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gerrit-public.fairphone.software / platform / external / webrtc / dd0e1e007092a6a87a3abf994c69e0028229811c
  1. dd0e1e0 GN: Build iOS framework in build_ios_libs.sh by Zeke Chin · 8 years ago
  2. e7c338f Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." (patchset #1 id:1 of https://codereview.webrtc.org/2402993002/ ) by sprang · 8 years ago
  3. 716978d Revert of Prune connections based on network name. (patchset #3 id:130001 of https://codereview.webrtc.org/2395243005/ ) by sprang · 8 years ago
  4. e891415 Moved MixerAudioSource and removed audio_mixer_defines.h. by aleloi · 8 years ago
  5. 14acf65 AudioTransport::NeedMorePlayData is no longer called from different threads using OpenSL ES on Android by henrika · 8 years ago
  6. 99df6c0 Fix bug in DTMF generation where events with level > 36 would be ignored. by solenberg · 8 years ago
  7. 113bdca Make sure VideoReceiveStream can be restarted by sprang · 8 years ago
  8. defc21e Removes usage of hardware AGC and any related APIs on Android. by henrika · 8 years ago
  9. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  10. a8b3855 Add a FlexfecReceiver class. by brandtr · 8 years ago
  11. aae2784 Prune connections based on network name. by honghaiz · 8 years ago
  12. 8ff52cc Remove useless debugging code by mattdr · 8 years ago
  13. 8cab52d Fix externalhmac.h/.cc to compile with libsrtp 1 and 2 by mattdr · 8 years ago
  14. 440b4be Use non-deprecated screen update callbacks. by erikchen · 8 years ago
  15. e606a17 Remove deprecated Gestalt methods. by erikchen · 8 years ago
  16. 41aab32 Fix delay plot crash in event_log_visualizer. by stefan · 8 years ago
  17. ae0b333 Prep to remove APM-related #defines from voice_engine_configurations.h by henrik.lundin · 8 years ago
  18. 57cb873 Revert of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." (patchset #1 id:1 of https://codereview.webrtc.org/2361053003/ ) by sprang · 8 years ago
  19. e0928d8 Added logging for audio send/receive stream configs. by ivoc · 8 years ago
  20. fc9414a Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." by johan · 8 years ago
  21. d36dd49 Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ ) by kjellander · 8 years ago
  22. cbfb033 MB: Update Android bots after moving to Swarming and client.webrtc.perf by Henrik Kjellander · 8 years ago
  23. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  24. 9e31cca iOS: Fix rotation bug for back camera by magjed · 8 years ago
  25. c19f312 This CL adds functionality in the level controller to by peah · 8 years ago
  26. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  27. 0d348d6 Avoid race in VideoReceiveStream shutdown by sprang · 8 years ago
  28. dda6ec0 Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete by nisse · 8 years ago
  29. 0d382ef Cleaning build file for audio network adaptor. by minyue · 8 years ago
  30. bf369fe Replace rtcp parser in rtc event log handlers. RTCPUtility is going away. by danilchap · 8 years ago
  31. 05f3ec1 Fix "left shift of negative value" bug by kwiberg · 8 years ago
  32. d2cf1ce Remove legacy Camera2Enumerator.isSupported method. by sakal · 8 years ago
  33. c7901c6 Delete macsocketserver.h and related files. by nisse · 8 years ago
  34. df494b0 Android: Split out EGL rendering from SurfaceViewRenderer to separate class by magjed · 8 years ago
  35. 3654251 Reland of https://codereview.webrtc.org/2396483002/ by aleloi · 8 years ago
  36. a485dab Revert of Made MixerAudioSource a pure interface. (patchset #7 id:350001 of https://codereview.webrtc.org/2396483002/ ) by aleloi · 8 years ago
  37. 2ae5fdf Made MixerAudioSource a pure interface. by aleloi · 8 years ago
  38. 2334b70 Add support for component builds on Windows Swarming. Attempt 2 by ehmaldonado · 8 years ago
  39. ab9f6e4 RTCIceCandidateStats[1] added. by hbos · 8 years ago
  40. 3dcfd64 Make bwe_simulator a separate test target that doesnt run on the bots by terelius · 8 years ago
  41. 2334559 Revert of Add support for component builds on Windows Swarming. (patchset #2 id:100001 of https://codereview.webrtc.org/2389133002/ ) by ehmaldonado · 8 years ago
  42. 66d4ce9 Add support for component builds on Windows Swarming. by ehmaldonado · 8 years ago
  43. ae04e36 Cleanup unused dependency on video_capture_module. by perkj · 8 years ago
  44. e402a14 Make process interval configurable for MaxCounter class. by asapersson · 8 years ago
  45. 9cc012d GN: Add missing config //webrtc:common_objc for rtc_sdk_common_objc by kjellander · 8 years ago
  46. 7056be9 Delete old video defines in engine config. by mflodman · 8 years ago
  47. 0b4b727 Use NtpTime in RtcpReceiver instead of pair of uints by danilchap · 8 years ago
  48. 15d8357 Remove OnLocalSsrcChanged and rename EncoderStateFeedback. by mflodman · 8 years ago
  49. 41b9c80 Adding audio network adaptor to AudioEncoderOpus. by minyue · 8 years ago
  50. 81b9291 The FFT functionality in aec_rdft* is based on legacy C by peah · 8 years ago
  51. 21a18ee Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ ) by ivoc · 8 years ago
  52. 327e9d0 Make MediaCodecEncoder fallback to a software encoder on failure. by sakal · 8 years ago
  53. 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 · 8 years ago
  54. 0e7c7ce Delete webrtc::VideoFrame::CopyFrame. by nisse · 8 years ago
  55. 93e451b Skip non-continuous FrameInfos when finding the next frame to return from FrameBuffer::NextFrame. by philipel · 8 years ago
  56. 55b9544 Add camera start/stop time stats to VideoCapturerAndroid. by sakal · 8 years ago
  57. 4f15ca5 Replaces the SessionPresets with AVCaptureDeviceFormats. by denicija · 8 years ago
  58. 68979ab H264 codec: Check profile-level-id when matching by magjed · 8 years ago
  59. fc5e050 rtc_stats: Update code to remove chromium style warnings suppression. by hbos · 8 years ago
  60. bf7c620 Fix lint errors in audio_coding_module_unittest.cc by henrik.lundin · 8 years ago
  61. 8e7577c Revert of Delete transformadapter.cc and transformadapter.h. (patchset #2 id:20001 of https://codereview.webrtc.org/2368243002/ ) by nisse · 8 years ago
  62. 2504c0a Drop _oldapi from ACM test file names by henrik.lundin · 8 years ago
  63. 2c3c3e2 Delete transformadapter.cc and transformadapter.h. by nisse · 8 years ago
  64. f156bdd Add onCameraDisconnected camera event. by sakal · 8 years ago
  65. 11d37ba Speculatively increase timeouts for DirectRTCClientTest to reduce flakiness. by sakal · 8 years ago
  66. 4e8baf4 Delete unused code, cpuinfo.{h,cc}. by nisse · 8 years ago
  67. daf88b1 Removing ERROR message for something that's expected to occur. by deadbeef · 8 years ago
  68. a84aa57 Use std::abs instead of C-style abs. by jianjun.zhu · 8 years ago
  69. 2ca8d5c Remove deprecated FSRef-manipulation methods. by erikchen · 8 years ago
  70. efc5ae9 Fixed flaky SharedExclusiveLock tests. by skvlad · 8 years ago
  71. 134af7a Disable flaky Android capture tests: by skvlad · 8 years ago
  72. 5fa51e2 Add iOS static library GN build script. NOTRY=True by tkchin · 8 years ago
  73. d93f50c Add UMA metrics for ICE regathering reasons. by Honghai Zhang · 8 years ago
  74. 425a6cc RTPReceiverAudio: Removed frequency from CNGPayloadType and cleaned up by ossu · 8 years ago
  75. b2d1e0d Resurrected test_api_audio.cc by ossu · 8 years ago
  76. b1fff92 Harmonize using-declarations in FEC header formatter fuzzers. by brandtr · 8 years ago
  77. 28b03eb Move RTCPHelp::RTCPReportBlockInformation into RTCPReceiver by danilchap · 8 years ago
  78. 62b1c35 Android: Move layout measure code from SurfaceViewRenderer to RendererCommon by Magnus Jedvert · 8 years ago
  79. 3360352 Make sure vp9 actually gets excluded in gn as well. by phoglund · 8 years ago
  80. 3dacb53 Add audio_mixer to WATCHLISTS by henrik.lundin · 8 years ago
  81. 58466f6 Relanding "Setting up an RTP input fuzzer for NetEq" by henrik.lundin · 8 years ago
  82. d020f3f Support for parsing CABAC coded bitstreams by kthelgason · 8 years ago
  83. 1f5a368 Readded a check to see if STRICT_HTTP_ERROR is defined before using it's value. by ivoc · 8 years ago
  84. de3f844 Fixed flaky clock_unittest by using relative comparison. by skvlad · 8 years ago
  85. 81e007a Delete profiler.h. by nisse · 8 years ago
  86. 3324cf7 Revert of Delete unused code httprequest, httpclient, and socketpool. (patchset #3 id:40001 of https://codereview.webrtc.org/2366333002/ ) by ivoc · 8 years ago
  87. 8a72369 Delete multipart.h. by nisse · 8 years ago
  88. 3d7901d Delete httpportallocator.h. by nisse · 8 years ago
  89. e643c59 Delete left-over SignalRenderFrame signal. by nisse · 8 years ago
  90. 4a255be Delete unused code httprequest, httpclient, and socketpool. by nisse · 8 years ago
  91. b5f2c3f Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
  92. f122a85 Delete webrtc::VideoFrame::CreateEmptyFrame. by nisse · 8 years ago
  93. 91e1a1a Delete winfirewall.h. by nisse · 8 years ago
  94. 2de66fe Delete unused file versionparsing.h. by nisse · 8 years ago
  95. b20f387 RTCStatsMember<bool> and RTCStatsMember<std::vector<bool>> added. by hbos · 8 years ago
  96. 5377bc7 Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere by kwiberg · 8 years ago
  97. cd97b2b Switched flaky ProfilerTest to use fake clock by skvlad · 8 years ago
  98. 5adaf73 AudioCodingModule: Specify decoders using SdpAudioFormat by kwiberg · 8 years ago
  99. 3168781 Add AlrDetector by isheriff · 8 years ago
  100. 52f6f20 Disable VideoSendStreamTest.ChangingNetworkRoute and VideoSendStreamTest.EncoderSetupPropagatesVp8Config on WinDrMemoryFull. by ivoc · 8 years ago