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gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / AUTHORS
  1. 12e555b Delete wrapper API ConvertToI420 for YUV conversion to I420 by mallikarjun82 · 7 years ago
  2. 149533a Move rendering code in SurfaceViewRenderer to a separate class. by Xiaolei Yu · 7 years ago
  3. 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
  4. 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
  5. f1e08d0 Fix the video buffer size should take rtt into consideration by gustavogb · 7 years ago
  6. f3a48ab Delete unused field from AndroidVideoTrackSource by korniltsev.anatoly · 7 years ago
  7. ff7acb1 Reset isFirstFrameRendered on init of SurfaceViewRenderer by tserng · 7 years ago
  8. c43f68e Fix do not unregister bluetooth receiver if it was not registered by Gustavo Garcia · 7 years ago
  9. 1b2469b Fix AVFoundation framework import by hansknoechel92 · 7 years ago
  10. 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
  11. ace5c88 This CL adds RTCMTLVideoView.h and RTCCameraVideoCapturer.h to WebRTC.h by hewwatt · 7 years ago
  12. a1fa491 Fix invalid output buffer usage by steweg · 7 years ago
  13. 0d335c7 Fixed that RTCCameraPreviewView did not rotate the video on device rotation. by meetAkshay99 · 7 years ago
  14. 9d65f39 Added support for changing the volume of AudioTrack as discussed in BUG=webrtc:6533 by dax · 7 years ago
  15. 0642b32 Remove duplicate entries from AUTHORS file by henrik.lundin · 7 years ago
  16. 9f2c18e Changed OLA window for neteq. Old code didnt work well with 48khz by soren · 7 years ago
  17. 4b37127 Fix compilation issues of std::unique_ptr by steweg · 7 years ago
  18. 28dc285 Adding cbr support for Opus by soren · 7 years ago
  19. 0248e7c Re-add author accidentally removed in https://codereview.webrtc.org/2534843002. by solenberg · 7 years ago
  20. 846e1be Fix iOS8 crash in background mode. by sdkdimon · 7 years ago
  21. 228c268 Support 4 channel mic in Windows Core Audio by jens.nielsen · 7 years ago
  22. 0d1305e Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533 by frederik.riedel · 7 years ago
  23. 8a855d6 Allow any unsignalled SSRC changes on default video receive channel. by mzanaty · 7 years ago
  24. b11fb25 Protect APM in webkit builds. by agouaillard · 7 years ago
  25. 888874f Allow GCC 4.9 to compile Chromium by floppymaster · 7 years ago
  26. e5dc62a PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. by kjellander · 8 years ago
  27. ba7e71b remove googViewLimitedResolution stat by philipp.hancke · 8 years ago
  28. 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
  29. 610c454 Add Datachannel support to Android AppRTCMobile by hekra01 · 8 years ago
  30. bcc5d87 Add a GN target for unit tests, get them working again and added a test. by adam.fedor · 8 years ago
  31. 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
  32. 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
  33. a7a01df Add field_trial_default dependency to libjingle_peerconnection by arlolra · 8 years ago
  34. 96b6b83 iOS: add type to peer connection local streams by vopatop.skam · 8 years ago
  35. 3f70562 Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015). by conceptgenesis · 8 years ago
  36. bedc17b Fixing integer underflow in FileAudioDevice (webrtc issue 4554) by A.Brauckmann · 9 years ago
  37. 978244e Adding a bunch of Agora IO team members to the watch lists by yujie.mao · 9 years ago
  38. f70568c So long and thanks for all the code reviews! by andrew · 9 years ago
  39. bb79127 Add Riku Voipio to AUTHORS. by Andrew MacDonald · 9 years ago
  40. 88799d9 RTCEAGLVideoView: Fix GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT error. by christoffer · 9 years ago
  41. 92068ee Android: Guard against switching camera on stopped camera by colin · 9 years ago
  42. 4de6622 Fix a bug in computing audio delay on ios device. Converts seconds to by Jiawei Ou · 9 years ago
  43. fcfdb08 Update AUTHORS file. by tkchin · 9 years ago
  44. 4988ca5 Removed unused variables and the need to include the d3dx9.h file. by dkirovbroadsoft · 9 years ago
  45. 3ee4fe5 Re-land: Add API to get negotiated SSL ciphers by pthatcher@webrtc.org · 9 years ago
  46. 2bf0e90 Revert 8275 "This CL adds an API to the SSL stream adapters and ..." by tommi@webrtc.org · 9 years ago
  47. 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 · 9 years ago
  48. db1ebf6 Add jakehilton@gmail.com to AUTHORS by tnakamura@webrtc.org · 9 years ago
  49. 0ba1533 Added support for an Origin header in STUN messages. by pthatcher@webrtc.org · 10 years ago
  50. ee9d61c This fixes a small memory leak (found using Xcode/Instruments on iOS) in by tkchin@webrtc.org · 10 years ago
  51. c569a49 Unit tests for SSLAdapter by tkchin@webrtc.org · 10 years ago
  52. 31c285b Update AUTHORS file. by henrike@webrtc.org · 10 years ago
  53. ddb85ab Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 by jiayl@webrtc.org · 10 years ago
  54. d798095 replace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics. by andrew@webrtc.org · 10 years ago
  55. 0402515 Implement command line flags for peerconnection client example on Windows by kjellander@webrtc.org · 10 years ago
  56. 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
  57. ceffdbc Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() by henrike@webrtc.org · 10 years ago
  58. 82387e4 Add ability to receive calls for iOS BUG=2701 R=fischman@webrtc.org by fischman@webrtc.org · 10 years ago
  59. a9bdee6 Add Christophe Dumez to AUTHORS. by andrew@webrtc.org · 11 years ago
  60. 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
  61. eb7def2 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  62. 7b2f955 Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. by henrike@webrtc.org · 11 years ago
  63. efdf778 BUG=1351 by mflodman@webrtc.org · 11 years ago
  64. 17b867a compile fix for get_nprocs() with uClibc by phoglund@webrtc.org · 11 years ago
  65. 5140e24 MIPS optimizations for Signal Processing Library patch01 by andrew@webrtc.org · 11 years ago
  66. 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 · 11 years ago
  67. bcb7174 .gitignore: Add *.mk, created as part of ChromiumOS build by andrew@webrtc.org · 12 years ago
  68. 1b6da28 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. by pwestin@webrtc.org · 12 years ago
  69. ef62929 Landing http://review.webrtc.org/914006/ by niklas.enbom@webrtc.org · 12 years ago
  70. 01ad758 ilbc: Mark untouched input arrays as const by turaj@webrtc.org · 12 years ago
  71. e22beab [MIPS] Adding support for MIPS architecture for WebRTC. by andrew@webrtc.org · 12 years ago
  72. 87c50f0 Adding author by niklas.enbom@webrtc.org · 13 years ago
  73. 3a9680b Adding author by niklas.enbom@webrtc.org · 13 years ago
  74. da159d6 by niklase@google.com · 13 years ago