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gerrit-public.fairphone.software / platform / external / webrtc / 0e96535be97916d8fcaa9873ffab3c636539f9d8 / call / rtp_config.h
  1. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  2. a0e9943 Negotiation of LNTF controls instantiation of RTPSenderVideo::rtp_sequence_number_map_ by Elad Alon · 5 years ago
  3. fadb181 Negotiate use of RTCP loss notification feedback (LNTF) by Elad Alon · 5 years ago
  4. fe68daa Add option to configure raw RTP packetization per payload type. by Mirta Dvornicic · 5 years ago
  5. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  6. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  7. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  8. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  9. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  10. 25d31ec Add shared frame id state to RtpVideoSender. by philipel · 6 years ago
  11. dbdb3a0 Refactoring PayloadRouter. by Stefan Holmer · 6 years ago
  12. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  13. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/rtp_config.h]
  14. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago