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webrtc / src / 2c72fe8a2a0e540a66028254b26e2409f9360cd8 / pc / webrtcsession.cc
  1. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  2. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/pc/webrtcsession.cc]
  3. 2a5e426 Reject the descriptions that attempt to change the order of m= sections by Zhi Huang · 8 years ago
  4. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 8 years ago
  5. 169629a Change WebRtcSession to have a vector of channels by Steve Anton · 8 years ago
  6. 24efa72 Fix RTCP transport not destroyed when channel creation fails by Steve Anton · 8 years ago
  7. ea5cc86 Delete unneeded include of videocapturer.h by Niels Möller · 8 years ago
  8. 300bf8e Reinstate "API for periodically regathering ICE candidates" by Steve Anton · 8 years ago
  9. 3beb207 Revert "API for periodically regathering ICE candidates" by Magnus Jedvert · 8 years ago
  10. aa41f0c API for periodically regathering ICE candidates by Steve Anton · 8 years ago
  11. d8cf08f Don't call CreateDtlsTransport_n from non-network thread in WebRtcSession by deadbeef · 8 years ago
  12. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  13. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  14. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  15. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  16. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  17. 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 · 8 years ago
  18. 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 · 8 years ago
  19. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 8 years ago
  20. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  21. 81bf7b0 Pass ownership of candidate to PeerConnection::OnIceCandidate by jbauch · 8 years ago
  22. 6dfd53a Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange by zstein · 8 years ago
  23. b789253 Accept SDP with TRANSPORT attributes missing from bundled m= sections. by deadbeef · 8 years ago
  24. 1a2183d Removing unnecessary parameters from CreateXChannel methods. by deadbeef · 8 years ago
  25. 5107246 Allow applications to limit the ICE check rate through RTCConfiguration by skvlad · 8 years ago
  26. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  27. 7ce109a Replace the easy cases of VERIFY usage. by nisse · 8 years ago
  28. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/webrtcsession.cc]
  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. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  33. a9dd4a1 Replace left-over ASSERTs in httpcommon.h and webrtcsession.cc. by nisse · 8 years ago
  34. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  35. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  36. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  37. eb4ca4e Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  38. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  39. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  40. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  41. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  42. fe4a8a4 Implement current/pending session description methods. by deadbeef · 8 years ago
  43. df6075a RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 8 years ago
  44. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  45. d1a38b5 Implement the "needs-ice-restart" logic for SetConfiguration. by deadbeef · 8 years ago
  46. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  47. ebbe4f2 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. by zhihuang · 8 years ago
  48. daf88b1 Removing ERROR message for something that's expected to occur. by deadbeef · 8 years ago
  49. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  50. d82eee0 Log how often DTLS negotiation failed because of incompatible ciphersuites. by zhihuang · 9 years ago
  51. 9763d56 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 9 years ago
  52. 907abe4 Revert of Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. (patchset #8 id:280001 of https://codereview.webrtc.org/2166873002/ ) by deadbeef · 9 years ago
  53. 34b54c3 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 9 years ago
  54. 29ff844 Add PeerConnection IsClosed check. by zhihuang · 9 years ago
  55. 5622c5e If continual gathering is enabled, by Honghai Zhang · 9 years ago
  56. e985111 Adding API for "presume writable when fully relayed" ICE option. by Taylor Brandstetter · 9 years ago
  57. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  58. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 9 years ago
  59. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  60. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 9 years ago
  61. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  62. 184a3fd Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 9 years ago
  63. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  64. d03c23b Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 9 years ago
  65. d7973cc Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ ) by hbos · 9 years ago
  66. 400781a Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 9 years ago
  67. 6aefc63 Move the ICE state transition ASSERTS to a lower level. by Taylor Brandstetter · 9 years ago
  68. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 9 years ago
  69. e9021a3 Propogate network-worker thread split to api by danilchap · 9 years ago
  70. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  71. a1c3035 Relanding: Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 9 years ago
  72. c55fb30 Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ ) by deadbeef · 9 years ago
  73. 48e9d05 Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 9 years ago
  74. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
  75. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 9 years ago
  76. 1c7fdd8 Remove calls to ScopedToUnique and UniqueToScoped by kwiberg · 9 years ago
  77. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  78. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  79. b4d01c4 A bunch of interfaces: Return scoped_ptr<SSLCertificate> by kwiberg · 9 years ago
  80. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  81. 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
  82. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  83. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  84. 7fb69db Reland the CL to remove candidates when doing continual gathering by Honghai Zhang · 9 years ago
  85. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  86. 6f59a4f Revert of Remove candidates when doing continual gathering (patchset #15 id:560001 of https://codereview.webrtc.org/1648813004/ ) by tommi · 9 years ago
  87. 84430da When doing candidate re-gathering in the same generation, Remove the existing local candidate on the same network by honghaiz · 9 years ago
  88. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  89. 049fbb1 Renaming variables in p2ptransportchannel to be consistent. by Honghai Zhang · 9 years ago
  90. a2a49d9 This CL provides interfaces that do not use constraints for by hta · 9 years ago
  91. 36f0137 Implement Turn/Turn first logic for connection selection. by guoweis · 9 years ago
  92. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  93. 03d6d57 Late initialize MediaController, for less resource i.e. ProcessThread, usage by PeerConnection. by solenberg · 9 years ago
  94. 0db023a Move suspend_below_min_bitrate from VideoOptions to MediaConfig. by nisse · 9 years ago
  95. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  96. 7324eb9 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago
  97. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  98. 0ed85b2 Track pending ICE restarts independently for different media sections. by deadbeef · 9 years ago
  99. 51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago
  100. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago