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gerrit-public.fairphone.software / platform / external / webrtc / 9bac68c0cc4444b852416396f0e0f31ea66a9cfe / media / base / fake_media_engine.h
  1. 9bac68c Reland "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 4 years, 6 months ago
  2. 00a3087 Revert "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 4 years, 6 months ago
  3. 133bf2b Reland "Distinguish between send and receive codecs" by Johannes Kron · 4 years, 6 months ago
  4. e57b266 Revert "Distinguish between send and receive codecs" by Steve Anton · 4 years, 6 months ago
  5. c0f25cf Distinguish between send and receive codecs by Johannes Kron · 4 years, 6 months ago
  6. f5ecb5f Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs"""" by Mirko Bonadei · 4 years, 6 months ago
  7. 9cad4dc Reland "Reland "Reland "Distinguish between send and receive video codecs""" by Johannes Kron · 4 years, 7 months ago
  8. b5159fe Revert "Reland "Reland "Distinguish between send and receive video codecs""" by Olga Sharonova · 4 years, 7 months ago
  9. 4e64e60 Reland "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 4 years, 7 months ago
  10. f9d92ed Revert "Reland "Distinguish between send and receive video codecs"" by Ilya Nikolaevskiy · 4 years, 7 months ago
  11. 77eb338 Reland "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 7 months ago
  12. f2d6fe6 Revert "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 4 years, 7 months ago
  13. 26e6afe Reland "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 7 months ago
  14. f22af3c Revert "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 7 months ago
  15. 18314bd Distinguish between send and receive video codecs by Johannes Kron · 4 years, 7 months ago
  16. 749f660 Enable SSRC 0 in MediaChannel methods by Saurav Das · 4 years, 8 months ago
  17. 32565f6 WebRtcVideoEngine: Enable encoded frame sink. by Markus Handell · 4 years, 8 months ago
  18. 934afc6 Deprecate RtpReceiver's SetParameters method by Saurav Das · 4 years, 8 months ago
  19. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 4 years, 10 months ago
  20. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 4 years, 10 months ago
  21. 6e9c2fd Delete StartRtcEventLog and StopRtcEventLog methods from FakeVoiceEngine by Niels Möller · 5 years ago
  22. e8e4dc4 Change StartAecDump methods to work with FILE* and FileWrapper by Niels Möller · 5 years ago
  23. 2d9d82e Implement RTCRtpTransceiver.setCodecPreferences by Florent Castelli · 5 years ago
  24. a3aa9bd Make VideoBitrateAllocatorFactory injectable. by Jonas Oreland · 5 years ago
  25. e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 5 years ago
  26. 493a650 Propagate base minimum delay from video jitter buffer to webrtc/api. by Ruslan Burakov · 5 years ago
  27. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 5 years ago
  28. c1a0bcb Implement the encoding RtpParameter scaleResolutionDownBy by Florent Castelli · 5 years ago
  29. 2c9ebef Use Abseil container algorithms in media/ by Steve Anton · 5 years ago
  30. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  31. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from media/base/fakemediaengine.h]
  32. 38332cd Add RTCP and simulcast support for RTCRtpReceiver::getParameters() by Florent Castelli · 6 years ago
  33. fa0aa39 Removes templating from CompositeMediaEngine. by Sebastian Jansson · 6 years ago
  34. 84848f2 Adds interfaces for audio and video engines. by Sebastian Jansson · 6 years ago
  35. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  36. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  37. 7e6b528 Removes FakeBaseEngine. by Sebastian Jansson · 6 years ago
  38. cb06cac Moves fake media engine implementation to cc file. by Sebastian Jansson · 6 years ago
  39. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  40. 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
  41. 49ac595 Add GetSources to VideoRtpReceiver by Jonas Oreland · 6 years ago
  42. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  43. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  44. 2d2c888 Returns RTCError for setting unimplemented RtpParameters. by Seth Hampson · 6 years ago
  45. ff40b14 Delete obsolete enable argument to SetVideoSend. by Niels Möller · 6 years ago
  46. f120cba Delete AudioMonitor and related code. by Niels Möller · 6 years ago
  47. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 6 years ago
  48. 4ab68ee Move sessiondescription.h/cc from p2p/base to pc/ by Steve Anton · 7 years ago
  49. aba85d1 Resolve circular dependency in rtc_media_base. by Patrik Höglund · 7 years ago
  50. 5f5918f Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks. by Zhi Huang · 7 years ago
  51. e78bcb9 Enable cpplint in media/ by Steve Anton · 7 years ago
  52. 8d3444d Reland "Rewrite WebRtcSession media tests as PeerConnection tests" by Steve Anton · 7 years ago
  53. f2662f0 Revert "Rewrite WebRtcSession media tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  54. 3df5dca Rewrite WebRtcSession media tests as PeerConnection tests by Steve Anton · 7 years ago
  55. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  56. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/media/base/fakemediaengine.h]
  57. 2475ae2 Simplify passing video coded factories in media engine by magjed · 7 years ago
  58. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  59. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  60. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  61. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  62. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  63. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  64. eb02c03 Allow WebRtcMediaEngine to be created from any thread. by deadbeef · 7 years ago
  65. f79ade1 Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 7 years ago
  66. d72098a Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 7 years ago
  67. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 7 years ago
  68. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  69. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 7 years ago
  70. 55c5be0 Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel. by solenberg · 7 years ago
  71. c16fa5e Replace all use of the VERIFY macro. by nisse · 7 years ago
  72. dc2b3f3 Delete unused class CompositeMediaEngineWithFakeVoiceEngine. by nisse · 7 years ago
  73. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  74. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  75. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  76. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  77. 95aa964 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  78. 39ce11f Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
  79. f6bcac5 Support external audio mixer in webrtc. by gyzhou · 8 years ago
  80. ebbe4f2 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. by zhihuang · 8 years ago
  81. acd935b Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ ) by nisse · 8 years ago
  82. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  83. 7341ab8 Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ ) by nisse · 8 years ago
  84. 45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  85. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  86. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  87. 84ef615 Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. by aleloi · 8 years ago
  88. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  89. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 8 years ago
  90. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  91. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 8 years ago
  92. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  93. 05b9803 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 8 years ago
  94. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 8 years ago
  95. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  96. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  97. a1c548b Add RtpHeaderExtension to avoid client breakage by isheriff · 8 years ago
  98. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  99. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  100. 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