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gerrit-public.fairphone.software / platform / external / webrtc / 44b6762a12a9cfb1f615942baa63668e4423b07d / pc / rtpsender.h
  1. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  2. aaaf1cf Revert "Remove nogncheck and add proper dependencies." by Patrik Höglund · 7 years ago
  3. eefd543 Remove nogncheck and add proper dependencies. by Patrik Höglund · 7 years ago
  4. c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
  5. c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
  6. e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
  7. 02ee47c Signal track ID correctly when Unified Plan semantics selected by Steve Anton · 7 years ago
  8. f9381f0 Implement PeerConnection::AddTrack/RemoveTrack for Unified Plan by Steve Anton · 7 years ago
  9. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  10. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  11. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/rtpsender.h]
  12. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  13. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  14. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  15. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  16. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  17. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  18. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago