My 2 cents on 1: yes we always pre-compute filter coefficients, especially
since they often involve trig functions which are expensive. I've rarely
seen them actually stored as a filter, but if your application is to have
many filters operating in parallel it's a good idea, but requires you to
use
I'm definitely not the most mathy person on the list, but I think there's
something about the complex exponentials, real transforms and the 2-point
case. For all real DFTs you should get a real-valued sample at DC and
Nyquist, which indeed you do get with your matrix. However, there should be
some
Because the o stands for overlap.
https://ieeexplore.ieee.org/document/319366. I'm not specifically familiar
with wsola, but if it's like the other overlap-add techniques then starting
with some overlap and modifying that relationship is the fundamental way to
change the sound.
Wikipedia has a dec
rily
> short delays and two feedback paths.
>
>
>
> --
> r b-j r...@audioimagination.com
>
> "Imagination is more important than knowledge."
>
>
>
>
>
> Original message
> From: Stefan Sullivan
> Date:
Yes. The term helmholz resonator should be a hint ;) Basically when a
sounds gets added to itself after a delay you end up adding energy to the
frequency that corresponds to that delay amount. For very long echos we
don't hear it as a resonance, but for shorter delays it will boost higher
and highe
Edits:
Paragraph 1
...assuming you're going to end up *masking* the aliased components
Masking ≠ making
Paragraph 2
...or otherwise *generic* samples
Generic ≠ genetic
Stefan
On Sun, Jun 3, 2018, 19:41 Stefan Sullivan
wrote:
> You can still take a heuristic approach. You probably h
You can still take a heuristic approach. You probably have some idea of the
max modulation rate, right? I would just indiscriminately apply a low-pass
filter at Nyquist/fastest rate of change (forgive my fast and loose math
here). You can also relax that a little bit by taking a perceptual criteria
Surprisingly, googling for mq synthesis produces fewer results than I
thought. It's not very recent but my understanding is that it was the
relatively modern approach to a phase vocoder. I didn't find any library
implementing it but I found a couple older papers about it.
https://www.ll.mit.edu/pu
Can you explain your notation a little bit? Is x[t] the sample index into
your signal? And t is time in samples?
I might formulate it as a Delta of indicies where a Delta of 1 is a normal
playback speed and you have some exponential rate. Would something like
this work?
delta *= rate
t += delta
y
Forgive me if you said this already, but did you try negative feedback
values? I wonder what that does to the aesthetics of the reverb.
Stefan
On Oct 1, 2017 16:24, "gm" wrote:
> and here's the impulse response, large 4APs Early- > 3AP Loop
>
> its pretty smooth without tweaking anything manua
Sometimes the simplest approach is the best approach. Sounds like a good
reverb paper to me. Some user evaluation and references to standard papers
and 😁
On Sep 29, 2017 8:51 AM, "gm" wrote:
> It's a totally naive laymans approach
> I hope the formatting stays in place.
>
> The feedback delay in
so there might be a phase
offset between the recorded
and the reproduced sound.
Ah, I think I might be understanding your question more intuitively. Is
your question about positive voltages from microphones being represented as
one direction of displacement, whereas the positive voltages from spe
Acoustic transducers (aka microphones and speakers) would be a good keyword
for finding more technical information. They convert pressure differentials
(not pressure per se) to +/- voltage. The pressure is change relative to a
baseline, which is usually right around 1 atmosphere (although it doesn'
It would be difficult to control for things like the time-varying behavior
of the audio processing on the phone, as well as the nonlinearities of the
same audio processing on the phone, not to mention the environmental noise
in the store, which would confound both of these behaviors. The audio
engi
Hey all,
Smule is hiring 2 audio/DSP-related positions (and several others).
We are looking to hire one Audio/DSP systems engineer:
https://www.smule.com/jobs?gh_jid=597566
and one audio effects engineer:
https://www.smule.com/jobs?gh_jid=660357
First and foremost, we hop you can help us make o
https://docs.scipy.org/doc/scipy/reference/generated/scipy.interpolate.lagrange.html
https://docs.scipy.org/doc/scipy/reference/interpolate.html
I guess I should have thought of "interpolators" when I suggested
interpolation :D
Stefan
On Mar 6, 2017 02:29, "Leonardo Gabrielli" wrote:
> Thank
Fractional sample delays are simply integer sample delays with
interpolators at the back of them. It's common to implement it as a
delay followed by an allpass filter. Take a look at Julius Smith's
book:
https://ccrma.stanford.edu/~jos/Interpolation/Simple_Interpolators_suitable_Real.html.
For th
A linear phase all-pass filter is a delay.
Stefan
On Dec 7, 2016 4:30 AM, "STEFFAN DIEDRICHSEN" wrote:
>
> On 07.12.2016|KW49, at 13:10, Uli Brueggemann
> wrote:
>
> Is there a solution to elegantly calculate the pulse response ap ? The
> calculation of p^-1 may be difficult or numerically uns
TL; DR
A high-pass filter? The first and second derivatives could be easily enough
described with first and second-order feedback filters, respectively, but
once you start fitting that stuff into DSP terminology, then you might as
well make a low-order high-pass filter that has the characteristics
FYI, numpy and scipy are python modules written in C that are
specifically designed to take advantage of machine memory allocation,
SIMD instructions, etc., and don't interface with python's memory
management (as much as possible. It is possible to create a numpy
ndarray from a python list).
http:/
I looked into this exact issue a little while ago. I found that my filters
sounded better/worse depending on the biquad topology. Basically if your
gaining your input going into states, then those states are more likely to
be very far off from where they should be when you change the parameters.
Bu
lol
On Fri, Aug 28, 2015, 13:54 Gunnar Eisenberg
wrote:
> Dear list,
>
> I'm trying to implement a 20,000 coefficients FIR filter but for some
> reason it is a bit slow on my system (Pentium III 500Mhz).
>
> Any suggestions on how to fix my problem?
>
> Have a nice weekend and carry on... :-)
>
Well that didn't take long
On Mon, Aug 24, 2015 at 2:08 PM Peter S
wrote:
> On 24/08/2015, Theo Verelst wrote:
> > I'm not going to confuse etiquette with thinking straight, it's clear if
> > people can be respected and have some things to learn or teach, or go on
> > about emotionally, that a
Perhaps the knowledge that you might risk exceeding your limit (which I'm
sure would not be pedantically enforced) would make you to consider for
yourself how much the given message is contributing to the discussion.
Thank you Douglas, for clarifying the etiquette and audience. It was needed
and I
It would be terrible if
the actual behavior changed because of a compiler change.
-stefan
On Wed, Feb 25, 2015 at 11:25 AM, Laurent de Soras
wrote:
> Stefan Sullivan wrote:
>>
>>
>> similar problems: the output is not bit-exact.
>
>
> Have you a simple project showing
Hey music-dsp folks,
I know that it is not exactly everybody's favorite compiler, in part
because of questions like the one I have right now. But suffice it to
say, there are situations in which I'm required to use Visual Studio.
A couple of months ago I was working on a project in which I attempt
I actually found by playing around with a particular biquad problem that
changing the topology of the filter had a greater impact on reducing
artifacts than proving bibo stability. In fact, any linearly interpolated
biquad coefficients between stable filters results in a stable filter,
including wi
Wow, thanks Paul for such a detailed answer. I've started to scratch the
surface with DSP coding, but I haven't even gotten as far as learning
assembly. However, I am a little afraid that I could spend a lot of time
becoming an expert in a system that doesn't provide me any skills in
another system
Hey music DSP folks,
I'm wondering if anybody knows much about using these open source compilers
to compile to various DSP architectures (e.g. SHARC, ARM, TI, etc). To be
honest I don't know so much about the compilers/toolchains for these
architectures (they are mostly proprietary compilers right
> On Mar 26, 2014, at 10:07 PM, Doug Houghton
wrote:
> > so is there a requirement for the signal to be periodic? or can any
series of numbers be cnsidered periodic if it is bandlimited, or infinit?
Periodic is the best word I can come up with.
> > --
>
> Well, no--you can decompose any portion o
>
> Marco
>
> -Messaggio originale-
> Da: music-dsp-boun...@music.columbia.edu
> [mailto:music-dsp-boun...@music.columbia.edu] Per conto di Stefan Sullivan
> Inviato: lunedì 3 marzo 2014 12:17
> A: A discussion list for music-related DSP
> Oggetto: Re: [music-
For matching just the magnitude response, MATLAB has a built-in function for it:
http://www.mathworks.com/help/signal/ref/yulewalk.html
And maybehaps some more parametric modelling techniques will be useful for you
http://www.mathworks.com/help/signal/ug/parametric-modeling.html
-Stefan
On Mon,
On Mon, Jan 13, 2014 at 2:24 PM, Thomas Strathmann wrote:
>
> On 13.01.14 09:46, Frank Sheeran wrote:
> > At this point, the #1 goal is to evaluate the language itself. Is a
> > functional, textual, programming language the best way to design a
> > patch? Better than Csound, better than visual e
There's nothing to say you can't write a non-linear IIR filter, though.
Especially if you're implementing the DSP yourself rather than some
pre-packaged things like biquads. The FIR representation looks something
like
y[n] = sqrt{ ( x[n]*x[n] + x[n-1]*x[n-1] + x[n-2]*x[n-2]) / 3 }
so I guess an
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