> On March 8, 2020 10:05 AM Ethan Duni <ethan.d...@gmail.com> wrote: > > > It is physically impossible to build a causal, zero-phase system with > non-trivial frequency response.
a system that operates in real time. when processing sound files you can pretend that you're looking at some "future" samples. i guess that would be acausal, so you're right, Ethan. -- r b-j r...@audioimagination.com "Imagination is more important than knowledge." > > > On Mar 7, 2020, at 7:42 PM, Zhiguang Eric Zhang <zez...@nyu.edu> wrote: > > > > 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://lists.columbia.edu/mailman/listinfo/music-dsp
