Hi, GM.

On 3 February 2018 at 18:39, gm <g...@voxangelica.net> wrote:

>
> If your goal is to isolate the lowest partial, why dont you use the
> measured frequency to steer a lowpass or lowpass/bandpass filter?
>
​I'm already piloting a lowpass in my current algorithm but I was thinking
to use a bandpass too, also for another algorithm which detects the loudest
partial. I haven't properly thought of that but a first idea was to have
the Q of the BP inversely proportional to the derivative of the output of
the integrator: that way, when the integrator is not changing we are around
the detected partial and we can then select even further by increasing the
Q. Though, I should also think of a way to "release" the Q when the input
signal changes or the system might enter an attractor.

Currently, in the algorithm that I described earlier, there is a very tiny
amount of energy added to the high spectrum of the crossover; tiny enough
to be negligible for the measurement, but big enough to move the cutoff up
when there is no energy in the crossover's input. So if I have a detected
lowest at 1k plus a bunch of partials at 10k and higher, if I remover the
1k partial the system will move to 10k.

>
> For my time domain estimator I use
>
> 4th order Lowpass, 2nd order BP -> HilbertTransform -> Phasedifferenz ->
> Frequency
>  |________________________cutoff________________________
> ____________________|
>
> ​I'm not familiar with the technique which uses the HT and the phase
difference to calculate the frequency. I'd be very grateful if you could
say a few words about that.​ Can you also control the SNR/sensitivy in your
system?

>
> This gives you an estimate (or rather measurement) per sample.
>
> A further improvement is to calculate the sub-sample time between phase
> wraparounds,
> this basically elimates any spurios modulations from the phase within a
> cycle,
> simlar to an integrator.
>
> You can have several measurements per cycle again by adding angle offsets
> to the phase and calculating the time between wraprounds of the new angles
> as well.
>
> I use SVFs for the filters with Q set to a Butterworth response for the LPs
> and Q=2 for the Bandpass.
>
> I dont know if this method has less overhead than yor method
> because you need the Hilbert Transform, but the prefiltering is more
> efficient
>
> Depending on your input sorces you can try to exchange the HT with a
> single allpass with adaptive corner frequency
>
​
All great ideas, thanks a lot for sharing. Have you published any paper on
this or other time-domain algorithms for feature-extration? The reason why
I implemented that algorithm is that it will probably be used in a
time-domain noisiness estimator which I'm working on and that I will
perhaps share here if it gets somewhere.

Cheers,
Dario


>
> Am 03.02.2018 um 14:49 schrieb Dario Sanfilippo:
>
> Thanks for the links, Steven!
>
> Vadim, what is the title of your book? We may have it here at uni.
>
> Hi, Robert. I'm working on some time-domain feature-extraction algorithms
> based on adaptive mechanisms. A couple of years ago, I implemented a
> spectral tendency estimator where the cutoff of a crossover (1p1z filters)
> is piloted by the RMS imbalance of the two spectra coming out of the same
> crossover. Essentially, a negative feedback loop for the imbalance pushes
> the cutoff towards the predominant spectrum until there's a "dynamical
> equilibrium" point which is the estimated tendency.
>
> A recent extension to that algorithm was to add a lowpass filter within
> the loop, at the top of the chain, as shown in this diagram:
> https://www.dropbox.com/s/a1dtk0ri64acssc/lowest%20partial.jpg?dl=0.
> (Some parts necessary to avoid the algorithm from entering attractors have
> been omitted.)
>
> If the same spectral imbalance also pilots the cutoff of the lowpass
> filter, we have a nested positive (the lowpass strengthens the imbalance
> which pushes the cutoff towards the same direction) and negative (the
> crossover's dynamical equilibrium point) feedback loop. So it is a
> recursive function which removes partials from top to bottom until there is
> nothing left to remove except the lowest partial in the spectrum.
>
> The order and type of the lowpass (I've tried 1p1z ones, cascading up to
> four of them), I believe, is what determines the SNR in the system, so what
> the minimum amplitude of the bottom partial should be to be considered
> signal or not. Large transition bands in the lowpass will affect the result
> as the top partials which are not fully attenuated will affect the
> equilibrium point. Since elliptic filters have narrow transition bands at
> low orders, I thought that they could have given more accurate results,
> although the ripples in the passing band would also affect the SNR of the
> system.
>
> Perhaps using Butterworth filters could be best as the flat passing band
> could make it easier to model a "threshold/sensitivity" parameter. With
> that regard, I should also have a look at fractional order filters. (I've
> quickly tried by linearly interpolating between filters of different orders
> but I doubt that that's the precise way to go.)
>
> Of course, an FFT algorithm would perhaps be easier to model, though this
> time-domain one should be CPU-less-expensive, not limited to the bin
> resolution, and would provide a continuous estimation not limited to the
> FFT period.
>
> I've tested the algorithm and it seems to have a convincing behaviour for
> most test signals, though it is not too accurate in some specific cases.
>
> Any comment on how to possibly improve that is welcome.
>
> Thanks,
> Dario
>
>
> Dario Sanfilippo - Research, Teaching and Performance
> Reid School of Music, Edinburgh University
> +447492094358 <07492%20094358>
> http://twitter.com/dariosanfilippo
> http://dariosanfilippo.tumblr.com
>
> On 3 February 2018 at 08:01, robert bristow-johnson <
> r...@audioimagination.com> wrote:
>
>> i'm sorta curious as to what a musical application is for elliptical
>> filters that cannot be better done with butterworth or, perhaps, type 2
>> tchebyshev filters?  the latter two are a bit easier to derive closed-form
>> solutions for the coefficients.
>>
>> whatever.  (but i am curious.)
>>
>> --
>>
>> r b-j                  r...@audioimagination.com
>>
>> "Imagination is more important than knowledge."
>>
>>
>>
>> ---------------------------- Original Message ----------------------------
>> Subject: Re: [music-dsp] Elliptic filters coefficients
>> From: "Dario Sanfilippo" <sanfilippo.da...@gmail.com>
>> Date: Fri, February 2, 2018 6:37 am
>> To: music-dsp@music.columbia.edu
>> ------------------------------------------------------------
>> --------------
>>
>>
>> > Thanks, Vadim.
>> >
>> > I don't have a math background so it might take me longer than I wished
>> to
>> > obtain the coefficients that way, but it's probably time to learn it.
>> With
>> > that regard, would you have a particularly good online resource that
>> you'd
>> > suggest for pole-zero analysis and filter design?
>> >
>> > Thanks to you too, Shannon.
>> >
>> > Best,
>> > Dario
>> >
>> > On 1 February 2018 at 11:16, Vadim Zavalishin <
>> > vadim.zavalis...@native-instruments.de> wrote:
>> >
>> >> Hmm, the Wikipedia article on elliptic filters has a formula to
>> calculate
>> >> the poles and further references the Wikipedia article on elliptic
>> rational
>> >> functions which effectively contains the formula for the zeros.
>> Obtaining
>> >> the coefficients from poles and zeros should be straightforward.
>> >>
>> >> Regards,
>> >> Vadim
>> >>
>> >>
>> >> On 01-Feb-18 12:00, Dario Sanfilippo wrote:
>> >>
>> >>> Hello, everybody.
>> >>>
>> >>> I was wondering if you could please help me with elliptic filters. I
>> had
>> >>> a look online and I couldn't find the equations to calculate the
>> >>> coefficients.
>> >>>
>> >>> Has any of you worked on that?
>> >>>
>> >>> Thanks,
>> >>> Dario
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> dupswapdrop: music-dsp mailing list
>> >>> music-dsp@music.columbia.edu
>> >>> https://lists.columbia.edu/mailman/listinfo/music-dsp
>> >>>
>> >>>
>> >> --
>> >> Vadim Zavalishin
>> >> Reaktor Application Architect
>> >> Native Instruments GmbH
>> >> +49-30-611035-0
>> >>
>> >> www.native-instruments.com
>> >> _______________________________________________
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>> >>
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