Added WebRTC audio quality test using fake device; improved fake device
This finally adds a end-to-end WebRTC audio quality test which includes
getUserMedia and the capture path (except, obviously, the os-specific
code which actually interacts with the capture device etc).
This patch also makes the audio input device generic. It will now
convert its input .wav file to suit the audio bus. I had to do this
because the PESQ tests require files to be mono, 16-bit and so on.
BUG=446859
Review URL: https://codereview.chromium.org/
831693004
Cr-Commit-Position: refs/heads/master@{#310736}