 years ago
  100. 16ab93b To accommodate some downstream WebRTC users we need to loosen by mbonadei · 7 years ago