Filtering of streaming audio
FFT -> zeroing coefficients -> IFFT is not the correct way of doing filtering - the actual filter realized by doing so has poor characteristics.

The correct way of filtering signals is to compute the coefficients of a digital filter, a process known as filter design and for which a large body of software tools/documentation is available, and apply it to your input sequence. In short, this consists in evaluating for each sample a linear combination of the past input samples, and the past output samples. Depending on the requirements of your filter in terms of stop-band rejection / ripple, only a few coefficients could be necessary, making it way more efficient than FFT. Since the only information required to compute an output sample is the few past input/output samples, there is no problem to apply it to streaming audio.

You will need to use FFT only if you decide to go with FIR filters, and if your filter requirements cause them to have an insanely large number of coefficients. In this particular case, it will be efficient to apply the filter to successive blocks of your input data through FFT and overlap-add.