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gerrit-public.fairphone.software / platform / external / webrtc / 15589a3be1a3106937a22283c466001796a7a8e2 / pc / channel_manager.h
  1. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  2. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/channelmanager.h]
  3. 4687915 Enable use of MediaTransportInterface for video streams. by Niels Möller · 6 years ago
  4. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  5. 98a462c Reland "Reland "Propagate media transport to media channel."" by Anton Sukhanov · 6 years ago
  6. 9accc9f Revert "Reland "Propagate media transport to media channel."" by Oleh Prypin · 6 years ago
  7. da65ed2 Reland "Propagate media transport to media channel." by Anton Sukhanov · 6 years ago
  8. 37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
  9. 8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
  10. a54daf1 Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Benjamin Wright · 6 years ago
  11. 8f4bc41 Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Oleh Prypin · 6 years ago
  12. ac2f3d1 Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h by Benjamin Wright · 6 years ago
  13. e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 7 years ago
  14. 2dfc42d Prepare to make BaseChannel depend on RtpTransportInternal only. by Zhi Huang · 7 years ago
  15. c9e1560 Modernize and cleanup ChannelManager by Steve Anton · 7 years ago
  16. d8970db Delete unneeded includes of fileutils.h by Niels Möller · 7 years ago
  17. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  18. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channelmanager.h]
  19. 774115c Change ChannelManager to use unique_ptr by Steve Anton · 7 years ago
  20. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  21. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  22. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  23. 2f45b6b Remove unused "crypto_options_" field. by jbauch · 7 years ago
  24. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  25. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
  26. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  27. 1a2183d Removing unnecessary parameters from CreateXChannel methods. by deadbeef · 8 years ago
  28. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  29. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  30. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  31. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  32. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  33. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  34. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  35. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  36. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  37. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  38. ebbe4f2 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. by zhihuang · 8 years ago
  39. 3cf8ece Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ ) by magjed · 8 years ago
  40. 9f71ec5 Stop caching supported codecs in WebRtcVideoEngine2 by magjed · 8 years ago
  41. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  42. 14d5dbe Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 8 years ago
  43. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 8 years ago
  44. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 8 years ago
  45. 05b9803 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 8 years ago
  46. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 8 years ago
  47. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 8 years ago
  48. c1513ee Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API. by ivoc · 8 years ago
  49. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  50. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  51. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 9 years ago
  52. fb45d17 Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/) by Per · 9 years ago
  53. 74622e0 Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ ) by perkj · 9 years ago
  54. e9c0cdf Removed unused cricket::VideoCapturer methods: by perkj · 9 years ago
  55. a509241 This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting. by Per · 9 years ago
  56. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  57. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/channelmanager.h]
  58. 162c339 Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #14 id:300001 of https://codereview.webrtc.org/1655793003/ ) by perkj · 9 years ago
  59. 4d19c5b This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it. by Per · 9 years ago
  60. 4b2a5a8 Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #12 id:260001 of https://codereview.webrtc.org/1655793003/ ) by perkj · 9 years ago
  61. 2f21789 This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it. by perkj · 9 years ago
  62. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  63. e73afba New rtc::VideoSinkInterface. by nisse · 9 years ago
  64. 2098fca Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) by nisse · 9 years ago
  65. a862d45 New rtc::VideoSinkInterface. by Niels Möller · 9 years ago
  66. d66b44d Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 9 years ago
  67. a4df27b Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ ) by ivoc · 9 years ago
  68. f4f5cb0 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 9 years ago
  69. 36d4c54 Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ ) by ivoc · 9 years ago
  70. ae2c5ad Added option to specify a maximum file size when recording an AEC dump. by ivoc · 9 years ago
  71. 822bdf9 Remove cricket::VideoEncoderConfig. by Peter Boström · 9 years ago
  72. 246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
  73. bd13838 Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack. by solenberg · 9 years ago
  74. 797ef12 Added StopAecDump function to PeerConnectionFactory. by ivoc · 9 years ago
  75. 112a3d8 Added functions on libjingle API to start and stop the recording of an RtcEventLog. by ivoc · 9 years ago
  76. d59daf8 Merging BaseSession code into WebRtcSession. by deadbeef · 9 years ago
  77. 4a3ccad Remove SetAudioDelayOffset() and friends. by solenberg · 9 years ago
  78. 61e933e Remove ChannelManager::GetCapabilities() by solenberg · 9 years ago
  79. facbbec Remove use of DeviceManager from ChannelManager. by solenberg · 9 years ago
  80. cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 9 years ago
  81. 7d17336 Remove the [Un]RegisterVoiceProcessor() API. by Fredrik Solenberg · 9 years ago
  82. a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 9 years ago
  83. 47ee2f3 TransportController refactoring. by deadbeef · 9 years ago
  84. c1a1b35 Remove the SetLocalMonitor() API. by solenberg · 9 years ago
  85. 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 9 years ago
  86. 9af63f4 TransportController refactoring. by deadbeef · 9 years ago
  87. 709ed67 Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels. by Fredrik Solenberg · 9 years ago
  88. c232096 Remove cricket::VideoProcessor and AddVideoProcessor() functionality by Magnus Jedvert · 9 years ago
  89. c28a896 VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation by Jelena Marusic · 9 years ago
  90. ccb49e7 Remove Soundclip handling from libjingle. by Fredrik Solenberg · 9 years ago
  91. 4b60c73 Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE. by Fredrik Solenberg · 10 years ago
  92. 81ea54e Remove WebRtcVideoEngine. by Peter Boström · 10 years ago
  93. 77f0e3f Remove GetStartCaptureFormat and some related code. by Peter Thatcher · 10 years ago
  94. 4aef5fe Add thread checks to the CaptureManager. by hbos@webrtc.org · 10 years ago
  95. 1e64263 Thread-safe ChannelManager.GetSupportedFormats, used by VideoSource by hbos@webrtc.org · 10 years ago
  96. 62f6e75 Refactoring WebRTC Java/JNI audio recording in C++ and Java. by henrika@webrtc.org · 10 years ago
  97. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  98. 28100cb Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." by henrike@webrtc.org · 10 years ago
  99. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  100. 1ecbe45 (Auto)update libjingle 77689511-> 77696841 by buildbot@webrtc.org · 10 years ago