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gerrit-public.fairphone.software / platform / external / webrtc / e507b0ce8e87dc5f0a1292ee1654ea9d2b1053c7 / pc / channel.h
  1. ee01a83 Remove MetricsObserverInterface. by Qingsi Wang · 6 years ago
  2. 66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 6 years ago
  3. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  4. 0e36a72 Delete unused class CurrentSpeakerMonitor. by Niels Möller · 6 years ago
  5. 0327c2d Move VideoStreamEncoderInterface to api/. by Niels Möller · 6 years ago
  6. c6ce9c5 New file api/video/BUILD.gn by Niels Möller · 6 years ago
  7. 365381f Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
  8. e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 7 years ago
  9. 95e7dbb Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" by Zhi Huang · 7 years ago
  10. 27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
  11. 97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
  12. ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
  13. db67ba1 Report SRTP error codes to UMA by Steve Anton · 7 years ago
  14. 0807d15 Remove more dead code from BaseChannel by Steve Anton · 7 years ago
  15. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  16. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
  17. e2a9318 Delete ConnectionMonitor. by Niels Möller · 7 years ago
  18. 0228485 Delete MediaMonitor. by Niels Möller · 7 years ago
  19. 053c1f8 Delete unused signal VoiceChannel::SignalAudioMonitor. by Niels Möller · 7 years ago
  20. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  21. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  22. dc8b5ab Remove dead code for media channel errors by Steve Anton · 7 years ago
  23. 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
  24. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  25. 593e325 Change RTCStatsCollector to only access channels from signaling thread by Steve Anton · 7 years ago
  26. 9a44f96 Delete rtc_base/window.h. by Niels Möller · 7 years ago
  27. 3828c06 Replace cricket::ContentAction with webrtc::SdpType by Steve Anton · 7 years ago
  28. 2dfc42d Prepare to make BaseChannel depend on RtpTransportInternal only. by Zhi Huang · 7 years ago
  29. cd3fc5d Use the DtlsSrtpTransport in BaseChannel. by Zhi Huang · 7 years ago
  30. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  31. 1d88d74 Remove the unused code. by Zhi Huang · 7 years ago
  32. 942bc2e Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  33. 8c316c1 Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal." by Zhi Huang · 7 years ago
  34. 7167745 Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  35. c99b6c7 Remove the SetEncryptedHeaderExtensionIds methods. by Zhi Huang · 7 years ago
  36. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  37. 6b63cd5 Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests by Steve Anton · 7 years ago
  38. b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 7 years ago
  39. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  40. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  41. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  42. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channel.h]
  43. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
  44. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  45. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
  46. 634977b SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 7 years ago
  47. e8ab543 Make BaseChannel::rtp_transport_ a unique_ptr. by zstein · 7 years ago
  48. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  49. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  50. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  51. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  52. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  53. 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
  54. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  55. 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
  56. 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
  57. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  58. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
  59. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  60. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  61. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  62. d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 8 years ago
  63. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
  64. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  65. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  66. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  67. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  68. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  69. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  70. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  71. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  72. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  73. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  74. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  75. 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
  76. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  77. 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
  78. 45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  79. d89ab14 Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit. by johan · 8 years ago
  80. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  81. 062ce9f Combining "SetTransportChannel" and "SetRtcpTransportChannel". by deadbeef · 8 years ago
  82. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
  83. 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 8 years ago
  84. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  85. 6bb1ef2 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 8 years ago
  86. 184a3fd Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 8 years ago
  87. 6379793 Removing obsolete method from channel.h. by deadbeef · 8 years ago
  88. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 8 years ago
  89. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  90. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  91. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 8 years ago
  92. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  93. dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 8 years ago
  94. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  95. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  96. 52dce73 Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  97. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  98. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  99. 194e3bc Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 9 years ago
  100. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago