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gerrit-public.fairphone.software / platform / external / webrtc / 8e13e6ed3dfbd2009f4c2e781dc19621d9dd8ca3 / AUTHORS
  1. 682dabd Add RTCStatisticsReport.h to WebRTC.framework. by CZ Theng · 5 years ago
  2. 36d171b Add Ramprakash Jelari to AUTHORS. by Sami Kalliomäki · 5 years ago
  3. fa77ba6 SetStreams API of RtpSender wrapped for iOS and Android by Cyril Lashkevich · 5 years ago
  4. 63173d5 pipewire: handle deleting the capturer while a D-Bus call is in progress by Michael Olbrich · 5 years ago
  5. ba5f8e9 Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5." by Guido Urdaneta · 5 years ago
  6. e715301 Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5. by Trevor Hayes · 5 years ago
  7. 1e00dbc Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly by Min Wang · 5 years ago
  8. 08f6a6c Import proto_library.gni when rtc_enable_protobuf is true by Kimmo Kinnunen · 6 years ago
  9. ce27875 [AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule by Alvaro Martinez · 6 years ago
  10. 8905d04 Add ',' between elements in RTCStatsReport::ToJson by Dirk-Jan C. Binnema · 6 years ago
  11. 74cdf78 add cstring include need for strncmp by Michel Promonet · 6 years ago
  12. 83aa5ac Adding Microsoft Corporation (*@microsoft.com) to WebRTC AUTHORS by James Cadd · 6 years ago
  13. 318da51 Reland "Add support for screen sharing with PipeWire on Wayland" by Tomas Popela · 6 years ago
  14. 3ddaf3c Revert "Add support for screen sharing with PipeWire on Wayland" by Patrik Höglund · 6 years ago
  15. bf0d0c1 Add IPv6 configuration parameters to iOS API by Uladzislau Susha · 6 years ago
  16. dd20c9c Add support for screen sharing with PipeWire on Wayland by Tomas Popela · 6 years ago
  17. 0eb7d3ff Always call ConvertToI420 with positive crop_height by Robert Bares · 6 years ago
  18. 6fcf6ca Modified PressEnterToContinue() to actualy check if Enter is pressed by Danail Kirov · 6 years ago
  19. d2fb1bf Generate module.modulemap file when building Mac Framework by Joel Sutherland · 6 years ago
  20. 289e980 Remove unused var in device info bits from video capture module for Linux by Jose Antonio Olivera Ortega · 6 years ago
  21. e0c8b23 Frame marking RTP header extension (PART 1: implement extension) by Johnny Lee · 6 years ago
  22. 8cec4fb Use default RTCConfiguration on iOS by Yuriy Pavlyshak · 6 years ago
  23. ccee56b Add certificate generate/set functionality to bring iOS closer to JS API by Michael Iedema · 6 years ago
  24. 25cc8ad Fixed issue with BGRA RTCCVPixelBuffer scale and crop by David Porter · 6 years ago
  25. e250645 Call callback in IDLE state by Ing. Jan Kaláb · 6 years ago
  26. 43800f9 Generalize SimulcastEncoderAdapter, use for H264 & VP8. by Sergio Garcia Murillo · 6 years ago
  27. 6f440ed Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." by Mirko Bonadei · 6 years ago
  28. 07efe43 Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8. by Sergio Garcia Murillo · 6 years ago
  29. b90e63c6 Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet by Peng Yu · 6 years ago
  30. e7e0602 ObjC: Notify local video track by Piasy Xu · 6 years ago
  31. f8d8d6d Use range-based-for instead of std::for_each and std::mem_fun by Yusuke Suzuki · 6 years ago
  32. ebd9abc Use IFA_LOCAL instead of IFA_ADDRESS over IPv4 network on ANDROID by Yongje Lee · 6 years ago
  33. a72b7fc ObjC: Add missing _lastDrawnFrame assignments by Maxim Pavlov · 7 years ago
  34. 3cfe9e1 Fixed video capturing on Mac. by Maksim Khobat · 7 years ago
  35. 2870b0a Expose a link-local network interfaces enumeration option by Daniel Lazarenko · 7 years ago
  36. 5e4833c Add missing stdio.h header in files using scanf/sscanf function. by Piotr Tworek · 7 years ago
  37. b887435 RemoteBitrateEstimatorAbsSendTime: check clock is a valid ref by Miguel Paris · 7 years ago
  38. d3c642b Fix typo in the include path of ooura_fft.h by Jiawei Ou · 7 years ago
  39. 6728003 Skip H246 scaling lists in SPS packets by Todd Wong · 7 years ago
  40. 5a7508a Fixed NPE inside org.webrtc.Camera1Session.create by Yura Yaroshevich · 7 years ago
  41. 1c62ffa Normalize main(..) routines for WinUWP by Robin Raymond · 7 years ago
  42. 12e555b Delete wrapper API ConvertToI420 for YUV conversion to I420 by mallikarjun82 · 7 years ago
  43. 149533a Move rendering code in SurfaceViewRenderer to a separate class. by Xiaolei Yu · 7 years ago
  44. e21be1d Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) by philipel · 7 years ago
  45. bdbc889 Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ ) by philipel · 7 years ago
  46. f1e08d0 Fix the video buffer size should take rtt into consideration by gustavogb · 7 years ago
  47. f3a48ab Delete unused field from AndroidVideoTrackSource by korniltsev.anatoly · 7 years ago
  48. ff7acb1 Reset isFirstFrameRendered on init of SurfaceViewRenderer by tserng · 7 years ago
  49. c43f68e Fix do not unregister bluetooth receiver if it was not registered by Gustavo Garcia · 7 years ago
  50. 1b2469b Fix AVFoundation framework import by hansknoechel92 · 7 years ago
  51. 8e857d1 Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device. by Tarun Chawla · 7 years ago
  52. ace5c88 This CL adds RTCMTLVideoView.h and RTCCameraVideoCapturer.h to WebRTC.h by hewwatt · 7 years ago
  53. a1fa491 Fix invalid output buffer usage by steweg · 7 years ago
  54. 0d335c7 Fixed that RTCCameraPreviewView did not rotate the video on device rotation. by meetAkshay99 · 8 years ago
  55. 9d65f39 Added support for changing the volume of AudioTrack as discussed in BUG=webrtc:6533 by dax · 8 years ago
  56. 0642b32 Remove duplicate entries from AUTHORS file by henrik.lundin · 8 years ago
  57. 9f2c18e Changed OLA window for neteq. Old code didnt work well with 48khz by soren · 8 years ago
  58. 4b37127 Fix compilation issues of std::unique_ptr by steweg · 8 years ago
  59. 28dc285 Adding cbr support for Opus by soren · 8 years ago
  60. 0248e7c Re-add author accidentally removed in https://codereview.webrtc.org/2534843002. by solenberg · 8 years ago
  61. 846e1be Fix iOS8 crash in background mode. by sdkdimon · 8 years ago
  62. 228c268 Support 4 channel mic in Windows Core Audio by jens.nielsen · 8 years ago
  63. 0d1305e Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533 by frederik.riedel · 8 years ago
  64. 8a855d6 Allow any unsignalled SSRC changes on default video receive channel. by mzanaty · 8 years ago
  65. b11fb25 Protect APM in webkit builds. by agouaillard · 8 years ago
  66. 888874f Allow GCC 4.9 to compile Chromium by floppymaster · 8 years ago
  67. e5dc62a PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. by kjellander · 8 years ago
  68. ba7e71b remove googViewLimitedResolution stat by philipp.hancke · 8 years ago
  69. bbfed52 Set OPENSSL_EC_NAMED_CURVE explicitly on EC key so that certificate has ASN1 OID and NIST curve info. Without this openSSL handshake negotiation fails throwing NO_SHARED_CIPHER error. the change made is along the lines of openssl behavior documented here: https://wiki.openssl.org/index.php/Elliptic_Curve_Diffie_Hellman#ECDH_and_Named_Curves by ssaroha · 8 years ago
  70. 610c454 Add Datachannel support to Android AppRTCMobile by hekra01 · 8 years ago
  71. bcc5d87 Add a GN target for unit tests, get them working again and added a test. by adam.fedor · 8 years ago
  72. a264ecc Copy RTCAudioSource.h and RTCMediaSource.h with other public header files when building the WebRTC framework for iOS / Mac by VladimirTechMan · 8 years ago
  73. 86ccd7b Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ ) by sakal · 8 years ago
  74. a7a01df Add field_trial_default dependency to libjingle_peerconnection by arlolra · 8 years ago
  75. 96b6b83 iOS: add type to peer connection local streams by vopatop.skam · 8 years ago
  76. 3f70562 Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015). by conceptgenesis · 9 years ago
  77. bedc17b Fixing integer underflow in FileAudioDevice (webrtc issue 4554) by A.Brauckmann · 9 years ago
  78. 978244e Adding a bunch of Agora IO team members to the watch lists by yujie.mao · 9 years ago
  79. f70568c So long and thanks for all the code reviews! by andrew · 9 years ago
  80. bb79127 Add Riku Voipio to AUTHORS. by Andrew MacDonald · 9 years ago
  81. 88799d9 RTCEAGLVideoView: Fix GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT error. by christoffer · 9 years ago
  82. 92068ee Android: Guard against switching camera on stopped camera by colin · 9 years ago
  83. 4de6622 Fix a bug in computing audio delay on ios device. Converts seconds to by Jiawei Ou · 9 years ago
  84. fcfdb08 Update AUTHORS file. by tkchin · 9 years ago
  85. 4988ca5 Removed unused variables and the need to include the d3dx9.h file. by dkirovbroadsoft · 9 years ago
  86. 3ee4fe5 Re-land: Add API to get negotiated SSL ciphers by pthatcher@webrtc.org · 10 years ago
  87. 2bf0e90 Revert 8275 "This CL adds an API to the SSL stream adapters and ..." by tommi@webrtc.org · 10 years ago
  88. 1d11c82 This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer. by pthatcher@webrtc.org · 10 years ago
  89. db1ebf6 Add jakehilton@gmail.com to AUTHORS by tnakamura@webrtc.org · 10 years ago
  90. 0ba1533 Added support for an Origin header in STUN messages. by pthatcher@webrtc.org · 10 years ago
  91. ee9d61c This fixes a small memory leak (found using Xcode/Instruments on iOS) in by tkchin@webrtc.org · 10 years ago
  92. c569a49 Unit tests for SSLAdapter by tkchin@webrtc.org · 10 years ago
  93. 31c285b Update AUTHORS file. by henrike@webrtc.org · 10 years ago
  94. ddb85ab Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 by jiayl@webrtc.org · 10 years ago
  95. d798095 replace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics. by andrew@webrtc.org · 10 years ago
  96. 0402515 Implement command line flags for peerconnection client example on Windows by kjellander@webrtc.org · 10 years ago
  97. 7c82ada AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial. by fischman@webrtc.org · 10 years ago
  98. ceffdbc Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() by henrike@webrtc.org · 10 years ago
  99. 82387e4 Add ability to receive calls for iOS BUG=2701 R=fischman@webrtc.org by fischman@webrtc.org · 11 years ago
  100. a9bdee6 Add Christophe Dumez to AUTHORS. by andrew@webrtc.org · 11 years ago