We're looking for someone interested in helping us prototype a concept for a new type of audio conferencing application. Here is the situation:
We have a stereo microphone that outputs 48k/16-bit audio as a USB device when plugged into a USB OTG port on an Android phone. This audio can be recorded as stereo in any basic Android recording application.
What we'd like to investigate is the possibility of compressing that audio with Opus (http://opus-codec.org/) and sending it as a stereo MediaStream over WebRTC (https://developer.mozilla.org/en-US/docs/Web/API/Media_Streams_API) at adjustable bitrates (from low to insane).
A successful use case would be: user 1 connects stereo microphone connected to an Android phone over USB > opens up simple Android App > select quality (low or high) > enters a server URL > application connects over WebRTC and begins streaming stereo audio to the peer(s) via Chrome or Firefox
You'd be working with me (an 'OK' software engineer but a decent back-end server architect) and a professional c++ programmer.
Note: Currently it looks like Apple does not support WebRTC natively in Chrome, I expect this changes very soon. Bonus points if you have any idea how to solve this!
Note 2: see www.cleanfeed.net for an example of the quality possible over WebRTC and stereo.
I am willing to pay higher rates for the most experienced freelancers