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gerrit-public.fairphone.software / platform / external / webrtc / 2281823a5515834b4f988891c3ca59ceddb097d1 / pc / rtpsender.h
  1. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  2. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  3. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/rtpsender.h]
  4. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  5. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  6. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  7. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  8. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  9. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  10. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago