I'm facing a real case of DSP application. I have a basic knowledge of digital filtering and telecommunication issues.
The input of my "black-box" is a digital audio signal (say a common WAV file, stereo 44100 Hz, 16-bit). I need to develop a digital equalizer and create the output audio samples.
The computation has to be done in real-time, i.e. with a live-stream. I'm aware that it means I have to define a buffer large enough to feed the filters.
The basic approach is to create several pass-band digital filters to change the gain of the frequencies in each band. Because the requested bands are a lot (> 16) I'm going to work in the frequency-domain.
- convert the signal in frequency-domain (DFT)
- define a shape of the desired response
- process the input samples with this shape
- revert back to time-domain to retrieve the actual audio samples
- what is the technical name of this method?
- do you think this approach is better(*) than use a lot of steep pass-band filters?
(*) better here is not just an opinion: I'm talking about code complexity, latency and accuracy