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gerrit-public.fairphone.software / platform / external / webrtc / c72af93cff8e79e97987c3bee1493668b7826e37 / pc / rtptransport_unittest.cc
  1. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  2. 942bc2e Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  3. 8c316c1 Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal." by Zhi Huang · 7 years ago
  4. 7167745 Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 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/pc/rtptransport_unittest.cc]
  7. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  8. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
  9. 634977b SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 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. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  14. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 8 years ago