WebRTC: Use the MediaStream Recording API for the audio_quality_browsertest.
This CL uses the MediaStream Recording API to record the audio received
by the right tag. It saves it as a webm file that is later converted to a
wav file using ffmpeg.
This might cause a performance regression because we're using webm+ffmpeg
to generate the wav file. I set a, hopefully high enough, bitrate to
4 files changed