Lossless FFT on audio data
FFT needs to sample at at least twice the frequency of your signal, and should not introduce any loss. Loss is introduced on compression or encoding. Your input and outputs should be able to match the frequency, amplitude, and frequencies of the signal you are sampling.

Your biggest challenge is processing the DSP algorithms in real time, which depends on the algorithms you use, how many, and how your processor can keep up with it, with minimal delay. Like fractions of a second, to stay in sync with anything else (Video, other audio, or live instruments).

You need to do some calculations on what you need, through testing, before you can find out if your platform is good enough, or where you need to rethink or cut around.