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webrtc / src / 2c72fe8a2a0e540a66028254b26e2409f9360cd8 / pc
  1. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  2. 581df61 Revert "Reland "Clean up libjingle API dependencies."" by Patrik Höglund · 7 years ago
  3. 5117b04 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  4. b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 8 years ago
  5. d8970db Delete unneeded includes of fileutils.h by Niels Möller · 8 years ago
  6. 7bcfc3b Revert "Clean up libjingle API dependencies." by Patrik Höglund · 8 years ago
  7. bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 8 years ago
  8. 57fb315 Clean up libjingle API dependencies. by Patrik Höglund · 8 years ago
  9. 3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 8 years ago
  10. 94286cb Add base fixture and PeerConnection wrapper for unit tests by Steve Anton · 8 years ago
  11. 02e7a19 Remove unnecessary video factory references in PeerConnectionFactory by Magnus Jedvert · 8 years ago
  12. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 8 years ago
  13. 835cc0c Remove unnecessary audio references in PeerConnectionFactory by Magnus Jedvert · 8 years ago
  14. 4e2deab Only return stats for the most recent unsignaled audio stream. by deadbeef · 8 years ago
  15. b19012e Remove the support of fallback from DTLS to SDES. by zhihuang · 8 years ago
  16. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 8 years ago
  17. 1d4db39 Revert of If SRTP sessions exist, don't create new ones when applying answer. (patchset #1 id:1 of https://codereview.webrtc.org/3019443002/ ) by henrika · 8 years ago
  18. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  19. d45aea8 Serialize "a=x-google-flag:conference". by deadbeef · 8 years ago
  20. 5ada7ac If SRTP sessions exist, don't create new ones when applying answer. by deadbeef · 8 years ago
  21. 58b0316 Expose new video codec factories in the PeerConnectionFactory API by Magnus Jedvert · 8 years ago
  22. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 8 years ago
  23. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 8 years ago
  24. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  25. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago