Dear FIR dreamers,
from my FIR convolution experience there are some hurdles to pass.
The hardware requirements for convolution is about the computing performance of a RPI 2B BruteFIR for 2 channels @192kHz, 65000taps. The miniDSP AD-chip, which is specialised on DSP calculations, provides e.g. in the OpenDRC for 2 channels @48kHz, 6144taps (which is already at the lower end of good results). For normal convolution the number of taps results in a frequency resolution per tap over the bandwith, e.g. 24kHz/6144taps, which is not very accurate at low frequencies. Some DSP therefore use IIR-PEQs for room modes.
For a symmetric algorithm the peak is in the middle of the impuls response, which means that there is a filter dependent delay for the sound reproduction, e.g. 65000/2 samples @48kHz result in about 0.7 seconds (so far from lip sync).
The next field of trouble is the filter generation. If there is a substantial time/phase correction (for which FIR filter are used for) with a high gradient, then the filter may produce preringing which results in an unpleasant noise before the tone. It is not easy to deal with these - you need lot of experience or spend many adaption loops. I do not know a one button software with convenient results. Dirac comes close, but takes a high licence fee.