if you are implementing an FIR filter using "fast convolution" which uses an
FFT and overlap-add or overlap-scrap (often the latter is called
"overlap-save"), then the window is *rectangular*, but if you do this right,
there is no ripple artifact from the windowing of the time-domain data.
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
> On March 7, 2020 6:43 PM zhiguang zhang 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
Yes, essentially you do have the inherent delay involving a window of
samples in addition to the compute time.
On Sat, Mar 7, 2020, 5:40 PM Spencer Russell wrote:
> On Sat, Mar 7, 2020, at 6:00 AM, Zhiguang Eric Zhang wrote:
>
> Traditional FIR/IIR filtering is ubiquitous but actually does
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
Sorry I meant Alan :)
On Wed, Jan 15, 2020, 11:20 PM Alan Wolfe wrote:
> probably pretty basic stuff for most people here but wanted to share a
> writeup and demo i made about FIRs.
>
> Post: https://blog.demofox.org/2020/01/14/fir-audio-data-filters/
>
This is a very cool blog, I need to spend some time with it. It's also
interesting to draw parallels between the graphics stuff that Alex writes
about to the audio realm.
Traditional FIR/IIR filtering is ubiquitous but actually does suffer from
drawbacks such as phase distortion and the inherent
