Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 84ae2b6efd12d818e492c304247c6852b3cd614a / call / rtp_stream_receiver_controller.cc
  1. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  2. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  3. ed09dc6 Don't check MIDs when demuxing RTP packets in Call by Steve Anton · 7 years ago
  4. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  5. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  6. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/rtp_stream_receiver_controller.cc]
  7. 5daecca Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ ) by eladalon · 7 years ago
  8. 59b603f Revert of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #15 id:280001 of https://codereview.webrtc.org/2968693002/ ) by zhihuang · 7 years ago
  9. 7b7e06f SSRC and RSID may only refer to one sink each in RtpDemuxer by eladalon · 7 years ago
  10. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  11. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  12. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  13. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago