Hello to the list!
I am working in a time stretching effect using a standad phase vocoder
approach.
Basically I have an input parameter to control the time stretch ratio,
which can change in real time from 0.5 ( sinthesys hop size = 1/2 analysys
hop size, x200% original time stretch) to 2.0
( sin
Maybe I'm missing something obvious, but shouldn't the filter bank itself
be constant? I.e., no change in the overlap or windowing. The time
stretch/compression is obtained by extrapolating/interpolating the analysis
parameters, not by shifting around the synthesis filter bank relative to
the analy
Thanks for your quick reply Ethan.
Indeed I was missing some relevant information on my first email. I am
using the FFT / IFFT approach instead of the filter bank one. So I made my
FFT frame size and analysis hop size fixed, achiving time stretching by
changing the synthesis hop size by the time s
completely the problem
with a small number of extra operations compared to the fft/ifft.
Best,
Alessandro
-Original Message-
From: "Daniel Varela"
Sent: 19/08/2014 23:49
To: "A discussion list for music-related DSP"
Subject: Re: [music-dsp] Real time variable time st
It's been a while since I've thought about phase vocoders, and I've never
gotten real hands-on in building a time stretch algorithm, so let me make
sure I understand what you're doing:
On the analysis side you've got a fixed frame size and a fixed hop size. On
the synthesizer side, you have the sa
The classic Phase Vocoder algorithm integrates the phase difference
between successive analysis frames to produce the phase component in
the synthesis frames. No explicit interpolation is needed. This is not
the same as results achieved by time domain methods. It seems that
reconstruction is never
> The classic Phase Vocoder algorithm integrates the phase difference
> between successive analysis frames to produce the phase component in
> the synthesis frames. No explicit interpolation is needed.
To do time-stretching, you then have to multiply that unwrapped phase by
the ratio of the hop si
Sorry for the late reply, long day at work...
Alessandro, working in a sample by sample approach in order to remove the
overlap add distortion is brilliant. Since I can compute the extact
distortion introduced, it would be possible to remoe it from the output
signal. Will try that and
share the re
Hi Daniel,
I have been playing with precisely this scheme recently so I went back
to my code and had a look at the reconstruction. As you say, in the
ideal case, the windowing is completely transparent and if you skip
the fft and related steps you will achieve a 'perfect' sample for
sample reconst
