: Monday, May 14, 2012 10:36 AM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Window presum synthesis
Hi, everyone, apologies for the delay...was on a short vacation ;)
On 25 April 2012 02:22, Wen Xue mr.x@gmail.com wrote:
Yes, it's very right that you can't recover a and b from a+b
.
w.x.
-Original Message- From: Domagoj Saric
Sent: Monday, May 14, 2012 10:36 AM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Window presum synthesis
Hi, everyone, apologies for the delay...was on a short vacation ;)
On 25 April 2012 02:22, Wen Xue mr.x@gmail.com
On 14/05/2012 10:45, Domagoj Saric wrote:
..
If you don't mind an honest suggestion: having looked at the Csound
source and mostly seeing the decades old C-style coding it would IMO
probably be better if you try and skip reinventing as many wheels (e.g.
CUDA, vectorization...) as possible with
Thanks for the info about the presum code in pvoc.c. I tried this technique in
Matlab, but I must be doing something wrong with the sinc windows - I got all
the expected doubling and echoing. Any additional hints on making this work,
without reverse-engineering the C code?
It's too bad there's
On Mon, Apr 23, 2012 at 2:57 AM, Domagoj Šarić dsar...@gmail.com wrote:
On 20 April 2012 17:15, Charles Henry czhe...@gmail.com wrote:
Don't let it bother you too much. I can tell by looking at it--This
is a stupid algorithm.
I sort of regret those words--it just seems so basic in terms of
On 20 April 2012 17:15, Charles Henry czhe...@gmail.com wrote:
Don't let it bother you too much. I can tell by looking at it--This
is a stupid algorithm.
It does seem strange and counterintuitive at first glance but it's
hard to just simply dismiss it thus once you've seen it examined in
: Saturday, April 21, 2012 12:30 AM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Window presum synthesis
Alessandro Saccoia alessandro.sacc...@gmail.com wrote:
http://web.archive.org/web/20060513150136/http://archive.chipcenter.com/dsp/DSP000315F1.html
The images haven't been archived
On 20 April 2012 19:30, Theo Verelst theo...@tiscali.nl wrote:
Hi
The theoretical background probably most appropriate for the FFT...
snip
Also, averaging the output of the IFFT is indeed some sort of a FIR low
frequency filter which cannot exactly do what you'd normally want (a great
smooth
On 20 April 2012 23:50, Chris Cannam chris.can...@eecs.qmul.ac.uk wrote:
Mark Dolson's CARL phase vocoder
(http://www.crca.ucsd.edu/cmusic/cmusic.html) from 1984 uses window
presum if the FFT is shorter than the window, and includes
resynthesis.
I suppose the idea is to reduce some of the
On 21 April 2012 01:30, s...@sfxmachine.com wrote:
This method is also discussed in Crochiere Rabiner's Multirate Digital
Signal Processing book, but it didn't make sense to me there either - I'm
assuming this is my problem, not theirs. Apparently this method windows
the input with a window
On 23 April 2012 12:20, Domagoj Šarić dsar...@gmail.com wrote:
More importantly, yes, the code does have resynthesis and the way it
avoids the flanging (or echo for larger frame sizes) artefacts (that
you get from adding the frame-size-delayed copy of the signal to
itself) is by applying a
On 23/04/2012 11:28, Domagoj Šarić wrote:
...
So Crochiere's book discusses synthesis? Unfortunately I don't have it...
In any case, simple periodic extension and windowing didn't work for
me (it produces echo/flanging as one might expect)...
I'm afraid to try this, because it doesn't make
be
observed, a similar illusion.
-Original Message-
From: Domagoj Šarić
Sent: Monday, April 23, 2012 11:40 AM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Window presum synthesis
On 23 April 2012 10:52, Wen Xue mr.x@gmail.com wrote:
Time-aliasing is just another
On 23/04/2012 11:49, Domagoj Šarić wrote:
On 23 April 2012 12:20, Domagoj Šarićdsar...@gmail.com wrote:
More importantly, yes, the code does have resynthesis and the way it
avoids the flanging (or echo for larger frame sizes) artefacts (that
you get from adding the frame-size-delayed copy of
And for people with a lot of time on their hands, we added the Sliding
DFT version (analysis frame updated every sample) a few years ago -
including a simple instance of frequency domain convolution for the
windowing. A real-time version of this was presented at ICMC last year,
running on a GPU.
Hello,
I haven't read your post in detail but
ps. I've seen this article
http://archive.chipcenter.com/dsp/DSP000315F1.html often being
mentioned as explaining it all but unfortunately the site no longer
exists…
always check archive.org for pages that are gone...
On Fri, Apr 20, 2012 at 8:15 AM, Domagoj Šarić dsar...@gmail.com wrote:
...but unfortunately all of these however seem to stop at the
analysis stage (i.e. we've gotten a more precise spectrum and that's
all we need). I've wraped my head around the idea itself and the
analysis procedure, I
On 20 April 2012 14:15, Domagoj Šarić dsar...@gmail.com wrote:
...but unfortunately all of these however seem to stop at the
analysis stage (i.e. we've gotten a more precise spectrum and that's
all we need).
Mark Dolson's CARL phase vocoder
(http://www.crca.ucsd.edu/cmusic/cmusic.html) from
Alessandro Saccoia alessandro.sacc...@gmail.com wrote:
http://web.archive.org/web/20060513150136/http://archive.chipcenter.com/dsp/DSP000315F1.html
The images haven't been archived, but you could still find it a useful
reference.
This link includes the images:
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