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gerrit-public.fairphone.software / platform / external / webrtc / cc91d284e43e266f97edb999eb2ebfc8a094beac
  1. cc91d28 Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  2. 6ab97ce RTCCertificateStats[1] added. by hbos · 8 years ago
  3. 607d9d7 rtc::FunctionView improvements: accept function pointers and nullptr by kwiberg · 8 years ago
  4. 07a224b webrtc/api/stats/OWNERS file added. by hbos · 8 years ago
  5. d227522 Change DCHECK in VCMDecodedFrameCallback back to just logging. by sakal · 8 years ago
  6. 8f90106 Revert of Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere (patchset #2 id:20001 of https://codereview.webrtc.org/2384693002/ ) by guidou · 8 years ago
  7. 65b42c2 Fix receiving H264 video from iPhone on Kitkat devices. by sakal · 8 years ago
  8. 8f741e9 Change Camera1Enumerator to create a Camera1Capturer instead of VideoCapturerAndroid. by sakal · 8 years ago
  9. 0aabdac Generalize UlpfecPacketGenerator into AugmentedPacketGenerator. by brandtr · 8 years ago
  10. b19d288 Remove chain of methods in RtpRtcp module to get current payload frequency for RTCP SRs, which anyway was stuck to defaults for video/audio. by solenberg · 8 years ago
  11. 71ca747 Style fixes in FecReceiver and ProducerFec unit tests. by brandtr · 8 years ago
  12. ab0b929 Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere by kwiberg · 8 years ago
  13. 1d477ea Default constructor for file by Viktor Palmkvist · 8 years ago
  14. 470c088 Fix modules_unittests on iOS. by Kári Tristan Helgason · 8 years ago
  15. a4545ee Rename FrameGenerator -> UlpfecPacketGenerator. by brandtr · 8 years ago
  16. cada34c Fuzzer for FEC header readers. by brandtr · 8 years ago
  17. 0496de2 Add FlexFEC header formatters. by brandtr · 8 years ago
  18. 8ff860a Add support for WeakPtr<T> The implementation is borrowed from Chromium. by perkj · 8 years ago
  19. fa10b55 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  20. a1d9ad0 Creating controller manager from config string in audio network adaptor. by minyue · 8 years ago
  21. 7411061 Use RtpPacketToSend in RtpSenderVideo. by danilchap · 8 years ago
  22. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  23. fe69a74 Making ContinueSSL synchronously set the state to "open". by Taylor Brandstetter · 8 years ago
  24. dd7fb43 Emit SignalReadyToSend even for "presumed writable" connections. by deadbeef · 8 years ago
  25. 89824f6 Relanding: Allow the DTLS fingerprint verification to occur after the handshake. by deadbeef · 8 years ago
  26. 3cdfcd8 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
  27. 61050f6 Fixig issues in BWE dynamics plot scripts. by gaetano.carlucci · 8 years ago
  28. 17b0263 Use sps and pps to determine decodability of H.264 frames. by Stefan Holmer · 8 years ago
  29. 55d932b Add logging statements to places where the frame might be dropped in WebRTC pipeline. by sakal · 8 years ago
  30. 115bd15 New helper function test::ReadI420Buffer, refactor FrameReader to use it. by nisse · 8 years ago
  31. 6f112cc Delete unused support for vp8 partitions. by nisse · 8 years ago
  32. 3cc47eb Add sanity check for decreasing RTP timestamp in RtpToNtpMs. by asapersson · 8 years ago
  33. f5297a0 Reland of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #1 id:1 of https://codereview.webrtc.org/2357113002/ ) by nisse · 8 years ago
  34. 280de9e Reland: Fix race / crash in OnNetworkRouteChanged(). by Stefan Holmer · 8 years ago
  35. 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
  36. edbae5e Remove Crit::Scope lock by using atomic bool property. by denicija · 8 years ago
  37. eb5040a Disable TCPChannelClientTest.testConnectIPv6 by ehmaldonado · 8 years ago
  38. 15e4ec3 Remove compat for iOS 7/8 by Kári Tristan Helgason · 8 years ago
  39. 3b703ed Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  40. b73d269 Replace RelayPort with TurnPort in p2ptransportchannel tests. by Honghai Zhang · 8 years ago
  41. 26105b4 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  42. 8b8459e suppress memcheck test errors in the Opus decoder and encoder. by honghaiz · 8 years ago
  43. 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
  44. 7851bda Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver by danilchap · 8 years ago
  45. c8d2171 Replace RelayPort with TurnPort in p2ptransportchannel tests. by Honghai Zhang · 8 years ago
  46. 7502401 Do not spam "Connect failed with 101/65" in logs. by skvlad · 8 years ago
  47. 591c709 Suppress a memcheck error in Opus decoder by henrik.lundin · 8 years ago
  48. 590cf28 Add autothread to pseudo-tcp fuzzer. by phoglund · 8 years ago
  49. 70736e4 Remove old presumably unused directory. by sakal · 8 years ago
  50. 8e6a761 ProbeController: Limit max probing bitrate by isheriff · 8 years ago
  51. 6060186 Add presubmit format requirement for webrtc/api/android by magjed · 8 years ago
  52. 5614566 Fix faulty include paths that break the build by Henrik Lundin · 8 years ago
  53. 5ec85fb Revert of Fix race / crash in OnNetworkRouteChanged(). (patchset #5 id:80001 of https://codereview.webrtc.org/2366333003/ ) by stefan · 8 years ago
  54. b6760f9 Format all Java in WebRTC. by sakal · 8 years ago
  55. a48ddb7 Add VideoSendStream::Stats::prefered_media_bitrate_bps by Per · 8 years ago
  56. fd0d426 Fix race / crash in OnNetworkRouteChanged(). by stefan · 8 years ago
  57. 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
  58. ff9793c Android: Remove onOutputFormatRequest from the VideoCapturer interface by magjed · 8 years ago
  59. 90ce01d The current default schedule delay of 30 ms prohibits by isheriff · 8 years ago
  60. 0fd22ef Rename P2PTransportChannel worker_thread_ to network_thread_. by johan · 8 years ago
  61. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  62. 51f2919 Update WebRTC to build against libsrtp 2.0 by mattdr · 8 years ago
  63. 24c7c12 Move FunctionView from AudioCodingModule to the rtc namespace by kwiberg · 8 years ago
  64. 35d43b9 Roll chromium_revision bdaa23ddfe..316b880c55 (421490:421519) by ehmaldonado · 8 years ago
  65. 7e146cb Fixing heap read overflow when "sctp-port" is in a video description. by deadbeef · 8 years ago
  66. 478681e Move the QP scaling thresholds to the relevant encoders. by kthelgason · 8 years ago
  67. e75f204 Expose Ivf logging through the native API by palmkvist · 8 years ago
  68. 242d8bd Unify the macOS and iOS capturer implementations by kthelgason · 8 years ago
  69. f5e3bbe Roll chromium_revision 386676ff4e..bdaa23ddfe (421470:421490) by buildbot · 8 years ago
  70. e5684c5 Delete method webrtc::VideoFrame::allocated_size and enum PlaneType. by nisse · 8 years ago
  71. 798896a Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo by danilchap · 8 years ago
  72. 9a8abcb Roll chromium_revision dd442d4812..386676ff4e (421425:421470) by ehmaldonado · 8 years ago
  73. 8fea199 [GN] Add missing framework headers by kthelgason · 8 years ago
  74. e0b2f15 Frame continuity is now tested as soon as a frame is inserted into the FrameBuffer. by philipel · 8 years ago
  75. 89a3a1a Moved Gn target rtc_event_log to one directory above. by charujain · 8 years ago
  76. b7446d7 GN: Fix incorrect include_dir for libjingle_peerconnection_jni target by charujain · 8 years ago
  77. f363d14 Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425) by Henrik Kjellander · 8 years ago
  78. 0c9e567 Landmine to clobber on Android and Windows. by kjellander · 8 years ago
  79. 5e3b5d1 CQ: Remove GYP Release trybots since we now only run GYP. by kjellander · 8 years ago
  80. e5e632f Hooking up target audio bitrate to audio network adaptor. by minyue · 8 years ago
  81. 72bebf1 Roll chromium_revision cede888c27..f86fb54ec3 (419407:420104) by buildbot · 8 years ago
  82. c3f549b Update expected Xcode version to 8.0. by kjellander · 8 years ago
  83. ee99696 Make 'webrtc' a static library. by kjellander · 8 years ago
  84. 822a16f Reland of Unify rtcp packet setters (patchset #1 id:1 of https://codereview.webrtc.org/2372713005/ ) by danilchap · 8 years ago
  85. 4151471 Add usage description strings to Info.plist by Kári Tristan Helgason · 8 years ago
  86. efc6e41 Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ ) by kjellander · 8 years ago
  87. 9532124 RTCPReceiver store cname as std::string. simplifying cname management. by danilchap · 8 years ago
  88. f1363fd Adds support for AVAudioSessionSilenceSecondaryAudioHintNotification on iOS by henrika · 8 years ago
  89. 46a8d18 ACM: Removed the code for InitialDelayManager by ossu · 8 years ago
  90. 29a44e3 This is a resubmission of https://codereview.webrtc.org/2047513002/ by kthelgason · 8 years ago
  91. 5f8ebae Add limitations of number of frames that can be created in I420BufferPool::CreateBuffer. by perkj · 8 years ago
  92. c8299f9 Posting Opus's set-force-channels functionality to WebRTC. by minyue · 8 years ago
  93. 20e77c7 Unify rtcp packet setters by danilchap · 8 years ago
  94. 4ecd970 GN: Fix incorrect include_dir for video_coding on iOS by kjellander · 8 years ago
  95. 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
  96. 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
  97. 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
  98. c637389 Delete unused file mock_audio_vector.h. by nisse · 8 years ago
  99. de2920c Delete unused file sessionid.h. by nisse · 8 years ago
  100. 89175a6 Trust that calls to RemoteEstimatorProxy::Process are done at the right frequency. by stefan · 8 years ago