We require a simple app that processes a microphone signal and plays it through the speakers in real time with low latency. This will eventually be used as a module in a more comprehensive app that we will develop.

The microphone signal needs to be modified on a per-sample basis, so the app needs direct access to the microphone signal buffers. The exact processing that is done is not important, we will replace it with our own code later. For this project, simply scale each sample by a factor of 2.

The tools used for the core of the app should be C++ with Core Audio + Audio Units. Other languages like Swift can be used for non-processing parts of the app (UI, etc).

The app can have a UI or be config/command-line driven (whatever is easiest). Any UI that is created will eventually be replaced by our own UI. We just need input/output device selection and frame length specification (either 8ms, 10ms, or 16ms).

Round-trip latency needs to be low: less than 20ms on 1-10 year old macbooks.

This is a proof-of-concept…