A simple way to stabilize most quadrature oscillators including
Martin's quadrature oscillator is to multiply each state variable by a
temporary variable:
g = 1.5 - 0.5*(u*u + v*v)
where u and v are unit-amplitude quadrature oscillator outputs. The
correction does not need to be done very
On Thu, Feb 21, 2019 at 11:16 PM robert bristow-johnson
wrote:
> But Martin, if you let this thing run for days on end, would not eventually
> the amplitude of the output change a bit?
Short answer: yes, sometimes significantly for audio purposes when
using 32-bit float state variables, but
On Fri, Feb 22, 2019 at 9:08 AM robert bristow-johnson <
r...@audioimagination.com> wrote:
> i just got in touch with Olli, and this "triangle wave to sine wave"
> shaper polynomial is discussed at this Stack Exchange:
>
>
>
>
Thomas,
you could try https://github.com/mangledjambon/drumbooth to separate
sinusoidal and impulse-like parts of sounds, and then do your additive
analysis on the sinusoidal part only. The deconstruction is based on Derry
FitzGerald HARMONIC/PERCUSSIVE SEPARATION USING MEDIAN FILTERING, Proc. of
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
typofix: "and their companion poles" -> "and their companion zeros"
-olli
On Sun, Feb 5, 2017 at 1:52 PM, Olli Niemitalo <o...@iki.fi> wrote:
> 90 deg phase difference all-pass filter pairs... Lemme wave my hands a bit:
>
> It's been years, but I recall
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
To make a reference microphone from scratch, one can use reciprocity
calibration. You take three uncalibrated mics that can act as
low-power speakers as well. Not all mics are suitable for this. Then
you pair the three mics the three possible ways. For each pair, the
first mic of the pair will act
will be satisfied. At least that part of your
theory seems consistent.
-olli
On Wed, Jul 16, 2014 at 1:39 PM, Vadim Zavalishin
vadim.zavalis...@native-instruments.de wrote:
On 16-Jul-14 12:31, Olli Niemitalo wrote:
What does O(B^N) mean?
-olli
This is the so called big O notation.
f^(N)(t)=O(B^N
If there, by chance, happens to be a feature in the noise that
catches the ear and creates a sort of (possibly first subconscious)
memory, then the choo-choo effect will be more audible as that feature
can be more easily recognized again, reinforcing the memory. I
generated 10 seconds of Gaussian
On Fri, Mar 14, 2014 at 4:46 PM, Richard Dobson
richarddob...@blueyonder.co.uk wrote:
On 14/03/2014 14:27, Olli Niemitalo wrote:
http://yehar.com/Fast%20Track%20Ultra%2048%20kHz%20output-input%20ir.jpg
It looks more like a minimum-phase lowpass filter. The marker at
sample #29 indicates what
There are no exactly bandlimited functions that have non-zero-length
constant-valued intervals in the time domain. So any transients in an
exactly bandlimited time domain signal will have to be premeditated. You
can't keep a signal exactly bandlimited if you make a causal decision at
some point in
On Tue, Apr 10, 2012 at 12:06 PM, Julian Schmidt
julian_schm...@chipmusik.de wrote:
okay, i used exactly RBJs code with 1024 samples tablesize and I get a -60
dB spectral distortion floor.
That's again what no interpolation would give, so probably there is
a bug in the interpolation code,
On Tue, Apr 10, 2012 at 12:06 PM, Julian Schmidt
julian_schm...@chipmusik.de wrote:
okay, i used exactly RBJs code with 1024 samples tablesize and I get a -60
dB spectral distortion floor.
That's again exactly what no interpolation gives, so probably there is
a bug in the interpolation code,
On Tue, Apr 10, 2012 at 6:54 PM, Nigel Redmon earle...@earlevel.com wrote:
Clicks, especially 2-3 seconds apart doesn't describe aliasing
Here are clicks created by aliasing for you to listen (loudness warning!):
If you don't interpolate the wavetable then you may be simply getting
some aliasing problems. Every Nth cycle of the generated waveform will
be one sample shorter than the others, and that will sound like a
click. Try setting your phaseInc to an integer value. Also, somewhere
along the analog
On Mon, Apr 9, 2012 at 7:17 PM, Julian Schmidt
julian_schm...@chipmusik.de wrote:
setting the phase increment to an integer value solves the problem.
[...]
adding linear or cubic interpolation makes it a little better, but the
pulsing is still very audible.
[...]
The nearer i get to an
On Mon, Apr 9, 2012 at 11:32 PM, Julian Schmidt
julian_schm...@chipmusik.de wrote:
I really think it is an aliasing problem.
but not due to the wrong wavetable content, but due to a cheap audio codec
with poor filters.
[...]
even when i output a single 440hz sine i get harmonics starting at
On Tue, Apr 10, 2012 at 12:25 AM, Julian Schmidt
julian_schm...@chipmusik.de wrote:
Am 09.04.2012 23:22, schrieb Olli Niemitalo:
On Mon, Apr 9, 2012 at 11:32 PM, Julian Schmidt
julian_schm...@chipmusik.de wrote:
I really think it is an aliasing problem.
but not due to the wrong wavetable
On Wed, Feb 29, 2012 at 1:37 AM, Andrew Jerrim andrew.jer...@gmail.com wrote:
[On bytebeat:] Oooh, Olli - that's fantastic! Wouldn't that make a great
little phone app :)
There's Glitch Machine for iPhone/iPad, does much the same but in
reverse Polish notation.
On Fri, Feb 10, 2012 at 8:48 AM, Ross Bencina
rossb-li...@audiomulch.com wrote:
On 9/02/2012 11:02 AM, Jerry wrote:
(Good grief, people.) You want the *very famous* Bauer's Law of Sines:
...
Sin theta_I (S_l - S_r)
--- = ---
Sin theta_A (S_l + S_r)
Solving for S_l^2 +
Knowing that you're panning chorus voices to be summed with the input
signal gives something to work on.
Let's say there's just one chorus voice and someone sets up the
delays, volume and whatnot so that it is actually identical to the
input signal. Now, it would be unreasonable if, compared to
On Tue, Feb 7, 2012 at 12:20 PM, Ross Bencina
rossb-li...@audiomulch.com wrote:
Hi Everyone,
Does anyone know if there's a standard way to calculate pan laws for
stereo-wide panning ?
By stereo-wide I mean panning something beyond the speakers by using
180-degree shifted signal in the
No, it was my doing that the paragraphs had the synth name as their
first word(s). We don't have the or a/an in the Finnish language,
so I'm not always sure if they are needed, like in front of names (of
synthesizers) here. But I'm going to claim that most of that text
looked even worse before.
That won't be a problem if you measure the correlation locally, but
how exactly? Certainly anything outside the cross-fade region should
be excluded from the measurement. And inside it matters most wherever
the mixing ratio is close to 50-50, as in that cases phase difference
of the two signals
On Thu, Jul 14, 2011 at 9:22 PM, robert bristow-johnson
r...@audioimagination.com wrote:
g(t) = 1/sqrt( (1+r)/2 + 2*(1-r)*(p(t))^2 )
might this result match what you have?
Yes! I only derived the formula for the linear ramp, p(t) = t/2,
because one can get the other shapes by warping
On Sat, Jul 9, 2011 at 10:53 PM, robert bristow-johnson
r...@audioimagination.com wrote:
On Dec 7, 2010, at 5:27 AM, Olli Niemitalo wrote:
[I] chose that the ratio a(t)/a(-t) [...] should be preserved
by preserved, do you mean constant over all t?
Constant over all r.
what
Find the roots, pair the complex conjugate roots and distribute the
pairs and single real roots evenly (how exactly?) in the two filters.
Matlab at least has facilities finding roots of large polynomials.
-olli
On Wed, Jan 19, 2011 at 4:56 PM, Uli Brueggemann
uli.brueggem...@gmail.com wrote:
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