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gerrit-public.fairphone.software / platform / external / webrtc / c1b36669db55a17ab7049bd2ea68db27653c097c / call / rtp_config.h
  1. fe68daa Add option to configure raw RTP packetization per payload type. by Mirta Dvornicic · 6 years ago
  2. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  3. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  4. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  5. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  6. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  7. 25d31ec Add shared frame id state to RtpVideoSender. by philipel · 6 years ago
  8. dbdb3a0 Refactoring PayloadRouter. by Stefan Holmer · 6 years ago
  9. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  10. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/rtp_config.h]
  11. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago