Not to threadjack from Alan Wolfe, but the FFT EQ was responsive written in C and running on a previous gen MacBook Pro from 2011. It wouldn't have been usable in a DAW even without any UI. It was running FFTW.
As far as linear / zero-phase, I didn't think about the impulse response but what you might get are ripple artifacts from the FFT windowing function. Otherwise the algorithm is inherently zero-phase. On Sat, Mar 7, 2020, 7:11 PM robert bristow-johnson < r...@audioimagination.com> wrote: > > > > On March 7, 2020 6:43 PM zhiguang zhang <zhiguangezh...@gmail.com> > wrote: > > > > > > Yes, essentially you do have the inherent delay involving a window of > samples in addition to the compute time. > > yes. but the compute time is really something to consider as a binary > decision of whether or not the process can be real time. > > assuming your computer can process these samples real time, the delay of > double-buffering is *twice* the delay of a single buffer or "window" (i > would not use that term, i might use the term "sample block" or "sample > frame") of samples. suppose your buffer or sample block is, say, 4096 > samples. while you are computing your FFT (which is likely bigger than > 4K), multiplication in the frequency domain, inverse FFT and overlap-adding > or overlap-scrapping, you are inputting the 4096 samples to be processed > for your *next* sample frame and you are outputting the 4096 samples of > your *previous* sample frame. so the entire delay, due to block > processing, is two frame lengths, in this case, 8192 samples. > > now if the processing time required to do the FFT, frequency-domain > multiplication, iFFT, and overlap-add/scrap exceeds the time of one frame, > then the process cannot be real time. but if the time required to do all > of that (including the overhead of interrupt I/O-ing the samples in/out of > the blocks) is less than that of a frame, the process is or can be made > into a real-time process and the throughput delay is two frames. > > > > On Sat, Mar 7, 2020, at 6:00 AM, Zhiguang Eric Zhang wrote: > > > ... FFT filtering is essentially zero-phase ... > > FFT filtering **can** be linear-phase (which is zero-phase plus a constant > delay) if the FIR filter impulse response is designed to be linear-phase > (or symmetrical). it doesn't have to be linear phase. > > -- > > r b-j r...@audioimagination.com > > "Imagination is more important than knowledge." > > > On Sat, Mar 7, 2020, 5:40 PM Spencer Russell <s...@media.mit.edu> wrote: > > > On Sat, Mar 7, 2020, at 6:00 AM, Zhiguang Eric Zhang wrote: > > > > Traditional FIR/IIR filtering is ubiquitous but actually does suffer > from drawbacks such as phase distortion and the inherent delay involved. > FFT filtering is essentially zero-phase, but instead of delays due to > samples, you get delays due to FFT computational complexity instead. > > > > > > I wouldn’t say the delay when using FFT processing is due to > computational complexity fundamentally. Compute affects your max throughput > more than your latency. In other words, if you had an infinitely-fast > computer you would still have to deal with latency. The issue is just that > you need at least 1 block of input before you can do anything. It’s the > same thing as with FIR filters, they need to be causal so they can’t be > zero-phase. In fact you could interchange the FFT processing with a bank of > FIR band pass filters that you sample from whenever you want to get your > DFT frame. (that’s basically just a restatement of what I said before about > the STFT.) > > > > > > -s > _______________________________________________ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > > https://urldefense.proofpoint.com/v2/url?u=https-3A__lists.columbia.edu_mailman_listinfo_music-2Ddsp&d=DwIGaQ&c=slrrB7dE8n7gBJbeO0g-IQ&r=w_CiiFx8eb9uUtrPcg7_DA&m=Rfp1ozFiVA9OFrUgHnk8k4E7erWKoT0p7iI83xAoYEo&s=PQF0MzrvvmNpczp2_8SZoGEb7ojFOWxu-5BZ5mBpcb0&e=
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