9fea80f50d
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers. Audio streams are using a fake audio device with file input. The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code. R=pbos@webrtc.org TBR=kjellander@webrtc.org BUG=webrtc:5263 Review URL: https://codereview.webrtc.org/1542653002 . Cr-Commit-Position: refs/heads/master@{#11171}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.