Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / fae640003cf028477819309777ecd29a74c25eff / pc / channel_manager.cc
  1. f5ecb5f Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs"""" by Mirko Bonadei · 4 years, 7 months ago
  2. 9cad4dc Reland "Reland "Reland "Distinguish between send and receive video codecs""" by Johannes Kron · 4 years, 7 months ago
  3. b5159fe Revert "Reland "Reland "Distinguish between send and receive video codecs""" by Olga Sharonova · 4 years, 7 months ago
  4. 4e64e60 Reland "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 4 years, 7 months ago
  5. f9d92ed Revert "Reland "Distinguish between send and receive video codecs"" by Ilya Nikolaevskiy · 4 years, 7 months ago
  6. 77eb338 Reland "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 7 months ago
  7. f2d6fe6 Revert "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 4 years, 8 months ago
  8. 26e6afe Reland "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 8 months ago
  9. f22af3c Revert "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 8 months ago
  10. 18314bd Distinguish between send and receive video codecs by Johannes Kron · 4 years, 8 months ago
  11. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 4 years, 10 months ago
  12. e8e4dc4 Change StartAecDump methods to work with FILE* and FileWrapper by Niels Möller · 5 years ago
  13. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  14. a3aa9bd Make VideoBitrateAllocatorFactory injectable. by Jonas Oreland · 5 years ago
  15. 3256225 "Remove" loophole in rtc::Thread::ScopedDisallowBlockingCalls by Karl Wiberg · 5 years ago
  16. 64b626b Use Abseil container algorithms in pc/ by Steve Anton · 5 years ago
  17. bcd39d4 Creating Simulcast offer and answer in Peer Connection. by Amit Hilbuch · 6 years ago
  18. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  19. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/channelmanager.cc]
  20. 4687915 Enable use of MediaTransportInterface for video streams. by Niels Möller · 6 years ago
  21. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  22. 6eb8a16 Exposing audio and video engines directly. by Sebastian Jansson · 6 years ago
  23. 3c7d599 Replace _stricmp with absl::EqualsIgnoreCase by Niels Möller · 6 years ago
  24. 98a462c Reland "Reland "Propagate media transport to media channel."" by Anton Sukhanov · 6 years ago
  25. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  26. 9accc9f Revert "Reland "Propagate media transport to media channel."" by Oleh Prypin · 6 years ago
  27. da65ed2 Reland "Propagate media transport to media channel." by Anton Sukhanov · 6 years ago
  28. 37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
  29. 8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
  30. a54daf1 Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Benjamin Wright · 6 years ago
  31. 8f4bc41 Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Oleh Prypin · 6 years ago
  32. ac2f3d1 Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h by Benjamin Wright · 6 years ago
  33. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  34. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  35. e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 6 years ago
  36. 95e7dbb Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" by Zhi Huang · 6 years ago
  37. 27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 6 years ago
  38. 97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 6 years ago
  39. ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 6 years ago
  40. 2dfc42d Prepare to make BaseChannel depend on RtpTransportInternal only. by Zhi Huang · 7 years ago
  41. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  42. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  43. c9e1560 Modernize and cleanup ChannelManager by Steve Anton · 7 years ago
  44. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  45. b140b9f Keep count of libsrtp clients, and only deinitialize when it goes to 0. by Taylor Brandstetter · 7 years ago
  46. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  47. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channelmanager.cc]
  48. 774115c Change ChannelManager to use unique_ptr by Steve Anton · 7 years ago
  49. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  50. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  51. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  52. 2f45b6b Remove unused "crypto_options_" field. by jbauch · 7 years ago
  53. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  54. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 7 years ago
  55. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  56. d48f488 Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. by jbauch · 7 years ago
  57. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 7 years ago
  58. 1a2183d Removing unnecessary parameters from CreateXChannel methods. by deadbeef · 7 years ago
  59. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 7 years ago
  60. 7d25426 Delete unneeded includes of base/common.h. by nisse · 8 years ago
  61. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  62. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  63. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  64. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  65. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  66. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  67. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  68. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  69. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  70. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  71. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  72. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  73. ebbe4f2 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. by zhihuang · 8 years ago
  74. ffc6118 Don't cache video codec list in VideoEngine2. by brandtr · 8 years ago
  75. 3cf8ece Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ ) by magjed · 8 years ago
  76. 9f71ec5 Stop caching supported codecs in WebRtcVideoEngine2 by magjed · 8 years ago
  77. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  78. 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
  79. 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
  80. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 8 years ago
  81. 05b9803 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 8 years ago
  82. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 8 years ago
  83. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 8 years ago
  84. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 8 years ago
  85. 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
  86. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  87. ff97631 - Add temporary VoEBase::audio_device_module() method. by solenberg · 8 years ago
  88. b252856 Remove all uses of the HAVE_CONFIG_H define. by Henrik Kjellander · 8 years ago
  89. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 8 years ago
  90. fb45d17 Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/) by Per · 8 years ago
  91. 74622e0 Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ ) by perkj · 8 years ago
  92. e9c0cdf Removed unused cricket::VideoCapturer methods: by perkj · 8 years ago
  93. 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 · 8 years ago
  94. 51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 8 years ago
  95. 65c7f67 Fix license headers in webrtc/pc by kjellander · 8 years ago
  96. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 8 years ago[Renamed (99%) from talk/session/media/channelmanager.cc]
  97. 162c339 Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #14 id:300001 of https://codereview.webrtc.org/1655793003/ ) by perkj · 8 years ago
  98. 4d19c5b This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it. by Per · 8 years ago
  99. 4b2a5a8 Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #12 id:260001 of https://codereview.webrtc.org/1655793003/ ) by perkj · 8 years ago
  100. 2f21789 This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it. by perkj · 8 years ago