1. 00ed864 Recently we moved webrtc/base to webrtc/rtc_base, so these by mbonadei · 7 years ago
  2. eb0ab39 Move SrtpSession and tests to their own files. by zstein · 8 years ago
  3. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  4. 138c150 Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 8 years ago
  5. 780e202 Add CreatePeerConnectionFactory overloads that take audio codec factory args by kwiberg · 8 years ago
  6. 25e4ac7 Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  7. 155a5b4 Negotiate H264 profiles in SDP by magjed · 8 years ago
  8. 14ccce8 Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  9. 0b54e5a Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 9 years ago
  10. de82d23 Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 9 years ago
  11. 181e867 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 9 years ago