Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 280de9e1c33efa420fa22496d313c232093e093f
  1. 280de9e Reland: Fix race / crash in OnNetworkRouteChanged(). by Stefan Holmer · 8 years ago
  2. 20a52e1 Reland of Unify the macOS and iOS capturer implementations (patchset #1 id:1 of https://codereview.webrtc.org/2381853002/ ) by kthelgason · 8 years ago
  3. edbae5e Remove Crit::Scope lock by using atomic bool property. by denicija · 8 years ago
  4. eb5040a Disable TCPChannelClientTest.testConnectIPv6 by ehmaldonado · 8 years ago
  5. 15e4ec3 Remove compat for iOS 7/8 by Kári Tristan Helgason · 8 years ago
  6. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  7. b73d269 Replace RelayPort with TurnPort in p2ptransportchannel tests. by Honghai Zhang · 8 years ago
  8. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  9. 8b8459e suppress memcheck test errors in the Opus decoder and encoder. by honghaiz · 8 years ago
  10. 75f6626 Revert of Replace RelayPort with TurnPort in p2ptransportchannel tests. (patchset #2 id:40001 of https://codereview.webrtc.org/2380923002/ ) by honghaiz · 8 years ago
  11. 7851bda Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver by danilchap · 8 years ago
  12. c8d2171 Replace RelayPort with TurnPort in p2ptransportchannel tests. by Honghai Zhang · 8 years ago
  13. 7502401 Do not spam "Connect failed with 101/65" in logs. by skvlad · 8 years ago
  14. 591c709 Suppress a memcheck error in Opus decoder by henrik.lundin · 8 years ago
  15. 590cf28 Add autothread to pseudo-tcp fuzzer. by phoglund · 8 years ago
  16. 70736e4 Remove old presumably unused directory. by sakal · 8 years ago
  17. 8e6a761 ProbeController: Limit max probing bitrate by isheriff · 8 years ago
  18. 6060186 Add presubmit format requirement for webrtc/api/android by magjed · 8 years ago
  19. 5614566 Fix faulty include paths that break the build by Henrik Lundin · 8 years ago
  20. 5ec85fb Revert of Fix race / crash in OnNetworkRouteChanged(). (patchset #5 id:80001 of https://codereview.webrtc.org/2366333003/ ) by stefan · 8 years ago
  21. b6760f9 Format all Java in WebRTC. by sakal · 8 years ago
  22. a48ddb7 Add VideoSendStream::Stats::prefered_media_bitrate_bps by Per · 8 years ago
  23. fd0d426 Fix race / crash in OnNetworkRouteChanged(). by stefan · 8 years ago
  24. 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
  25. ff9793c Android: Remove onOutputFormatRequest from the VideoCapturer interface by magjed · 8 years ago
  26. 90ce01d The current default schedule delay of 30 ms prohibits by isheriff · 8 years ago
  27. 0fd22ef Rename P2PTransportChannel worker_thread_ to network_thread_. by johan · 8 years ago
  28. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  29. 51f2919 Update WebRTC to build against libsrtp 2.0 by mattdr · 8 years ago
  30. 24c7c12 Move FunctionView from AudioCodingModule to the rtc namespace by kwiberg · 8 years ago
  31. 35d43b9 Roll chromium_revision bdaa23ddfe..316b880c55 (421490:421519) by ehmaldonado · 8 years ago
  32. 7e146cb Fixing heap read overflow when "sctp-port" is in a video description. by deadbeef · 8 years ago
  33. 478681e Move the QP scaling thresholds to the relevant encoders. by kthelgason · 8 years ago
  34. e75f204 Expose Ivf logging through the native API by palmkvist · 8 years ago
  35. 242d8bd Unify the macOS and iOS capturer implementations by kthelgason · 8 years ago
  36. f5e3bbe Roll chromium_revision 386676ff4e..bdaa23ddfe (421470:421490) by buildbot · 8 years ago
  37. e5684c5 Delete method webrtc::VideoFrame::allocated_size and enum PlaneType. by nisse · 8 years ago
  38. 798896a Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo by danilchap · 8 years ago
  39. 9a8abcb Roll chromium_revision dd442d4812..386676ff4e (421425:421470) by ehmaldonado · 8 years ago
  40. 8fea199 [GN] Add missing framework headers by kthelgason · 8 years ago
  41. e0b2f15 Frame continuity is now tested as soon as a frame is inserted into the FrameBuffer. by philipel · 8 years ago
  42. 89a3a1a Moved Gn target rtc_event_log to one directory above. by charujain · 8 years ago
  43. b7446d7 GN: Fix incorrect include_dir for libjingle_peerconnection_jni target by charujain · 8 years ago
  44. f363d14 Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425) by Henrik Kjellander · 8 years ago
  45. 0c9e567 Landmine to clobber on Android and Windows. by kjellander · 8 years ago
  46. 5e3b5d1 CQ: Remove GYP Release trybots since we now only run GYP. by kjellander · 8 years ago
  47. e5e632f Hooking up target audio bitrate to audio network adaptor. by minyue · 8 years ago
  48. 72bebf1 Roll chromium_revision cede888c27..f86fb54ec3 (419407:420104) by buildbot · 8 years ago
  49. c3f549b Update expected Xcode version to 8.0. by kjellander · 8 years ago
  50. ee99696 Make 'webrtc' a static library. by kjellander · 8 years ago
  51. 822a16f Reland of Unify rtcp packet setters (patchset #1 id:1 of https://codereview.webrtc.org/2372713005/ ) by danilchap · 8 years ago
  52. 4151471 Add usage description strings to Info.plist by Kári Tristan Helgason · 8 years ago
  53. efc6e41 Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ ) by kjellander · 8 years ago
  54. 9532124 RTCPReceiver store cname as std::string. simplifying cname management. by danilchap · 8 years ago
  55. f1363fd Adds support for AVAudioSessionSilenceSecondaryAudioHintNotification on iOS by henrika · 8 years ago
  56. 46a8d18 ACM: Removed the code for InitialDelayManager by ossu · 8 years ago
  57. 29a44e3 This is a resubmission of https://codereview.webrtc.org/2047513002/ by kthelgason · 8 years ago
  58. 5f8ebae Add limitations of number of frames that can be created in I420BufferPool::CreateBuffer. by perkj · 8 years ago
  59. c8299f9 Posting Opus's set-force-channels functionality to WebRTC. by minyue · 8 years ago
  60. 20e77c7 Unify rtcp packet setters by danilchap · 8 years ago
  61. 4ecd970 GN: Fix incorrect include_dir for video_coding on iOS by kjellander · 8 years ago
  62. 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
  63. 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
  64. 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
  65. c637389 Delete unused file mock_audio_vector.h. by nisse · 8 years ago
  66. de2920c Delete unused file sessionid.h. by nisse · 8 years ago
  67. 89175a6 Trust that calls to RemoteEstimatorProxy::Process are done at the right frequency. by stefan · 8 years ago
  68. 25337bb Android: Update clang-format to follow Google style guide by magjed · 8 years ago
  69. fd8e33d Removing a useless ctor in AudioNetworkAdaptorImpl. by minyue · 8 years ago
  70. 8af4fd0 Disabled flaky VideoSendStreamTest.ChangingNetworkRoute by hbos · 8 years ago
  71. 98088dc header_usage.sh script: Exclude matches in gyp and gn files. by nisse · 8 years ago
  72. 660312b Enable //build/config/clang:extra_warnings for rtc_media by kjellander · 8 years ago
  73. 464382d Remove duplicated entry for bwe_simulations.cc by kjellander · 8 years ago
  74. 3901128 Remove unnecessary jsoncpp includes. by kjellander · 8 years ago
  75. 2068411 r14326 added '-Wno-unused-result' to 'WARNING_CFLAGS!' which removes the by jianjun.zhu · 8 years ago
  76. c59bf04 Remove differ from ScreenCapturer implementations by zijiehe · 8 years ago
  77. d3d230f - Make RtpSenderAudio not inherit from DtmfQueue. by solenberg · 8 years ago
  78. 92ea601 Move class RTCPHelp::RTCPPacketInformation into RTCPReceiver by danilchap · 8 years ago
  79. dda3666 Fixes minor issue in AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex for iOS. by henrika · 8 years ago
  80. f8c5f2b Fix vie_encoder_unittest.cc. by Per · 8 years ago
  81. 44428a8 iOS: Always build H264 HW encoder/decoder by magjed · 8 years ago
  82. 512ecb3 Let ViEEncoder tell VideoSendStream about reconfigurations. by Per · 8 years ago
  83. 2a27b0a Delete unused class FakeScreenCapturerFactory. by nisse · 8 years ago
  84. 347ec5c Change thread check to race check. Also, add comment to explain implementation of RaceChecker. by solenberg · 8 years ago
  85. f1b08da Stopped using the NetEqDecoder enum internally in NetEq. by ossu · 8 years ago
  86. 1490f7a Add histogram for end-to-end delay: "WebRTC.Video.EndToEndDelayInMs" by asapersson · 8 years ago
  87. 6d4c8c3 Renaming a proto target in GYP for audio network adaptor. by minyue · 8 years ago
  88. e87d673 Return texture frame when dropping frames in CameraCapturer. by sakal · 8 years ago
  89. b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
  90. 25f6a39 Relanding of "Adding debug dump to audio network adaptor." by minyue · 8 years ago
  91. 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
  92. 7e4f892 Adding debug dump to audio network adaptor. by minyue · 8 years ago
  93. a78213e Add tools/determinism to setup_links. by ehmaldonado · 8 years ago
  94. 893a7ee Support more QCOM specific color formats for Android HW decoder. by glaznev · 8 years ago
  95. 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
  96. 3e02430 Fix a stun attribute leak. by Honghai Zhang · 8 years ago
  97. d3af58b Rename AppRTCDemo on Android and iOS to AppRTCMobile by magjed · 8 years ago
  98. 051d151 Adds audio session status to logs for each valid audio route change on iOS by henrika · 8 years ago
  99. c37e983 Add custom info.plist to modules_unittests by kthelgason · 8 years ago
  100. f292e31 Relax too strict DCHECKs while parsing rtcp reports by danilchap · 8 years ago