there had been a mistake in my structure which caused the phase to be
set to zero
now it sounds more like the original when there is no pitch shift applied
(which is a good indicator that there is something wrong when it does not)
https://soundcloud.com/traumlos_kalt/freq-domain-pv-shift-test
Am 28.10.2018 um 22:28 schrieb gm:
I am thinking now that resetting the phase to the original when the
amplitude exceeds the previous value
is probably wrong too, because the phase should be different when
shifted to a different bin
if you want to preserve the waveshape
I am not sure about
Thanks for tip, I had a brief look at this paper before.
I think the issue it adresses is not the problem I encounter now.
But it might be interesting again at a later stage or if I return to the
time domain pitch shift.
This is how I do it now, it seems simple & correct but I am not 100% sure,
Am 29.10.2018 um 05:43 schrieb Ethan Duni:
You should have a search for papers by Jean Laroche and Mark Dolson,
such as "About This Phasiness Business" for some good information on
phase vocoder processing. They address time scale modification mostly
in that specific paper, but many of the i
Am 29.10.2018 um 19:12 schrieb gm:
From the structure displayed in the book, he adds two neighbouring
complex numbered bins,
multiplied. That is, he multiplies their real and imaginary part
respectivly
and adds that to the values of the bin - (Fig 9.18 p. 293).
Unfortunately this is not
But why is there no artefact of this kind when the signal is only stretched,
but not shifted?
Am 29.10.2018 um 19:50 schrieb Scott Cotton:
On Mon, 29 Oct 2018 at 19:12, gm <mailto:g...@voxangelica.net>> wrote:
Am 29.10.2018 um 05:43 schrieb Ethan Duni:
> You should ha
Unfortunately I would have to stick with the "sliding" PD phase locking
structure from the book for now,
iterating through the spectrum to search for peaks and identify groups
will add too many frames of additional latency in Reaktor.
But for me this method unfortunately defintively gave wo
Ok, heres a final idea, can't test any of this so it's pure science fiction:
-Take a much larger FFT spectrogramme offline, with really fine overlap
granularity.
-Take the cesptrum, identify regions/groups of transients by new peaks
in the cepstrum.
-Pick peaks in the spectrum, by amplitud
Am 30.10.2018 um 16:30 schrieb gm:
-Compress the peaks (without the surrounding regions) and noise into
smaller spectra.
(but how? - can you simply add those that fall into the same bins?)
snip...
I am curious about the spectrum compression part, would this work and
if not why not
Original Message
Subject: [music-dsp] two fundamental questions Re: FFT for realtime
synthesis?
From: "gm"
Date: Tue, October 30, 2018 8:17 pm
To: music-dsp@music.columbia.edu
-
Thanks for your time
My question rephrased:
Lets assume a spectrum of size N, can you create a meaningfull spectrum
of size N/2
by simply adding every other bin together?
Neglecting the artefacts of the forward transform, lets say an
artificial spectrum
(or a spectrum after peak picking that
Now the synth works quite well with an FFT size of 4096, I had a severe bug
all the time which was messing every other frames phase up.
I have simple peak picking now for sines+noise synthesis
which sounds much nicer when the sound is frozen.
It's a peak if its larger then two adjacent bins a
Am 02.11.2018 um 21:40 schrieb gm:
Any other ideas?
ok the answer is already in my post: just analyze backwards
It's possibly part of a transient when the backwards tracked partial
stops to exist.
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An I think you can model them simply by adding their phasors/bins/numbers...
for opposite angles they will cancel, for the same angle they will be
amplified
so the model is correct at the center of the window, but it models just
an instance in time and spreads this instance
in this way of
h the fact that you need two successive spectra
to represent he same information
but I dont really see the effect of that other than it has a better time
resolution
Am 03.11.2018 um 10:48 schrieb Ross Bencina:
[resending, I think I accidentally replied off-list]
On 1/11/2018 5:00 AM, gm wrote:
Am 04.11.2018 um 03:03 schrieb Theo Verelst:
It might help to understand why in this case you'd chose for the
computation according to a IFFT scheme for synthesis. Is it for
complimentary processing steps, efficiency, because you have data that
fits the practical method in terms of granula
Maybe you could make the analysis with a filterbank, and do the
resynthesis with FFT?
Years ago I made such a synth based on "analog" Fourier Transforms,
(the signal is modulated and rotated down to 0 Frequency and that
frequencies around DC are lowpass filtered
depending on the bandwitdh y
to go, with some
refinements-
Am 04.11.2018 um 14:55 schrieb gm:
Maybe you could make the analysis with a filterbank, and do the
resynthesis with FFT?
Years ago I made such a synth based on "analog" Fourier Transforms,
(the signal is modulated and rotated down to 0 Frequenc
Am 04.11.2018 um 17:00 schrieb gm:
ok I now I tried a crude and quick multiresolution FFT analysis at log
2 basis
I half the window size (Hann) for every FFT.
To compensate for the smaller window, I multiply by the factor that it
is smaller, that is 2, 4, 8,
But it appears that there
aren't, in my opinion, a
solved problem.
Scott
On Sun, 4 Nov 2018 at 19:55, gm <mailto:g...@voxangelica.net>> wrote:
Am 04.11.2018 um 17:00 schrieb gm:
>
> ok I now I tried a crude and quick multiresolution FFT analysis
at log
> 2 basis
I half
Am 04.11.2018 um 23:49 schrieb Scott Cotton:
On Sun, 4 Nov 2018 at 22:50, gm <mailto:g...@voxangelica.net>> wrote:
note that in polyphonic sources, transients may only apply to one of
the sources, so
if you define transient as a slice of time say of a percussive onset,
like guit
bear with me, I am a math illiterate.
I understand you can do a Discrete Fourier Transform in matrix form,
and for 2-point case it is simply
[ 1, 1
1,-1]
like the Haar transform, average and difference.
My idea is, to use two successive DFT frames, and to transform
resepctive bins of two su
Am 05.11.2018 um 01:39 schrieb robert bristow-johnson:
mr. g,
I think what you're describing is the Cooley-Tukey Radix-2 FFT algorithm.
yes that seems kind of right, though I am not describing something but
posting a question actually
and the "other thing" was an answer to a question
may
Am 05.11.2018 um 01:56 schrieb gm:
so you do the "radix 2 algorithm" if you will on a subband, and now what?
the bandlimits are what? the neighbouring upper and lower bands?
how do I get a frequency estimate "in between" out of these two real
values that describe the u
Am 05.11.2018 um 16:17 schrieb Ethan Fenn:
Of course it's possible you'll be able to come up with a clever
frequency estimator using this information. I'm just saying it won't
be exact in the way Cooley-Tukey is.
Maybe, but not the way I laid it out.
Also it seems wiser to interpolate sp
The background of the idea was to get a better time resolution
with shorter FFTs and then to refine the freuqency resolution.
You would think at first glance that you would get the same time resolution
as you would with the longer FFT, but I am not sure, if you do overlaps
you get kind of a sli
blurred too much I assume.
Unfortunately I am not sure what quality can be achieved and where the
limits are with this approach.
Am 06.11.2018 um 14:20 schrieb Ross Bencina:
On 7/11/2018 12:03 AM, gm wrote:
A similar idea would be to do some basic wavelet transfrom in octaves
for instance and th
window length from that
for that band?
I understand that bandwitdh is inversly proportional to window length.
So it seems very easy actually but I am stuck here...
Am 06.11.2018 um 16:13 schrieb gm:
At the moment I am using decreasing window sizes on a log 2 scale.
It's still pretty bl
I think I figured it out.
I use 2^octave * SR/FFTsize -> toERBscale -> * log2(FFTsize)/42 as a
scaling factor for the windows.
Means the window of the top octave is about 367 samples at 44100 SR -
does that seem right?
Sounds better but not so different, still pretty blurry and somewhat
re
Am 06.11.2018 um 19:35 schrieb gm:
I use 2^octave * SR/FFTsize -> toERBscale -> * log2(FFTsize)/42 as a
scaling factor for the windows.
Means the window of the top octave is about 367 samples at 44100 SR -
does that seem right?
ok this was wrong...
it should be just windowsi
Am 07.11.2018 um 12:32 schrieb gm:
but when I use windowsize = 1/bandwitdhERB I get windows that are too
small for
the phase vocoder, for instance for the lowest band I get
bandwidthERB (~10Hz) ~= 26 Hz bandwidth, ~ 1705 samples window length
This gives too much modulation or too much
This is brining up my previous question again, how do you decimate a
spectrum
by an integer factor properly, can you just add the bins?
the orginal spectrum represents a longer signal so I assume folding
of the waveform occurs? but maybe this doesn't matter in practice for
some applications?
ve to apply a sinc filter before and then discard every other bin?
If so, can this be done with an other FFT like a cepstrum on the bins?
If anyone knows of an easy explanation of down- and up sampling spectra
it would be much appreciated.
Am 09.11.2018 um 19:16 schrieb Ethan Duni:
gm wrote:
>
--
Subject: Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for
realtime synthesis?
From: "gm"
Date: Fri, November 9, 2018 4:19 pm
To: music-dsp@music.columbia.edu
--
>
> hm, my applica
o do time scaling. lemme know if that might be helpful.
L8r,
r b-j
Original Message
Subject: Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for
realtime synthesis?
From: &quo
Am 10.11.2018 um 00:19 schrieb gm:
FFT size is 4096, and now I search for ways to improve it, mostly
regarding transients.
But I am not sure if that's possible with FFT cause I still have
pre-ringing, and I cant see
how to avoid that completely cause you can only shorten the windows on
I wonder if anyone has thought about this?
I am aware that it may have little practical use and may actually worsen the
"fractal noise" behaviour at higher feedback levels.
(Long ago I tested this with a tuned delay in the feedback path and
thats what I recall)
But still I am interested.
If
Hi
I am looking for a combined patent and media or software rights lawyer
in Berlin, Germany.
Can someone recommend somebody?
I am not sure if this is too OT for the list, but I guess its too dirty
laundry in all details.
And too sad, really.
So no story. Although the story would possible
I need some possibly quotable real world opinions and experiences on how
long stuff
can take to design or develop, especially takeing Hofstadter's Law into
account
For instance reverberators, hard to estimate, and I dont recall all the
times I spent exactly
I tried so many things on differe
Am 22.05.2020 um 03:55 schrieb robert bristow-johnson:
i just relate the lengths of the delays by some ratio and then look
for prime numbers. isn't that what Jot did? i don't remember.
I don't know, I am bad with names and original sources.
But prime numbers have common multiples. I think we
you can answer here or on surveymonkey if you prefer
https://www.surveymonkey.de/r/56R8RJH
location:
type of employment (employed, freelance / self employed, other):
rates per hour (or per day):
estimated ratio paid/unpaid work:
job description (examples of what you do, or did, typcially
ey-pay-you replys, thanks
Am 22.05.2020 um 21:27 schrieb gm:
you can answer here or on surveymonkey if you prefer
https://www.surveymonkey.de/r/56R8RJH
location:
type of employment (employed, freelance / self employed, other):
rates per hour (or per day):
estimated ratio paid/unpaid work:
job
now here is the link to the results page:
https://de.surveymonkey.com/results/SM-8LXS53ZN7/
questions are still open, I didn't set any date limit, assume they will
stay open:
https://www.surveymonkey.de/r/56R8RJH
___
dupswapdrop: music-dsp mailing
Am 26.05.2020 um 15:14 schrieb Theo Verelst:
"I need some possibly quotable real world opinions and experiences on how
long stuff can take to design or develop"
Sound like that's interesting. But why? Project management, funding,
hobby schedule,
historic insight, or .. ?
During the past 1
answers to some questions that had been asked:
1. "Rates per day" including tax or not?
Before any taxes etc., since these vary widely and depend on lots of things
2. Paid/Unpaid Ratio: What does 0 or 100 mean? Does 0 mean 0 unpaid work?
typical lapse of mine: I actually ment percentage...
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