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gerrit-public.fairphone.software / platform / external / webrtc / 34029e209d6d116c338e7d65d6980cc395184658 / call / rtp_transport_controller_send.cc
  1. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  2. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/rtp_transport_controller_send.cc]
  3. 5c8942a Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
  4. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  5. 76e62b0 Address some violations of chromium-style. by nisse · 7 years ago
  6. 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 7 years ago
  7. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 7 years ago