Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

2020-04-13 Thread Spencer Russell
On Mon, Apr 13, 2020, at 1:36 PM, Spencer Russell wrote: > > Andreas - is this the general approach you use for Gaborator? > Whoops, just clicked through to the documentation and it looks like this is the track you're on also. I'm curious if you have any insight into the window-

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

2020-04-13 Thread Spencer Russell
On Fri, Mar 20, 2020, at 4:58 PM, Andreas Gustafsson wrote: > robert bristow-johnson wrote: but i would be excited to see a good > > implementation of constant Q filterbank that is very close to > > perfect reconstruction if the modification in the frequency domain > > is null. > > Isn't this

Re: [music-dsp] FIR blog post & interactive demo

2020-03-10 Thread Spencer Russell
on each frame 7. overlap-add each frame to get the resulting time-domain signal See below for more. On Mon, Mar 9, 2020, at 5:44 PM, robert bristow-johnson wrote: > > > On March 9, 2020 10:15 AM Spencer Russell wrote: > > > > > > I think we're mostly on the same

Re: [music-dsp] FIR blog post & interactive demo

2020-03-09 Thread Spencer Russell
. -s On Mon, Mar 9, 2020, at 12:52 AM, Ethan Duni wrote: > > > On Sun, Mar 8, 2020 at 8:02 PM Spencer Russell wrote: >> In fact, the the standard STFT analysis/synthesis pipeline is the same thing >> as overlap-add "fast convolution" if you: >> >>

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Spencer Russell
On Sun, Mar 8, 2020, at 7:41 PM, Ethan Duni wrote: > FFT filterbanks are time variant due to framing effects and the circular > convolution property. They exhibit “perfect reconstruction” if you design the > windows correctly, but this only applies if the FFT coefficients are not > altered

Re: [music-dsp] FIR blog post & interactive demo

2020-03-07 Thread Spencer Russell
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

Re: [music-dsp] FIR blog post & interactive demo

2020-03-04 Thread Spencer Russell
On Tue, Mar 3, 2020, at 4:21 PM, robert bristow-johnson wrote: > > Like a lotta things, sometimes people use the same term to mean something > different. A "phase vocoder" (an STFT thing a la Portnoff) is not the same as > a "channel vocoder" (which is a filter bank thing). It’s maybe worth

Re: [music-dsp] phase vocoder

2018-10-15 Thread Spencer Russell
Alex, A number of experienced DSP engineers have spent considerable time trying to help you understand the problem you're describing, yet it doesn't seem like you've made much progress. Your questions often seem to end up with asking folks to basically write your code for you. I don't want to be

Re: [music-dsp] Finding discontinuity in a sine wave.

2018-01-10 Thread Spencer Russell
I think the PLL approach will be much more robust, and will let you detect phase changes. -s On Wed, Jan 10, 2018, at 11:51 AM, Benny Alexandar wrote: > Here is what I was planning. The sine wave frequency is known. > > For example if sine wave is having a frequency of 1 kHz and sampling >

Re: [music-dsp] Seminar: Listening and Learning Systems for Composition and Live Performance (by Nick Collins)

2012-05-30 Thread Spencer Russell
Looks like an extremely interesting seminar. Unfortunately I'm US-based and won't be able to make it (though 90 EUR would be a bargain!). Will any slides/materials/videos be posted online for the general public? Thanks, Spencer On Wed, May 30, 2012 at 9:54 PM, Charlie Morrow c...@cmorrow.com