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gerrit-public.fairphone.software / platform / external / webrtc / 6a35590d14fd8eb3bdfa085095b2362b84338d14
  1. 6a35590 Add code for dummy file audio to fallback to dummy audio. by noahric · 8 years ago
  2. 7c0f8ee Avoid null pointer exception if Android getCameraInfo fails. by Alex Glaznev · 8 years ago
  3. d8a72f0 Close input file in FileAudioDevice::StopRecording. by noahric · 8 years ago
  4. 78810b6 Expose media constraint string constants as ObjC NSStrings by magjed · 8 years ago
  5. d22854b FilePlayer: Remove unused default values for arguments by kwiberg · 8 years ago
  6. 4a42900 Removes redundant log warning in WebRtcAudioManager. by henrika · 8 years ago
  7. 86c9694 Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ ) by danilchap · 8 years ago
  8. 5a25d95 FileRecorder + FilePlayer: Let Create functions return unique_ptr by kwiberg · 8 years ago
  9. 4466782 StartTimestamp generated randomly in RtpSender constructor by Danil Chapovalov · 8 years ago
  10. 2ae1fb6 Fix get_landmines.py script. by ehmaldonado · 8 years ago
  11. 144dd27 FileRecorderImpl and FilePlayerImpl don't need their own .h and .cc files by kwiberg · 8 years ago
  12. c54071d WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. by ossu · 8 years ago
  13. a93d5ac Don't simulate probing based on rtc event logs since we don't have that info logged. by stefan · 8 years ago
  14. eb680ea CongestionController::SetBweBitrates may now trigger probing. by philipel · 8 years ago
  15. c594aa61 Add a gyp/gn option to use dummy audio file devices. by noahric · 8 years ago
  16. e05bcc2 Do not switch a connection if the new connection is not ready to send packets. by Honghai Zhang · 8 years ago
  17. 49c01d7 Currently there is not way to programmically test whether a ScreenCapturer by zijiehe · 8 years ago
  18. 895e1a9 Change the default backup connection ping interval to 25 seconds. by Honghai Zhang · 8 years ago
  19. 287e548 Cleanup RtcpReceiver::TMMBRReceived function by danilchap · 8 years ago
  20. f095012 Revert of Adding audio to video_quality_test. (patchset #10 id:230001 of https://codereview.webrtc.org/2136573002/ ) by minyue · 8 years ago
  21. 65a6578 Adding audio to video_quality_test. by minyue · 8 years ago
  22. 75c287e Fix incorrect example in mod_ops.h by philipel · 8 years ago
  23. a06ce49 Run "git cl format" on some files before I start to modify them by kwiberg · 8 years ago
  24. b789439 Roll chromium_revision 2b53ee0889..915e47250f (411979:412201) by buildbot · 8 years ago
  25. 90920d5 GN: Enable msse2 flag in Mac. by ehmaldonado · 8 years ago
  26. 9d7eb13 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ ) by kwiberg · 8 years ago
  27. e252d3c MB: Fix incorrect iOS builder names. by kjellander · 8 years ago
  28. 427ce3d Move FilePlayer and FileRecorder to Voice Engine by kwiberg · 8 years ago
  29. 2f69ce9 Cleaned out candidateSet member from TMMBRHelp class by danilchap · 8 years ago
  30. 1c814e7 iOS: Update MB and JSON configs + enable Goma by kjellander · 8 years ago
  31. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  32. 6910537 Add gn target for audio_device_tests. by maxmorin · 8 years ago
  33. 70f866c Added new mixer to |check_targets| in .gn and fixed include/depend errors. by aleloi · 8 years ago
  34. 7522a28 Removed old probe cluster logic and logic related to ssrcs from DelayBasedBwe. by philipel · 8 years ago
  35. b7186d0 Migrated GN target :isac_fix_test by aleloi · 8 years ago
  36. b24b1ce Moving mock classes around so that they may be reused in other unittests by hbos · 8 years ago
  37. 88e31a3 Fix warnings, simplify ADM. by maxmorin · 8 years ago
  38. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  39. 82dda1a [WebRTC] Disable DirectX capturer tests if the system does not support it. by zijiehe · 8 years ago
  40. e1b4d24 Skip AUD while extracting SPS and PPS on iOS. by jianjun.zhu · 8 years ago
  41. 6c687e7 Make prior H264 QP adjustments iOS specific. by tkchin · 8 years ago
  42. 3473288 Remove VERBOSE logs in (android) audio device code. by noahric · 8 years ago
  43. 43ba317 Roll chromium_revision 4b42aa218b..2b53ee0889 (411951:411979) by buildbot · 8 years ago
  44. b1e6611 GN: Fix audio_decoder_unittests for android. by ehmaldonado · 8 years ago
  45. 4a1ec1e Added ProbeBitrate(bitrate_bps, num_probes) to BitrateProber. by philipel · 8 years ago
  46. 1aee0b5 Remove old methods in AudioTransport, make it pass a gn build by maxmorin · 8 years ago
  47. c8c71f4 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ ) by kwiberg · 8 years ago
  48. dc65ea2 Move FilePlayer and FileRecorder to Voice Engine by kwiberg · 8 years ago
  49. f96c51a GN: Add video_capture_tests for Mac by kjellander · 8 years ago
  50. 2a75801 Revert of CQ: Temporarily disable iOS Simulator trybots (patchset #1 id:1 of https://codereview.webrtc.org/2244183002/ ) by kjellander · 8 years ago
  51. e34c19c Clarify some function names in visualization tool. by terelius · 8 years ago
  52. 2ab1da7 Revert of Added new mixer to |check_targets| in .gn and fixed include/depend errors. (patchset #1 id:1 of https://codereview.webrtc.org/2234293002/ ) by olka · 8 years ago
  53. d700bef Added new mixer to |check_targets| in .gn and fixed include/depend errors. by aleloi · 8 years ago
  54. 963be23 RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods by kwiberg · 8 years ago
  55. 8f956de FakeTiming added, an implementation of Timing that can be used for tests. by hbos · 8 years ago
  56. 96dbc8f Adding comment regarding the disabling the flaky test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff by peah · 8 years ago
  57. 3ab6614 Add video_loopback to gn. by stefan · 8 years ago
  58. 92c0950 Make CameraCapturer.switchCamera try again if session is still opening. by sakal · 8 years ago
  59. d7d05f8 Disabling the test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff by peah · 8 years ago
  60. da07af2 Roll chromium_revision 7f405ec2b6..4b42aa218b (411933:411951) by buildbot · 8 years ago
  61. 3e3ebe6 remove unnecessary double allocation by kthelgason · 8 years ago
  62. 0ccff57 VoERTP_RTCP: Remove GetREDStatus and SetREDStatus by kwiberg · 8 years ago
  63. 5bcc00e Changed folder structure in new mixer and fixed simple lint errors. by aleloi · 8 years ago
  64. 714dd4e GN: Update tests to have the correct shard timeout value on Android. by sakal · 8 years ago
  65. 5093b38 Make variable for selecting if intervals without samples should be included in stats configurable (for rate counters). by asapersson · 8 years ago
  66. c61ae74 Roll chromium_revision 941118827f..7f405ec2b6 (411223:411933) by buildbot · 8 years ago
  67. 414eb18 CQ: Temporarily disable iOS Simulator trybots by kjellander · 8 years ago
  68. 4cb5b64 Fix for data channels perpetually stuck in "closing" state. by Taylor Brandstetter · 8 years ago
  69. 64a7eab Update tests and DTX check for Opus 1.1.3. by flim · 8 years ago
  70. 9591e3e Convert PeerConnectionTest to use the new capture APIs. by sakal · 8 years ago
  71. 62351c9 Fixing problems with ICE candidate pair prioritization. by Taylor Brandstetter · 8 years ago
  72. 6f82535 Enabling IPv6 socket recv timestamp test, and making less flaky. by Taylor Brandstetter · 8 years ago
  73. 588783a Return nil from RTCPeerConnectionFactory when creation fails by skvlad · 8 years ago
  74. fe1ffb1 Remove unused SessionId from TransportChannel and PortAllocatorSession. by johan · 8 years ago
  75. c8762a8 Remove StartSSLWithServer from SSLStreamAdapter. by Taylor Brandstetter · 8 years ago
  76. f10976e Roll chromium_revision db8d32de07..941118827f (410624:411223) by kjellander · 8 years ago
  77. 3b74768 Remove pbos@webrtc.org from WATCHLISTS. by Peter Boström · 8 years ago
  78. 2e5cfcd Add periodic logging of video stats. by asapersson · 8 years ago
  79. b179767 Add an HD resolution perf test. by stefan · 8 years ago
  80. 17deeb4 PacketBuffer is now ref counted. by philipel · 8 years ago
  81. a3a1fde Add Mac bots to MB. by ehmaldonado · 8 years ago
  82. d30e0ad Session based capturing for Camera2Capturer. by sakal · 8 years ago
  83. bd59c71 GN: Add dependency libjingle_peerconnection_java to modules_unittests. by sakal · 8 years ago
  84. 0ae7878 MB: Add Windows configurations by kjellander · 8 years ago
  85. 3d31bd6 Do not create incompatible TurnPort if the server address family is known. by Honghai Zhang · 8 years ago
  86. bf8a2c9 Probe bitrate estimator correction. by philipel · 8 years ago
  87. 68815bf MB: Make all Android debug builds static by kjellander · 8 years ago
  88. c99d5a6 Add stefan@ to webrtc/OWNERS. by stefan · 8 years ago
  89. 63cb172 MB: Fix typo for android_arm64_rel trybot. by kjellander · 8 years ago
  90. fb372f0 iOS render: Handle frame rotation in OpenGL by magjed · 8 years ago
  91. 4556b45 Fix tools_unittests in GN. by ehmaldonado · 8 years ago
  92. ccbbf8d Visualize delay changes based on both abs-send-time and capture time. by terelius · 8 years ago
  93. d49a37b Rename main file for visualization tool. by terelius · 8 years ago
  94. c4ac700 Migrated GN target :neteq_pcmu_quality_test by aleloi · 8 years ago
  95. 6df36dc Migrated GN target :neteq_isac_quality_test by aleloi · 8 years ago
  96. e6ca9ec Broke out 'level_indicator' of the voice_engine GN target. This is by aleloi · 8 years ago
  97. 0e0be0a Migrated GN target :neteq_ilbc_quality_test by aleloi · 8 years ago
  98. 6e6e70f Add magjed@webrtc.org as owner of webrtc/base/java/ by Magnus Jedvert · 8 years ago
  99. 6391012 Migrated GN target :audio_classifier_test by aleloi · 8 years ago
  100. bcdad0f Generate random rtp packets with RtpPacketToSend instead of RtpSender by Danil Chapovalov · 8 years ago