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gerrit-public.fairphone.software / platform / external / webrtc / 5b4f075f9c599d49785430e8619f47b7cff1c102 / AUTHORS
  1. 2870b0a Expose a link-local network interfaces enumeration option by Daniel Lazarenko · 7 years ago
  2. 5e4833c Add missing stdio.h header in files using scanf/sscanf function. by Piotr Tworek · 7 years ago
  3. b887435 RemoteBitrateEstimatorAbsSendTime: check clock is a valid ref by Miguel Paris · 7 years ago
  4. d3c642b Fix typo in the include path of ooura_fft.h by Jiawei Ou · 7 years ago
  5. 6728003 Skip H246 scaling lists in SPS packets by Todd Wong · 7 years ago
  6. 5a7508a Fixed NPE inside org.webrtc.Camera1Session.create by Yura Yaroshevich · 7 years ago
  7. 1c62ffa Normalize main(..) routines for WinUWP by Robin Raymond · 7 years ago
  8. 12e555b Delete wrapper API ConvertToI420 for YUV conversion to I420 by mallikarjun82 · 7 years ago
  9. 149533a Move rendering code in SurfaceViewRenderer to a separate class. by Xiaolei Yu · 7 years ago
  10. 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
  11. 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
  12. f1e08d0 Fix the video buffer size should take rtt into consideration by gustavogb · 7 years ago
  13. f3a48ab Delete unused field from AndroidVideoTrackSource by korniltsev.anatoly · 7 years ago
  14. ff7acb1 Reset isFirstFrameRendered on init of SurfaceViewRenderer by tserng · 7 years ago
  15. c43f68e Fix do not unregister bluetooth receiver if it was not registered by Gustavo Garcia · 7 years ago
  16. 1b2469b Fix AVFoundation framework import by hansknoechel92 · 7 years ago
  17. 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
  18. ace5c88 This CL adds RTCMTLVideoView.h and RTCCameraVideoCapturer.h to WebRTC.h by hewwatt · 7 years ago
  19. a1fa491 Fix invalid output buffer usage by steweg · 7 years ago
  20. 0d335c7 Fixed that RTCCameraPreviewView did not rotate the video on device rotation. by meetAkshay99 · 7 years ago
  21. 9d65f39 Added support for changing the volume of AudioTrack as discussed in BUG=webrtc:6533 by dax · 7 years ago
  22. 0642b32 Remove duplicate entries from AUTHORS file by henrik.lundin · 7 years ago
  23. 9f2c18e Changed OLA window for neteq. Old code didnt work well with 48khz by soren · 7 years ago
  24. 4b37127 Fix compilation issues of std::unique_ptr by steweg · 7 years ago
  25. 28dc285 Adding cbr support for Opus by soren · 7 years ago
  26. 0248e7c Re-add author accidentally removed in https://codereview.webrtc.org/2534843002. by solenberg · 7 years ago
  27. 846e1be Fix iOS8 crash in background mode. by sdkdimon · 7 years ago
  28. 228c268 Support 4 channel mic in Windows Core Audio by jens.nielsen · 7 years ago
  29. 0d1305e Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533 by frederik.riedel · 7 years ago
  30. 8a855d6 Allow any unsignalled SSRC changes on default video receive channel. by mzanaty · 8 years ago
  31. b11fb25 Protect APM in webkit builds. by agouaillard · 8 years ago
  32. 888874f Allow GCC 4.9 to compile Chromium by floppymaster · 8 years ago
  33. e5dc62a PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. by kjellander · 8 years ago
  34. ba7e71b remove googViewLimitedResolution stat by philipp.hancke · 8 years ago
  35. 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
  36. 610c454 Add Datachannel support to Android AppRTCMobile by hekra01 · 8 years ago
  37. bcc5d87 Add a GN target for unit tests, get them working again and added a test. by adam.fedor · 8 years ago
  38. 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
  39. 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
  40. a7a01df Add field_trial_default dependency to libjingle_peerconnection by arlolra · 8 years ago
  41. 96b6b83 iOS: add type to peer connection local streams by vopatop.skam · 8 years ago
  42. 3f70562 Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015). by conceptgenesis · 9 years ago
  43. bedc17b Fixing integer underflow in FileAudioDevice (webrtc issue 4554) by A.Brauckmann · 9 years ago
  44. 978244e Adding a bunch of Agora IO team members to the watch lists by yujie.mao · 9 years ago
  45. f70568c So long and thanks for all the code reviews! by andrew · 9 years ago
  46. bb79127 Add Riku Voipio to AUTHORS. by Andrew MacDonald · 9 years ago
  47. 88799d9 RTCEAGLVideoView: Fix GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT error. by christoffer · 9 years ago
  48. 92068ee Android: Guard against switching camera on stopped camera by colin · 9 years ago
  49. 4de6622 Fix a bug in computing audio delay on ios device. Converts seconds to by Jiawei Ou · 9 years ago
  50. fcfdb08 Update AUTHORS file. by tkchin · 9 years ago
  51. 4988ca5 Removed unused variables and the need to include the d3dx9.h file. by dkirovbroadsoft · 9 years ago
  52. 3ee4fe5 Re-land: Add API to get negotiated SSL ciphers by pthatcher@webrtc.org · 10 years ago
  53. 2bf0e90 Revert 8275 "This CL adds an API to the SSL stream adapters and ..." by tommi@webrtc.org · 10 years ago
  54. 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
  55. db1ebf6 Add jakehilton@gmail.com to AUTHORS by tnakamura@webrtc.org · 10 years ago
  56. 0ba1533 Added support for an Origin header in STUN messages. by pthatcher@webrtc.org · 10 years ago
  57. ee9d61c This fixes a small memory leak (found using Xcode/Instruments on iOS) in by tkchin@webrtc.org · 10 years ago
  58. c569a49 Unit tests for SSLAdapter by tkchin@webrtc.org · 10 years ago
  59. 31c285b Update AUTHORS file. by henrike@webrtc.org · 10 years ago
  60. ddb85ab Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 by jiayl@webrtc.org · 10 years ago
  61. d798095 replace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics. by andrew@webrtc.org · 10 years ago
  62. 0402515 Implement command line flags for peerconnection client example on Windows by kjellander@webrtc.org · 10 years ago
  63. 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
  64. ceffdbc Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() by henrike@webrtc.org · 10 years ago
  65. 82387e4 Add ability to receive calls for iOS BUG=2701 R=fischman@webrtc.org by fischman@webrtc.org · 11 years ago
  66. a9bdee6 Add Christophe Dumez to AUTHORS. by andrew@webrtc.org · 11 years ago
  67. af320fd The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation. by fischman@webrtc.org · 11 years ago
  68. eb7def2 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  69. 7b2f955 Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. by henrike@webrtc.org · 11 years ago
  70. efdf778 BUG=1351 by mflodman@webrtc.org · 11 years ago
  71. 17b867a compile fix for get_nprocs() with uClibc by phoglund@webrtc.org · 11 years ago
  72. 5140e24 MIPS optimizations for Signal Processing Library patch01 by andrew@webrtc.org · 12 years ago
  73. 73a702c This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware. by andrew@webrtc.org · 12 years ago
  74. bcb7174 .gitignore: Add *.mk, created as part of ChromiumOS build by andrew@webrtc.org · 12 years ago
  75. 1b6da28 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. by pwestin@webrtc.org · 12 years ago
  76. ef62929 Landing http://review.webrtc.org/914006/ by niklas.enbom@webrtc.org · 12 years ago
  77. 01ad758 ilbc: Mark untouched input arrays as const by turaj@webrtc.org · 12 years ago
  78. e22beab [MIPS] Adding support for MIPS architecture for WebRTC. by andrew@webrtc.org · 12 years ago
  79. 87c50f0 Adding author by niklas.enbom@webrtc.org · 13 years ago
  80. 3a9680b Adding author by niklas.enbom@webrtc.org · 13 years ago
  81. da159d6 by niklase@google.com · 13 years ago