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webrtc / src / 91a5759733acb58646d5e6269a7667e322bb55d8
  1. 91a5759 Roll chromium_revision 316b880c55..2cabef4e7d (421519:424936) by Henrik Kjellander · 9 years ago
  2. 163b1a2 Remove tools dir from root webrtc target by charujain · 9 years ago
  3. 614f68f Remove duplicate entry in webrtc .gn file exec_script_whitelist by fbarchard · 9 years ago
  4. db158f9 Fix experiment name in BitrateControllerTest. by Stefan Holmer · 9 years ago
  5. 77c663d Give FeedbackTimeout experiment the correct name. by stefan · 9 years ago
  6. 12a39f4 Don't crash on unexpected stap-a or fu-a. by stefan · 9 years ago
  7. 75c8fb4 DataChannelInterface default impl of [messages/bytes]_[sent/received]. by hbos · 9 years ago
  8. 84ffdee DataChannel[Interface]::[message/bytes]_[sent/received]() added. by hbos · 9 years ago
  9. 73fdc31 Several fixes to screen_capturer_mac. by erikchen · 9 years ago
  10. e280cde Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz. by ossu · 9 years ago
  11. 872f614 Android audio playout now supports non-call media streams. by henrika · 9 years ago
  12. 116ec6d Implemented further mixer interface change suggestions from https://codereview.webrtc.org/2386383003/ by aleloi · 9 years ago
  13. 7e30432 Hooking up audio network adaptor to VoE. by minyue · 9 years ago
  14. 917d4e1 Removed the legacy behavior of stopping playout when setting new receive codecs. by solenberg · 9 years ago
  15. e97974d Cleanup of the mixer interface. by aleloi · 9 years ago
  16. 73a28ee The AudioProcessing class is used as an interface by peah · 9 years ago
  17. 4b8bfb8 Changed ramping functionality of the AudioMixer. by aleloi · 9 years ago
  18. 3ec3da6 Set screenshare.enabled parameter to false when running video_loopback test. by asapersson · 9 years ago
  19. c6ca544 Using relative path for GN for iOS. by nicholss · 9 years ago
  20. c47a0c3 RTCIceCandidatePairStats[1] added. by hbos · 9 years ago
  21. dd0e1e0 GN: Build iOS framework in build_ios_libs.sh by Zeke Chin · 9 years ago
  22. e7c338f Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." (patchset #1 id:1 of https://codereview.webrtc.org/2402993002/ ) by sprang · 9 years ago
  23. 716978d Revert of Prune connections based on network name. (patchset #3 id:130001 of https://codereview.webrtc.org/2395243005/ ) by sprang · 9 years ago
  24. e891415 Moved MixerAudioSource and removed audio_mixer_defines.h. by aleloi · 9 years ago
  25. 14acf65 AudioTransport::NeedMorePlayData is no longer called from different threads using OpenSL ES on Android by henrika · 9 years ago
  26. 99df6c0 Fix bug in DTMF generation where events with level > 36 would be ignored. by solenberg · 9 years ago
  27. 113bdca Make sure VideoReceiveStream can be restarted by sprang · 9 years ago
  28. defc21e Removes usage of hardware AGC and any related APIs on Android. by henrika · 9 years ago
  29. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 9 years ago
  30. a8b3855 Add a FlexfecReceiver class. by brandtr · 9 years ago
  31. aae2784 Prune connections based on network name. by honghaiz · 9 years ago
  32. 8ff52cc Remove useless debugging code by mattdr · 9 years ago
  33. 8cab52d Fix externalhmac.h/.cc to compile with libsrtp 1 and 2 by mattdr · 9 years ago
  34. 440b4be Use non-deprecated screen update callbacks. by erikchen · 9 years ago
  35. e606a17 Remove deprecated Gestalt methods. by erikchen · 9 years ago
  36. 41aab32 Fix delay plot crash in event_log_visualizer. by stefan · 9 years ago
  37. ae0b333 Prep to remove APM-related #defines from voice_engine_configurations.h by henrik.lundin · 9 years ago
  38. 57cb873 Revert of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." (patchset #1 id:1 of https://codereview.webrtc.org/2361053003/ ) by sprang · 9 years ago
  39. e0928d8 Added logging for audio send/receive stream configs. by ivoc · 9 years ago
  40. fc9414a Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." by johan · 9 years ago
  41. d36dd49 Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ ) by kjellander · 9 years ago
  42. cbfb033 MB: Update Android bots after moving to Swarming and client.webrtc.perf by Henrik Kjellander · 9 years ago
  43. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 9 years ago
  44. 9e31cca iOS: Fix rotation bug for back camera by magjed · 9 years ago
  45. c19f312 This CL adds functionality in the level controller to by peah · 9 years ago
  46. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 9 years ago
  47. 0d348d6 Avoid race in VideoReceiveStream shutdown by sprang · 9 years ago
  48. dda6ec0 Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete by nisse · 9 years ago
  49. 0d382ef Cleaning build file for audio network adaptor. by minyue · 9 years ago
  50. bf369fe Replace rtcp parser in rtc event log handlers. RTCPUtility is going away. by danilchap · 9 years ago
  51. 05f3ec1 Fix "left shift of negative value" bug by kwiberg · 9 years ago
  52. d2cf1ce Remove legacy Camera2Enumerator.isSupported method. by sakal · 9 years ago
  53. c7901c6 Delete macsocketserver.h and related files. by nisse · 9 years ago
  54. df494b0 Android: Split out EGL rendering from SurfaceViewRenderer to separate class by magjed · 9 years ago
  55. 3654251 Reland of https://codereview.webrtc.org/2396483002/ by aleloi · 9 years ago
  56. a485dab Revert of Made MixerAudioSource a pure interface. (patchset #7 id:350001 of https://codereview.webrtc.org/2396483002/ ) by aleloi · 9 years ago
  57. 2ae5fdf Made MixerAudioSource a pure interface. by aleloi · 9 years ago
  58. 2334b70 Add support for component builds on Windows Swarming. Attempt 2 by ehmaldonado · 9 years ago
  59. ab9f6e4 RTCIceCandidateStats[1] added. by hbos · 9 years ago
  60. 3dcfd64 Make bwe_simulator a separate test target that doesnt run on the bots by terelius · 9 years ago
  61. 2334559 Revert of Add support for component builds on Windows Swarming. (patchset #2 id:100001 of https://codereview.webrtc.org/2389133002/ ) by ehmaldonado · 9 years ago
  62. 66d4ce9 Add support for component builds on Windows Swarming. by ehmaldonado · 9 years ago
  63. ae04e36 Cleanup unused dependency on video_capture_module. by perkj · 9 years ago
  64. e402a14 Make process interval configurable for MaxCounter class. by asapersson · 9 years ago
  65. 9cc012d GN: Add missing config //webrtc:common_objc for rtc_sdk_common_objc by kjellander · 9 years ago
  66. 7056be9 Delete old video defines in engine config. by mflodman · 9 years ago
  67. 0b4b727 Use NtpTime in RtcpReceiver instead of pair of uints by danilchap · 9 years ago
  68. 15d8357 Remove OnLocalSsrcChanged and rename EncoderStateFeedback. by mflodman · 9 years ago
  69. 41b9c80 Adding audio network adaptor to AudioEncoderOpus. by minyue · 9 years ago
  70. 81b9291 The FFT functionality in aec_rdft* is based on legacy C by peah · 9 years ago
  71. 21a18ee Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ ) by ivoc · 9 years ago
  72. 327e9d0 Make MediaCodecEncoder fallback to a software encoder on failure. by sakal · 9 years ago
  73. a669a3a Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )" by stefan · 9 years ago
  74. 0e7c7ce Delete webrtc::VideoFrame::CopyFrame. by nisse · 9 years ago
  75. 93e451b Skip non-continuous FrameInfos when finding the next frame to return from FrameBuffer::NextFrame. by philipel · 9 years ago
  76. 55b9544 Add camera start/stop time stats to VideoCapturerAndroid. by sakal · 9 years ago
  77. 4f15ca5 Replaces the SessionPresets with AVCaptureDeviceFormats. by denicija · 9 years ago
  78. 68979ab H264 codec: Check profile-level-id when matching by magjed · 9 years ago
  79. fc5e050 rtc_stats: Update code to remove chromium style warnings suppression. by hbos · 9 years ago
  80. bf7c620 Fix lint errors in audio_coding_module_unittest.cc by henrik.lundin · 9 years ago
  81. 8e7577c Revert of Delete transformadapter.cc and transformadapter.h. (patchset #2 id:20001 of https://codereview.webrtc.org/2368243002/ ) by nisse · 9 years ago
  82. 2504c0a Drop _oldapi from ACM test file names by henrik.lundin · 9 years ago
  83. 2c3c3e2 Delete transformadapter.cc and transformadapter.h. by nisse · 9 years ago
  84. f156bdd Add onCameraDisconnected camera event. by sakal · 9 years ago
  85. 11d37ba Speculatively increase timeouts for DirectRTCClientTest to reduce flakiness. by sakal · 9 years ago
  86. 4e8baf4 Delete unused code, cpuinfo.{h,cc}. by nisse · 9 years ago
  87. daf88b1 Removing ERROR message for something that's expected to occur. by deadbeef · 9 years ago
  88. a84aa57 Use std::abs instead of C-style abs. by jianjun.zhu · 9 years ago
  89. 2ca8d5c Remove deprecated FSRef-manipulation methods. by erikchen · 9 years ago
  90. efc5ae9 Fixed flaky SharedExclusiveLock tests. by skvlad · 9 years ago
  91. 134af7a Disable flaky Android capture tests: by skvlad · 9 years ago
  92. 5fa51e2 Add iOS static library GN build script. NOTRY=True by tkchin · 9 years ago
  93. d93f50c Add UMA metrics for ICE regathering reasons. by Honghai Zhang · 9 years ago
  94. 425a6cc RTPReceiverAudio: Removed frequency from CNGPayloadType and cleaned up by ossu · 9 years ago
  95. b2d1e0d Resurrected test_api_audio.cc by ossu · 9 years ago
  96. b1fff92 Harmonize using-declarations in FEC header formatter fuzzers. by brandtr · 9 years ago
  97. 28b03eb Move RTCPHelp::RTCPReportBlockInformation into RTCPReceiver by danilchap · 9 years ago
  98. 62b1c35 Android: Move layout measure code from SurfaceViewRenderer to RendererCommon by Magnus Jedvert · 9 years ago
  99. 3360352 Make sure vp9 actually gets excluded in gn as well. by phoglund · 9 years ago
  100. 3dacb53 Add audio_mixer to WATCHLISTS by henrik.lundin · 9 years ago