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gerrit-public.fairphone.software / platform / external / webrtc / a2644c06ee28c97ab670d59c416e59b8325fc678 / talk
  1. 6140fcc Move RTCFileLogger to webrtc/base/objc. by Jon Hjelle · 9 years ago
  2. 9bf5cde Update build_ios_libs.sh script to build new Objective-C API and gather header files. by hjon · 9 years ago
  3. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
  4. 5ad1297 Rename webrtc/media/webrtc -> webrtc/media/engine by kjellander@webrtc.org · 9 years ago
  5. 8fb3557 rtc::Buffer: Replace an internal rtc::scoped_ptr with std::unique_ptr by kwiberg · 9 years ago
  6. 162c339 Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #14 id:300001 of https://codereview.webrtc.org/1655793003/ ) by perkj · 9 years ago
  7. 4d19c5b This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it. by Per · 9 years ago
  8. 4b2a5a8 Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #12 id:260001 of https://codereview.webrtc.org/1655793003/ ) by perkj · 9 years ago
  9. 2f21789 This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it. by perkj · 9 years ago
  10. e2812e7 Cleanup after talk/media move. by kjellander@webrtc.org · 9 years ago
  11. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago
  12. dfb769d Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete by perkj · 9 years ago
  13. 47b6263 Remove Java PC support. This cl removes none Android Java support. by perkj · 9 years ago
  14. f6b5509 Fix GYP and GN references that are invalid in Chromium builds. by kjellander · 9 years ago
  15. fd6706a Log Android HW decoder delay time statistics. by glaznev · 9 years ago
  16. 8e8908a Delete FrameInput method and FrameInputWrapper class. by nisse · 9 years ago
  17. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  18. ae95ff3 Add more logging and fix PTS overflow for HW decoder. by glaznev · 9 years ago
  19. 20834ca Adds a nullptr check to prevent a rare crash when starting or stopping an RtcEventLog. by ivoc · 9 years ago
  20. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  21. 08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 9 years ago
  22. 8cb910d Delete backwards compatibility cruft from cricket::VideoFrame and VideoSourceInterface. by nisse · 9 years ago
  23. 9031d63 Remove the network with empty name or NONE connection type from the network list. by honghaiz · 9 years ago
  24. 14d024d Do not notify networkconnect if the connection type is known. by Honghai Zhang · 9 years ago
  25. 45b683f Call static method getConnectionType using the class name. by Honghai Zhang · 9 years ago
  26. cedff02 Remove dead code from WebRtcVideoEngine2. by Peter Boström · 9 years ago
  27. e03ac51 Implement NullVideoDecoder to avoid crash on unsupported decoders. by jbauch · 9 years ago
  28. 1088001 Support multiple rtx codecs. by Stefan Holmer · 9 years ago
  29. abe095b Roll chromium_revision c6076f2..609aa24 (372974:373145) by kjellander · 9 years ago
  30. 7f77749 Disable flaky test WebRtcSessionTest.TestRtxRemovedByCreateAnswer on Win and Mac. by honghaiz · 9 years ago
  31. 27a3485 Fixing a DCHECK failure on unknown connection type from OS. by honghaiz · 9 years ago
  32. a7ad7c3 Get the adapter type information from Android OS. by honghaiz · 9 years ago
  33. ed3277b Deprecate VideoDecoder::Reset() and remove calls. by Peter Boström · 9 years ago
  34. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  35. 9429148 Extra logging for HW codec. by glaznev · 9 years ago
  36. a6c39d9 Remove unimplemented VideoChannel code. by Peter Boström · 9 years ago
  37. eee86a6 Add option to disable particular HW video codec from app. by Alex Glaznev · 9 years ago
  38. b163c3f Delete unused members from VideoOptions by nisse · 9 years ago
  39. 378dc77 Consolidate setters into SetRecvParameters. by pbos · 9 years ago
  40. 46eed76 Removing "candidates" attribute from TransportDescription. by deadbeef · 9 years ago
  41. 6043f2e Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #5 id:80001 of https://codereview.webrtc.org/1581693006/ ) by terelius · 9 years ago
  42. e73afba New rtc::VideoSinkInterface. by nisse · 9 years ago
  43. bec70ab https://github.com/w3c/webrtc-stats/pull/10/files added mediaType to the tracks. The closest in the current stats is the ssrc type. by fippo · 9 years ago
  44. 6a062bd Deleted method AudioTrackInterface::GetRenderer. by nisse · 9 years ago
  45. ab8f82f Make ECDSA default for RTCPeerConnection by tkchin · 9 years ago
  46. d162a5e Add shouldDisableBuffering to RTCFileLogger. by tkchin · 9 years ago
  47. 919ff75 Use high QP threshold for HW VP8 encoder frame downscaling. by glaznev · 9 years ago
  48. 08a6eab Adding "first packet received" notification to PeerConnectionObserver. by Taylor Brandstetter · 9 years ago
  49. 7b3c72f Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #4 id:60001 of https://codereview.webrtc.org/1581693006/ ) by deadbeef · 9 years ago
  50. 42265a8 Adding "first packet received" notification to PeerConnectionObserver. by Taylor Brandstetter · 9 years ago
  51. 3afc8c4 Consolidate SetSendParameters into one setter. by Peter Boström · 9 years ago
  52. ec2922f Change PeerConnectionFactory.setVideoHwAccelerationOptions to create shared Egl context for harware encoders and decoders. by Per · 9 years ago
  53. 2098fca Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) by nisse · 9 years ago
  54. a862d45 New rtc::VideoSinkInterface. by Niels Möller · 9 years ago
  55. b11e97a Move talk/media/webrtc/OWNERS to talk/media. by Peter Boström · 9 years ago
  56. bab934b H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. by hbos · 9 years ago
  57. 3ea1852 Sync build_ios_libs.sh script with http://webrtc.org/native-code/ios/ by hjon · 9 years ago
  58. 4cb3e39 Fix compilation if HAVE_WEBRTC_VIDEO is not defined. by jbauch · 9 years ago
  59. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  60. 9de632a Deleted unused enums MediaChannelOptions and VoiceMediaChannelOptions, by nisse · 9 years ago
  61. 0a37497 Deleted unused method SetDumpPath and unneeded includes. by nisse · 9 years ago
  62. c8930ba Disable WebRtcSessionTest.TestStunError on Win. by minyue · 9 years ago
  63. 8947a01 Fixing an uninitialized variable in webrtcsession_unittest. by deadbeef · 9 years ago
  64. 3c16978 Remove cast to LocalAudioSource from AudioRtpSender. by Tommi · 9 years ago
  65. 0b98cf7 Delete CaptureRenderAdapter::VideoRenderInfo struct, it is unused since the recent deletion of SetSize. by nisse · 9 years ago
  66. 5082c83 Make type and constructors in EglBase14 public. by noahric · 9 years ago
  67. d26fadb Delete GetRenderer method, used only by the tests. by nisse · 9 years ago
  68. 057ecf0 Making WebRtcSession fire a destroyed signal. by deadbeef · 9 years ago
  69. 1d61a51 Send key frame if time difference between incoming frames exceeds a certain limit. by asapersson · 9 years ago
  70. 8a2c31d Make it possible to run peerconnection_unittests on Android. by perkj · 9 years ago
  71. c4c8485 Deleted renderer-related SetSize methods, and all uses. by nisse · 9 years ago
  72. 81354f5 Added mute logic to VideoTrackRenderers. by nisse · 9 years ago
  73. f5a3a93 Add 5-argument wrapper WebRtcVideoFrame::InitToBlack by Niels Möller · 9 years ago
  74. 8b1e431 Delete remnants of non-square pixel support from cricket::VideoFrame. by nisse · 9 years ago
  75. cec0a08 Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set. by honghaiz · 9 years ago
  76. f4decb5 Add QP statistics logging to Android HW encoder. by glaznev · 9 years ago
  77. 884f585 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  78. 79a5a83 Adapt to boringssl's new defaults. by torbjorng · 9 years ago
  79. d66b44d Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 9 years ago
  80. 0f7d293 Revert changes to default option setting in https://codereview.webrtc.org/1500633002/ by solenberg · 9 years ago
  81. dc305db Add ApplyPacketOptions() by Sergey Ulanov · 9 years ago
  82. 20ac434 Fix a test bot failure. by Honghai Zhang · 9 years ago
  83. e1f9d83 Adding AddTrack/RemoveTrack to native PeerConnection API. by deadbeef · 9 years ago
  84. 67b1e1a Put options as the argument of the java PeerConnectionFactory constructor. by honghaiz · 9 years ago
  85. fcfc804 Eliminate defines in talk/ by kjellander · 9 years ago
  86. 3542013 Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) by sprang · 9 years ago
  87. 31c8d2e Update with new default boringssl no-aes cipher suites. Re-enable tests. by Torbjorn Granlund · 9 years ago
  88. 688e308 Re-land: "Use an explicit identifier in Config" by aluebs · 9 years ago
  89. 268493a Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ ) by nisse · 9 years ago
  90. 709513d Delete remnants of non-square pixel support from cricket::VideoFrame. by nisse · 9 years ago
  91. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  92. fca54f4 Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) by tommi · 9 years ago
  93. 306efad Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan by kjellander · 9 years ago
  94. 25249d9 Use an explicit identifier in Config by aluebs · 9 years ago
  95. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  96. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  97. 3e1cfa7 Delete unused method webrtc::VideoRendererInterface::SetSize. by nisse · 9 years ago
  98. 127782b Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal. by nisse · 9 years ago
  99. b2328d1 Remove additional channel constraints when Beamforming is enabled in AudioProcessing by aluebs · 9 years ago
  100. a7446d2 Change DTLS default from 1.0 to 1.2 for webrtc. by Guo-wei Shieh · 9 years ago