> 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
