Re: [music-dsp] Practical filter programming questions

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

Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?

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

Re: [music-dsp] WSOLA

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

Re: [music-dsp] What is resonance?

delays and two feedback paths. > > > > -- > r b-j r...@audioimagination.com > > "Imagination is more important than knowledge." > > > > > > Original message > From: Stefan Sullivan > Date: 7/22/2018 2:20

Re: [music-dsp] What is resonance?

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

Re: [music-dsp] Antialias question

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 have s

Re: [music-dsp] Antialias question

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

Re: [music-dsp] Real-time pitch shifting?

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.

Re: [music-dsp] Reading a buffer at variable speed

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

Re: [music-dsp] Reverb, magic numbers and random generators #2 solution?

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

Re: [music-dsp] Reverb, magic numbers and random generators #2 the Go approach

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. >

Re: [music-dsp] PCM audio amplitudes represent pressure or displacement?

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

Re: [music-dsp] PCM audio amplitudes represent pressure or displacement?

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

Re: [music-dsp] basic in-store speaker evaluation

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

[music-dsp] Smule Is Hiring

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

Re: [music-dsp] Fractional delay filters in Python?

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"

Re: [music-dsp] Fractional delay filters in Python?

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

Re: [music-dsp] Allpass filter

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 >

Re: [music-dsp] BW limited peak computation?

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

Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state?

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.

Re: [music-dsp] [admin] list etiquette

Well that didn't take long On Mon, Aug 24, 2015 at 2:08 PM Peter S peter.schoffhau...@gmail.com wrote: On 24/08/2015, Theo Verelst theo...@theover.org 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

Re: [music-dsp] [admin] list etiquette

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

Re: [music-dsp] MSVC 2012/2013 upgrade with audio differences

because of a compiler change. -stefan On Wed, Feb 25, 2015 at 11:25 AM, Laurent de Soras laurent.de.so...@free.fr wrote: Stefan Sullivan wrote: similar problems: the output is not bit-exact. Have you a simple project showing the difference? I’m not sure why you need this, but there are a lot

[music-dsp] MSVC 2012/2013 upgrade with audio differences

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

[music-dsp] LLVM or GCC for DSP Architectures

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

Re: [music-dsp] Nyquist-Shannon sampling theorem

On Mar 26, 2014, at 10:07 PM, Doug Houghton doug_hough...@sympatico.ca 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

Re: [music-dsp] Iterative decomposition of an arbitrary frequency response by biquad IIR

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,

Re: [music-dsp] R: Iterative decomposition of an arbitrary frequency response by biquad IIR

: 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-dsp] Iterative decomposition of an arbitrary frequency response by biquad IIR

Re: [music-dsp] Moselle Alpha 0.1.3 Released

On Mon, Jan 13, 2014 at 2:24 PM, Thomas Strathmann tho...@pdp7.org 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