Sampo, it is not the poles and zeros that alternate on the real line but
the poles of the two all-pass filter paths. The 90 deg phase difference
band is almost from 0 to Nyquist. In my filter pair they are from 0.001 pi
to 0.999 pi. On z-plane those corner frequencies are at (0.95, 0.003)
and
> how do you quadrature modulate without Hilbert filters?
>
Perhaps I'm using the wrong term - the operation in question is just the
multiplication of a signal by e^jwn. Or, equivalently, multiplying the real
part by cos(wn) and the imaginary part by sin(wn) - a pair of "quadrature
oscillators."
Original Message
Subject: Re: [music-dsp] � 45� Hilbert transformer using pair of IIR APFs
From: "Ethan Duni" <ethan.d...@gmail.com>
Date: Thu, February 9, 2017 5:06 pm
To: "A discussion list for music-re
On Tue, Feb 7, 2017 at 6:49 AM, Ethan Fenn wrote:
> So I guess the general idea with these frequency shifters is something
> like:
>
> pre-filter -> generate Hilbert pair -> multiply by e^iwt -> take the real
> part
>
> Am I getting that right?
>
Exactly, this is a
Am 09.02.2017 um 14:15 schrieb Theo Verelst:
The idea of estimating a single sine wave frequency, amplitude and
phase with a short and easy as possible filter appeals to me though.
Did you listen to the example I posted? Do you think it's useful? Or too
many artefacts?
On 2017-02-07, Theo Verelst wrote:
Like with many transforms, I can't help but practically think that
it's hard to make a tradeoff between the meaning of the results, such
as [...]
Here there's an rather simple optimization criterion: a constant 45
degree phase offset, or perhaps a pair of
Thinking about it, I recall there was some from of transform used for frequency/time
analysis for instance for radar problems (maybe books from before WWII, or more recent
frequency/time analyzers) and without checking though it was in popular DSP speak
something like the Hilbert transform, but
On 2017-02-06, robert bristow-johnson wrote:
[...] and analytic signal
a(t) = x(t) + j y(t)
= g(t) cos(w t) + j g(t) sin(w t)
= g(t) e^(j w t)
the analytic envelope is
|a(t)| = sqrt( x(t)^2 + y(t)^2 )
= g(t)
so that works great for a
Yes, there are lots of interesting things that can be done with
frequency shifting. Feedback suppression in a PA system by frequency
shifting was suggested by Manfred Schroeder a long time ago. I have
occasionally found it to be useful to broaden a mono signal by feeding
it through a hilbert
A nice thing are the endless phase shifts, if you feed back a frequency
shifter. It’s like a Shepard tone.
If you have Logic Pro or MainStage, try the RingShifter, it can do such tricks.
It has 2x6 Allpass filters for the constant phase shift and a quadrature
oscillator with FM and a delay.
On 02/07/2017 07:49 AM, Ethan Fenn wrote:
So I guess the general idea with these frequency shifters is something like:
pre-filter -> generate Hilbert pair -> multiply by e^iwt -> take the
real part
Am I getting that right?
Now that I think about it, another application might be in stereo
So I guess the general idea with these frequency shifters is something like:
pre-filter -> generate Hilbert pair -> multiply by e^iwt -> take the real
part
Am I getting that right?
Now that I think about it, another application might be in stereo imaging.
Start with a mono signal, generate the
On Feb 6, 2017, at 11:19 PM, robert bristow-johnson wrote:
> with a harmonic musical note this beat frequency would be the difference
> between two harmonics, which could be any integer times the fundamental.
> that's much faster than beating. an r.m.s. envelope would be the square root
>
On 2/7/17 12:39 AM, Eric Brombaugh wrote:
On Feb 6, 2017, at 8:24 PM, robert bristow-johnson wrote:
On 02/05/2017 07:52 PM, robert bristow-johnson wrote:
I'm curious what aspects of a music make the complex magnitude of the
analytic signal inappropriate for estimating the envelope? In
On Feb 6, 2017, at 8:24 PM, robert bristow-johnson wrote:
> > On 02/05/2017 07:52 PM, robert bristow-johnson wrote:
> >>
> >> > I'm curious what aspects of a music make the complex magnitude of the
> >> analytic signal inappropriate for estimating the envelope? In
> >> communications signal
If there are zeroes on the unit circle in the z-plane, I don't think it
will be a very good all-pass filter... (Because some sinusoid frequencies
would be blocked entirely)
Neat topic so far! I'd love to see an illustration of this real-line
"zebra stripe" design. Also wondering what sort of
�
Original Message
Subject: Re: [music-dsp] � 45� Hilbert transformer using pair of IIR APFs
From: "Eric Brombaugh" <ebrombau...@cox.net>
Date: Mon, February 6, 2017 11:37 am
To: music-dsp@mu
On 2017-02-06, Eric Brombaugh wrote:
well, with a single sinusoid, there should be no intermodulation product
so the analytic envelope should be exactly correct. but consider:
[...]
I might be way off base here, but... As Olli said, both the poles and
the zeroes sorta "like to be" on the
On 02/05/2017 07:52 PM, robert bristow-johnson wrote:
> I'm curious what aspects of a music make the complex magnitude of the
analytic signal inappropriate for estimating the envelope? In
communications signal processing we use this often, even for signals
that are fairly wide-band with
Funny that no one mentioned this
https://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/VAFilterDesign_1.1.1.pdf
Particularly, formula 7.43
Regards,
Vadim
--
Vadim Zavalishin
Reaktor Application Architect
Native Instruments GmbH
+49-30-611035-0
www.native-instruments.com
Original Message
Subject: Re: [music-dsp] � 45� Hilbert transformer using pair of IIR APFs
From: "Eric Brombaugh" <ebrombau...@cox.net>
Date: Sun, February 5, 2017 8:22 pm
To: "A discussion list for music-re
On Feb 5, 2017, at 12:54 PM, robert bristow-johnson wrote:
> using the analytic filter to get the instantaneous amplitude envelope (and,
> also, instantaneous frequency by differentiating phase) is something that
> works only with single sinusoids that are AM'd or FM'd. for music, i think i
Original Message
Subject: Re: [music-dsp] � 45� Hilbert transformer using pair of IIR APFs
From: "Ethan Fenn" <et...@polyspectral.com>
Date: Sun, February 5, 2017 1:39 pm
To: music-dsp@mu
nking about how to approach that. APFs so the gains stay 1 and we don't
> care about the phase as long as the difference is 90°. still don't know
> how i might set up an optimization problem. probably the best measure is
> the negative frequency rejection with regard to the positive frequ
problem. probably the best measure is the negative frequency
> rejection with regard to the positive frequency gain.
>
> r b-j
>
>
>
> ------------ Original Message
> Subject: Re: [music-dsp] ± 45° Hilbert transformer using
typofix: "and their companion poles" -> "and their companion zeros"
-olli
On Sun, Feb 5, 2017 at 1:52 PM, Olli Niemitalo wrote:
> 90 deg phase difference all-pass filter pairs... Lemme wave my hands a bit:
>
> It's been years, but I recall I first tried a structure with complex
>
90 deg phase difference all-pass filter pairs... Lemme wave my hands a bit:
It's been years, but I recall I first tried a structure with complex
conjugate pairs of poles (and their companion poles to make the filters
all-pass). Globally optimizing that using Differential Evolution, the poles
robert bristow-johnson wrote:
cool, so it appears from your comments to be real poles.
complex conjugate poles are not considered in the design. i
wonder why that is?
I’m not the author of the formula, I just implemented it
and don’t understand how it produces these filters.
I guess the
�
cool, so it appears from your comments to be real poles. �complex conjugate
poles are not considered in the design. �i wonder why that is?
r b-j
Original Message
Subject: Re: [music-dsp] � 45� Hilbert transformer using pair of IIR
robert bristow-johnson wrote:
so Olli, how do you get your coefficients? (if i may ask?)
If I may reply, I wrote long time ago a C++ library implementing
this kind of polyphase halfband filter, including a coefficient
calculator involving elliptic stuff (way beyond my knowledge).
Check the
-McClellan or Prony). maybe that's what
> Olli is doing.
>
>
> --
>
> r b-j r...@audioimagination.com
>
> "Imagination is more important than knowledge."
>
>
>
> ------------ Original Message -------
-------------- Original Message --------
>
> Subject: Re: [music-dsp] � 45� Hilbert transformer using pair of IIR APFs
>
>
From: "Eric Brombaugh" <ebrombau...@cox.net>
>
> Date: Sat, February 4, 2017 8:55 pm
>
> To: musi
[music-dsp] � 45� Hilbert transformer using pair of IIR APFs
From: "Eric Brombaugh" <ebrombau...@cox.net>
Date: Sat, February 4, 2017 8:55 pm
To: music-dsp@music.columbia.edu
--
> The original Csound sou
The original Csound source has a set of coefficients for this type of
hilbert transform but they don't say how the coefficients were derived
and the performance is fairly lackluster. The only reference I've come
across for this was Olli Niemitalo's:
http://yehar.com/blog/?p=368
He gives a
�
hey guys,
what's the best online reference you can tell me for theory and practice of
designing APF pairs with � 45� phase (+ a linear phase that represents
causality delay)�to result in a Hilbert pair for audio processing. �i know how
to do this for FIR
and to use half-band symmetry, but
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