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gerrit-public.fairphone.software / platform / external / webrtc / 7851bda9bc1df4f1efbaf8eba66bce4abe43005d
  1. 7851bda Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver by danilchap · 8 years ago
  2. c8d2171 Replace RelayPort with TurnPort in p2ptransportchannel tests. by Honghai Zhang · 8 years ago
  3. 7502401 Do not spam "Connect failed with 101/65" in logs. by skvlad · 8 years ago
  4. 591c709 Suppress a memcheck error in Opus decoder by henrik.lundin · 8 years ago
  5. 590cf28 Add autothread to pseudo-tcp fuzzer. by phoglund · 8 years ago
  6. 70736e4 Remove old presumably unused directory. by sakal · 8 years ago
  7. 8e6a761 ProbeController: Limit max probing bitrate by isheriff · 8 years ago
  8. 6060186 Add presubmit format requirement for webrtc/api/android by magjed · 8 years ago
  9. 5614566 Fix faulty include paths that break the build by Henrik Lundin · 8 years ago
  10. 5ec85fb Revert of Fix race / crash in OnNetworkRouteChanged(). (patchset #5 id:80001 of https://codereview.webrtc.org/2366333003/ ) by stefan · 8 years ago
  11. b6760f9 Format all Java in WebRTC. by sakal · 8 years ago
  12. a48ddb7 Add VideoSendStream::Stats::prefered_media_bitrate_bps by Per · 8 years ago
  13. fd0d426 Fix race / crash in OnNetworkRouteChanged(). by stefan · 8 years ago
  14. eddb757 Revert of Unify the macOS and iOS capturer implementations (patchset #4 id:60001 of https://codereview.webrtc.org/2309253005/ ) by kthelgason · 8 years ago
  15. ff9793c Android: Remove onOutputFormatRequest from the VideoCapturer interface by magjed · 8 years ago
  16. 90ce01d The current default schedule delay of 30 ms prohibits by isheriff · 8 years ago
  17. 0fd22ef Rename P2PTransportChannel worker_thread_ to network_thread_. by johan · 8 years ago
  18. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  19. 51f2919 Update WebRTC to build against libsrtp 2.0 by mattdr · 8 years ago
  20. 24c7c12 Move FunctionView from AudioCodingModule to the rtc namespace by kwiberg · 8 years ago
  21. 35d43b9 Roll chromium_revision bdaa23ddfe..316b880c55 (421490:421519) by ehmaldonado · 8 years ago
  22. 7e146cb Fixing heap read overflow when "sctp-port" is in a video description. by deadbeef · 8 years ago
  23. 478681e Move the QP scaling thresholds to the relevant encoders. by kthelgason · 8 years ago
  24. e75f204 Expose Ivf logging through the native API by palmkvist · 8 years ago
  25. 242d8bd Unify the macOS and iOS capturer implementations by kthelgason · 8 years ago
  26. f5e3bbe Roll chromium_revision 386676ff4e..bdaa23ddfe (421470:421490) by buildbot · 8 years ago
  27. e5684c5 Delete method webrtc::VideoFrame::allocated_size and enum PlaneType. by nisse · 8 years ago
  28. 798896a Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo by danilchap · 8 years ago
  29. 9a8abcb Roll chromium_revision dd442d4812..386676ff4e (421425:421470) by ehmaldonado · 8 years ago
  30. 8fea199 [GN] Add missing framework headers by kthelgason · 8 years ago
  31. e0b2f15 Frame continuity is now tested as soon as a frame is inserted into the FrameBuffer. by philipel · 8 years ago
  32. 89a3a1a Moved Gn target rtc_event_log to one directory above. by charujain · 8 years ago
  33. b7446d7 GN: Fix incorrect include_dir for libjingle_peerconnection_jni target by charujain · 8 years ago
  34. f363d14 Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425) by Henrik Kjellander · 8 years ago
  35. 0c9e567 Landmine to clobber on Android and Windows. by kjellander · 8 years ago
  36. 5e3b5d1 CQ: Remove GYP Release trybots since we now only run GYP. by kjellander · 8 years ago
  37. e5e632f Hooking up target audio bitrate to audio network adaptor. by minyue · 8 years ago
  38. 72bebf1 Roll chromium_revision cede888c27..f86fb54ec3 (419407:420104) by buildbot · 8 years ago
  39. c3f549b Update expected Xcode version to 8.0. by kjellander · 8 years ago
  40. ee99696 Make 'webrtc' a static library. by kjellander · 8 years ago
  41. 822a16f Reland of Unify rtcp packet setters (patchset #1 id:1 of https://codereview.webrtc.org/2372713005/ ) by danilchap · 8 years ago
  42. 4151471 Add usage description strings to Info.plist by Kári Tristan Helgason · 8 years ago
  43. efc6e41 Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ ) by kjellander · 8 years ago
  44. 9532124 RTCPReceiver store cname as std::string. simplifying cname management. by danilchap · 8 years ago
  45. f1363fd Adds support for AVAudioSessionSilenceSecondaryAudioHintNotification on iOS by henrika · 8 years ago
  46. 46a8d18 ACM: Removed the code for InitialDelayManager by ossu · 8 years ago
  47. 29a44e3 This is a resubmission of https://codereview.webrtc.org/2047513002/ by kthelgason · 8 years ago
  48. 5f8ebae Add limitations of number of frames that can be created in I420BufferPool::CreateBuffer. by perkj · 8 years ago
  49. c8299f9 Posting Opus's set-force-channels functionality to WebRTC. by minyue · 8 years ago
  50. 20e77c7 Unify rtcp packet setters by danilchap · 8 years ago
  51. 4ecd970 GN: Fix incorrect include_dir for video_coding on iOS by kjellander · 8 years ago
  52. c1815cf Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ ) by Magnus Jedvert · 8 years ago
  53. 0a52c70 THis CL enables possibility to select full-duplex OpenSL ES audio in AppRTCDemo, i.e., it adds support for OpenSL ES for input as well. The user must explicitly select this new mode in the debug UI hence it is not the default selection. There is no separate UI for input and output; instead both are enabled/disabled by the same switch. by henrika · 8 years ago
  54. 64ec8f8 Reland of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #1 id:1 of https://codereview.webrtc.org/2354223002/ ) by nisse · 8 years ago
  55. c637389 Delete unused file mock_audio_vector.h. by nisse · 8 years ago
  56. de2920c Delete unused file sessionid.h. by nisse · 8 years ago
  57. 89175a6 Trust that calls to RemoteEstimatorProxy::Process are done at the right frequency. by stefan · 8 years ago
  58. 25337bb Android: Update clang-format to follow Google style guide by magjed · 8 years ago
  59. fd8e33d Removing a useless ctor in AudioNetworkAdaptorImpl. by minyue · 8 years ago
  60. 8af4fd0 Disabled flaky VideoSendStreamTest.ChangingNetworkRoute by hbos · 8 years ago
  61. 98088dc header_usage.sh script: Exclude matches in gyp and gn files. by nisse · 8 years ago
  62. 660312b Enable //build/config/clang:extra_warnings for rtc_media by kjellander · 8 years ago
  63. 464382d Remove duplicated entry for bwe_simulations.cc by kjellander · 8 years ago
  64. 3901128 Remove unnecessary jsoncpp includes. by kjellander · 8 years ago
  65. 2068411 r14326 added '-Wno-unused-result' to 'WARNING_CFLAGS!' which removes the by jianjun.zhu · 8 years ago
  66. c59bf04 Remove differ from ScreenCapturer implementations by zijiehe · 8 years ago
  67. d3d230f - Make RtpSenderAudio not inherit from DtmfQueue. by solenberg · 8 years ago
  68. 92ea601 Move class RTCPHelp::RTCPPacketInformation into RTCPReceiver by danilchap · 8 years ago
  69. dda3666 Fixes minor issue in AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex for iOS. by henrika · 8 years ago
  70. f8c5f2b Fix vie_encoder_unittest.cc. by Per · 8 years ago
  71. 44428a8 iOS: Always build H264 HW encoder/decoder by magjed · 8 years ago
  72. 512ecb3 Let ViEEncoder tell VideoSendStream about reconfigurations. by Per · 8 years ago
  73. 2a27b0a Delete unused class FakeScreenCapturerFactory. by nisse · 8 years ago
  74. 347ec5c Change thread check to race check. Also, add comment to explain implementation of RaceChecker. by solenberg · 8 years ago
  75. f1b08da Stopped using the NetEqDecoder enum internally in NetEq. by ossu · 8 years ago
  76. 1490f7a Add histogram for end-to-end delay: "WebRTC.Video.EndToEndDelayInMs" by asapersson · 8 years ago
  77. 6d4c8c3 Renaming a proto target in GYP for audio network adaptor. by minyue · 8 years ago
  78. e87d673 Return texture frame when dropping frames in CameraCapturer. by sakal · 8 years ago
  79. b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
  80. 25f6a39 Relanding of "Adding debug dump to audio network adaptor." by minyue · 8 years ago
  81. 161b390 Revert of Adding debug dump to audio network adaptor. (patchset #5 id:140001 of https://codereview.webrtc.org/2356763002/ ) by minyue · 8 years ago
  82. 7e4f892 Adding debug dump to audio network adaptor. by minyue · 8 years ago
  83. a78213e Add tools/determinism to setup_links. by ehmaldonado · 8 years ago
  84. 893a7ee Support more QCOM specific color formats for Android HW decoder. by glaznev · 8 years ago
  85. 87ef6f7 Revert of Rename AppRTCDemo on Android and iOS to AppRTCMobile (patchset #2 id:20001 of https://codereview.webrtc.org/2343403002/ ) by magjed · 8 years ago
  86. 3e02430 Fix a stun attribute leak. by Honghai Zhang · 8 years ago
  87. d3af58b Rename AppRTCDemo on Android and iOS to AppRTCMobile by magjed · 8 years ago
  88. 051d151 Adds audio session status to logs for each valid audio route change on iOS by henrika · 8 years ago
  89. c37e983 Add custom info.plist to modules_unittests by kthelgason · 8 years ago
  90. f292e31 Relax too strict DCHECKs while parsing rtcp reports by danilchap · 8 years ago
  91. aac9d6f Added graph for plotting the audio level from an Rtc event log. by ivoc · 8 years ago
  92. d0ede44 Adding FecController to audio network adaptor. by minyue · 8 years ago
  93. 1c6d3f7 Add new Camera2Enumerator.isSupported with Context parameter. by sakal · 8 years ago
  94. 09baefe GN: Remove non-existing header file: media/base/videorenderer.h by charujain · 8 years ago
  95. 799a9d0 Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ ) by danilchap · 8 years ago
  96. a70695a Moved Opus-specific payload splitting into AudioDecoderOpus. by ossu · 8 years ago
  97. 2beb429 Remove unnecessary interface TelephoneEventHandler. by solenberg · 8 years ago
  98. 6f5a6c3 New class AdaptedVideoTrackSource. by nisse · 8 years ago
  99. bc77ed7 Adding reordering logic in audio network adaptor. by minyue · 8 years ago
  100. 4aec1d4 Relanding of "Adding BitrateController to audio network adaptor." by minyue · 8 years ago