Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

[Ross Bencina]

On 3/11/2018 3:41 AM, Ethan Fenn wrote:
No length of FFT will distinguish between a mixture of these sine waves 
and a single amplitude-modulated one, because they're mathematically 
identitical! Specifically:


sin(440t) + sin(441t) = 2*cos(0.5t)*sin(440.5t)

So the question isn't whether an algorithm can distinguish between them 
but rather which one of these two interpretations it should pick. And I 
would say in most audio applications the best answer is that it should 
pick the same interpretation that the human hearing system would. In 
this example it's clearly the right-hand side. In the case of a large 
separation (e.g. 440Hz and 550Hz, a major third) it's clearly the 
left-hand side. And somewhere in between I guess it must be a toss-up.


I guess you could model both simultaneously, with some kind of 
probability weighting.


Ross.
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Re: [music-dsp] transient detection Re: FFT for realtime synthesis?

[gm]

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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[music-dsp] transient detection Re: FFT for realtime synthesis?

[gm]

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 and if it was a peak 
candidate or peak in either
three bins in the frame before - since this would miss all the onsets of 
peaks
the analysis is done backwards with the unwanted side effect that the 
last bin of a fading

frquency is recognized as a random part, but you dont seem to notice that.

I use random amplitudes for the noise, but keep the phases
so they are not out of phase when a frequency changes from bin to bin.

What I want to do now is some analysis that detects whether or not
a phase should be reset to the original phase for a transient.

I am not sure if the idea is good or not, since the window is rather large
so it will have the wrong phase before the transient.

Any ideas for that? I thought of taking the cepstrum and look for new peaks
to identify new tones. But I think this idea is flawed cause it misses
the actual broadband transient and comes in too late.

A simple time transient detector on the other hand would reset all partials
which is not wanted either.

Any other ideas?


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Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

[Ethan Fenn]

>
> In any case, most signals are not sums of stationary sinusoids. And since
> signals are typically buried in noise, or superimposed on top of each
> other, so the problem is not well posed. For two very simple examples:
> consider two stable sine waves at 440Hz and 441Hz -- you will need a very
> long FFT to distinguish this from a single amplitude-modulated sine wave?
> or consider a sine wave plus white noise -- the accuracy of frequency and
> phase recovery will depend on how much input you have to work with.


This is an interesting thing to think about in any sort of spectral
modeling.

No length of FFT will distinguish between a mixture of these sine waves and
a single amplitude-modulated one, because they're mathematically
identitical! Specifically:

sin(440t) + sin(441t) = 2*cos(0.5t)*sin(440.5t)

So the question isn't whether an algorithm can distinguish between them but
rather which one of these two interpretations it should pick. And I would
say in most audio applications the best answer is that it should pick the
same interpretation that the human hearing system would. In this example
it's clearly the right-hand side. In the case of a large separation (e.g.
440Hz and 550Hz, a major third) it's clearly the left-hand side. And
somewhere in between I guess it must be a toss-up.

-Ethan



On Wed, Oct 31, 2018 at 2:00 PM, gm  wrote:

> 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 discards the region around the
> peaks)
>
> Lets say two sinusoids in two adjacent bins, will summing them into a
> single bin of a half sized spectrum
> make sense and represent them adequately?
> In my limited understanding, yes, but I am not sure, and would like to
> know why not
> if that is not the case.
>
>
>
>
>
> I'm not sure what "compress" means in this context, nor am I sure what
>> "fall together" means. But here's some points to note:
>>
>> A steady state sine wave in the time domain will be transformed by a
>> short-time fourier transform into a spectral peak, convolved (in the
>> frequency domain) by the spectrum of the analysis envelope. If you know
>> that all of your inputs are sine waves, then you can perform "spectral peak
>> picking" (AKA MQ analysis) and reduce your signal to a list of sine waves
>> and their frequencies and phases -- this is the sinusoidal component of
>> Serra's SMS (explained in the pdf linked above).
>>
>> Note that since a sinusoid ends up placing non-zero values in every FFT
>> bin, you'd need to account for that in your spectral estimation, which
>> basic MQ does not -- hence it does not perfectly estimate the sinusoids.
>>
>> In any case, most signals are not sums of stationary sinusoids. And since
>> signals are typically buried in noise, or superimposed on top of each
>> other, so the problem is not well posed. For two very simple examples:
>> consider two stable sine waves at 440Hz and 441Hz -- you will need a very
>> long FFT to distinguish this from a single amplitude-modulated sine wave?
>> or consider a sine wave plus white noise -- the accuracy of frequency and
>> phase recovery will depend on how much input you have to work with.
>>
>> I think by "compression" you mean "represent sparsely" (i.e. with some
>> reduced representation.) The spectral modeling approach is to "model" the
>> signal by assuming it has some particular structure (e.g. sinusoids+noise,
>> or sinusoids+transients+noise) and then work out how to extract this
>> structure from the signal (or to reassemble it for synthesis).
>>
>> An alternative (more mathematical) approach is to simply assume that the
>> signal is sparse in some (unknown) domain. It turns out that if your signal
>> is sparse, you can apply a constrained random dimensionality reduction to
>> the signal and not lose any information. This is the field of compressed
>> sensing. Note that in this case, you haven't recovered any structure.
>>
>> Ross
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
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Re: [music-dsp] Roland AX-Synth

[Roshan Wijetunge]

Looks like she's using something like this to send midi to another bit of
kit

https://www.thomann.de/gb/panda_audio_midibeam.htm

Tosh

---

Roshan ‘Tosh’ Wijetunge

Musician / Producer / Engineer

www.wijproductions.com
rwi...@googlemail.com :: +44 (0) 7787 783 783 :: @wijproductions




On Fri, 2 Nov 2018 at 04:41, robert bristow-johnson <
r...@audioimagination.com> wrote:

>
>
> do you folks know if the Roland AX-Synth has a real-time vocal pitch shift
> function like the Digitech Vocalist or the TC-Helicon VoiceTone?  i know
> that Korg and Roland and Novation had/have other synths with a vocoder of
> some sort in it.  i know that, back in the olden daze, Roland had a similar
> cheapie Voice Transformer.
>
> it's just that Imogen Heap is playing an AX and doing vocal shifting in
> this video: https://www.youtube.com/watch?v=Th0qY3d6oHA .  but there is
> nothing about this function in any of the descriptions of the AX-Synth.
> the AX was meant to be totally portable and battery powered.  does that
> mean there is a remote box that can output MIDI from what she plays on the
> keyboard?  it appears she has some kinda transmitter taped to the bottom of
> her AX.
>
> i was just curious what she uses to do vocal pitch shifting and can't
> quite figure it out what it is.
>
>
> --
>
> r b-j r...@audioimagination.com
>
> "Imagination is more important than knowledge."
>
>
>
>
>
>
>
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Re: [music-dsp] Book: The Art of VA Filter Design 2.1.0

[Vadim Zavalishin]

On 01-Nov-18 16:16, Fabian-Robert Stöter wrote:
I appreciate that but it would still be nice if your book could be cited 
appropriately. Have you thought about putting it on arxiv or zenodo.org 
?
This would give you the possibility to version the book and make folks 
from academia happy with a proper DOI reference.


Nobody complained so far, but I will consider your suggestion, thank you!

Best regards,
Vadim

--
Vadim Zavalishin
Reaktor Application Architect
Native Instruments GmbH
+49-30-611035-0

www.native-instruments.com
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Re: [music-dsp] Book: The Art of VA Filter Design 2.1.0

[Vadim Zavalishin]

On 02-Nov-18 03:06, Andrew Simper wrote:
If you prize symmetry then you can use a cascade with 2 x one pole HP 
and 2 x one pole LP to make a 4 pole BP (band pass) then you can use the 
same old FIR based output tap mixing to generate all the different 
responses. It may not be so easy to do in a real circuit, but in 
software we're not bound by what is easy to build :)


https://cytomic.com/files/dsp/cascade-all-to-all-responses.pdf


Symmetry is one of the things. The other is the shape of the amplitude 
response. I'm personally not convinced by the -4dB dip prior to the 
resonance, although YMMV. At any rate it doesn't qualify as a "bread and 
butter" LP IMHO ;) With BP8 it's getting way worse.


Incidentally, another way to come at more or less the same structure is 
raising the orders of LP and HP filters (by stacking identical 1-poles 
in series) in the transposed Sallen-Key (Fig.5.23 of the book). Since 
TSK is essentially a bandpass ladder with a special output mode, it's 
actually the same. Further ways (originating at lowpass ladder) to look 
at this can be found here:

https://www.kvraudio.com/forum/viewtopic.php?p=6844369#p6844369
and here
https://www.kvraudio.com/forum/viewtopic.php?p=6844470#p6844470


Regards,
Vadim

--
Vadim Zavalishin
Reaktor Application Architect
Native Instruments GmbH
+49-30-611035-0

www.native-instruments.com
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