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gerrit-public.fairphone.software / platform / external / webrtc / c3d0da097cd5040c6108d099f0291b3e138db0cb / call / DEPS
  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/DEPS]
  3. 16adf03 Recently we moved webrtc/base to webrtc/rtc_base, so these by mbonadei · 7 years ago
  4. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  5. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  6. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  7. cc91d28 Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  8. 94a23f0 Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 8 years ago
  9. 56cf60e Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 8 years ago
  10. 086f851 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 8 years ago