Re: [music-dsp] FIR blog post & interactive demo

e I actually worked on FFT filtering (via > phase vocoder) before learning FIR/IIR filters ... ? > > if anyone's interested in that source code it's here: > https://www.github.com/kardashevian > > On Wed, Jan 15, 2020 at 11:20 PM Alan Wolfe wrote: > >> probably pretty ba

[music-dsp] FIR blog post & interactive demo

probably pretty basic stuff for most people here but wanted to share a writeup and demo i made about FIRs. Post: https://blog.demofox.org/2020/01/14/fir-audio-data-filters/ Demo: http://demofox.org/DSPFIR/FIR.html Some simple ~175 lines of code C++:

Re: [music-dsp] high & low pass correlated dither noise question

dither algorithm without > multiplies, I think the way to extend this would be to include bit shifts > in the summation. Then you can get some reasonable spectral shapes. The > simple summation approach is too constrained for orders>1. > > Ethan > > On Thu, Jun 27, 2019 at 7:43 AM

[music-dsp] high & low pass correlated dither noise question

I read a pretty cool article the other day: https://www.digido.com/ufaqs/dither-noise-probability-density-explained/ It says that if you have two dice (A and B) that you can roll both dice and then... 1) Re-roll die A and sum A and B 2) Re-roll die B and sum A and B 3) Re-roll die A and sum A and

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

I wrote something in time domain using granular synthesis that doesn't sound too awful to me. There's explanation and samples on the page, as well as source code. https://blog.demofox.org/2018/03/05/granular-audio-synthesis/ On Thu, May 17, 2018 at 1:24 PM, Matt Ingalls wrote:

Re: [music-dsp] granular synth write up / samples / c++

glitch free. it can sound *very* good and companies like > Eventide have been doing something like that since the early-to-mid 80s. > (ever since the H949.) and i imagine any modern DAW does this (and some > might do frequency-domain pitch-shifting and/or time-scaling using > something

[music-dsp] granular synth write up / samples / c++

Hey Guys, Figured I'd share this here. An explanation of basic granular synth stuff, and some simple standalone C++ i wrote that implements it. https://blog.demofox.org/2018/03/05/granular-audio-synthesis/ Kind of amazed at how well it works (: Thanks for the answer to my question BTW Jeff.

[music-dsp] Is this granular synthesis?

Someone was explaining some algorithms to me that I thought were interesting. I was curious, is this granular synthesis? It seems to be but after i read this link https://granularsynthesis.com/guide.php. I'm unsure if it is, or is just close... --Adjust Audio Length Without Affecting

Re: [music-dsp] Reverb, magic numbers and random generators

I was just about to suggest that maybe something like a low discrepancy sequence could be interesting to explore - such as the golden ratio (which strongly relates to fib of course!). On Mon, Oct 16, 2017 at 10:22 AM, Andy Farnell wrote: > > Bit late to the thread,

Re: [music-dsp] Sampling theory "best" explanation

This is neat, thanks for sharing Nigel On Aug 25, 2017 6:22 PM, "Nigel Redmon" wrote: > Well, it’s quiet here, why not… > > Please check out my new series on sampling theory, and feel free to > comment here or there. The goal was to be brief, but thorough, and > avoid

Re: [music-dsp] Recognizing Frequency Components

ion. > > >> On 26/01/17 19:28, Bjorn Roche wrote: >> >> On Thu, Jan 26, 2017 at 2:09 PM, Alan Wolfe <alan.wo...@gmail.com> wrote: >> >>> It's some HTML filtering happening somewhere between (or including) his >>> machine and yours. >>> >>

Re: [music-dsp] Recognizing Frequency Components

It's some HTML filtering happening somewhere between (or including) his machine and yours. The less than of the for loop is being seen as the start of an HTML tag, or just possibly part of the start of an HTML tag and being stripped away. A common problem when providing code snippets on the web

Re: [music-dsp] Faster Fourier transform from 2012?

Thanks for the info, very interesting! (: On Sun, Aug 21, 2016 at 8:34 PM, Ross Bencina wrote: > [Sorry about my previous truncated message, Thuderbird is buggy.] > > I wonder what the practical musical applications of sFFT are, and whether > any work has been

[music-dsp] Faster Fourier transform from 2012?

This article has been getting shared and reshared by some graphics professionals / researchers I know on twitter. The article itself and arxiv paper are from 2012 though, which makes me wonder why we haven't heard more about this? Does anyone know if this is real?

[music-dsp] Are kalman filters used often in music or audio DSP?

I've read about kalman filters being used in dsp for things like flight controls. I was wondering though, do they have much use in audio and/or music applications? Thanks!! ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu

Re: [music-dsp] Trouble Implementing Huggins Binaural Pitch

e >> magnitude responses after FFT results in quite big deviations, even > 10 dB. >> So it seems that the allpass delays are not really allpasses without >> influencing the magnitude response. >> >> >> 2016-06-26 3:14 GMT+02:00 Alan Wolfe <alan.wo...@gmail.c

Re: [music-dsp] Trouble Implementing Huggins Binaural Pitch

- are you using an allpass filter to delay specific frequencies? - Jon On Jun 25, 2016, at 4:15 PM, Alan Wolfe <alan.wo...@gmail.com> wrote: Hey Guys, I'm trying to make an implementation of the Huggins Binaural Pitch illusion, which is where if you play whitenoise into each ear, but offs

Re: [music-dsp] Trouble Implementing Huggins Binaural Pitch

filter to make the sound less harsh. > > Also I did not hear the tone until I noticed the melody. Try playing a > simple melody or scale. > > Phil Burk > > > On Sat, Jun 25, 2016 at 2:08 PM, Alan Wolfe <alan.wo...@gmail.com> wrote: > >> Hey Guys, >>

[music-dsp] Trouble Implementing Huggins Binaural Pitch

Hey Guys, I'm trying to make an implementation of the Huggins Binaural Pitch illusion, which is where if you play whitenoise into each ear, but offset one ear by a period T that it will create the illusion of a tone of 1/T. Unfortunately when I try this, I don't hear any tone. I've found a

Re: [music-dsp] up to 11

In case you don't get any other responses, in Andy Farnell's book "Designing Sound" there is a section on psycho-acoustics that I'm reading right now that I think may be able to answer this question. I don't understand enough of it to answer it for you, but it is talking in great detail about

Re: [music-dsp] a family of simple polynomial windows and waveforms - DINisnoise

speaking of Bezier, the graphs shown earlier look a lot like gain ( http://blog.demofox.org/2012/09/24/bias-and-gain-are-your-friend/) and also SmoothStep which is y=3x^2+2x^3 Interestingly (to me anyways, before i learned more math) smoothstep is equivelant to a cubic bezier curve where the

Re: [music-dsp] Will Pirkle's "Designing Software Synthesizer Plug-Ins in C++"

Agreed, i like this book a lot and I used the information within to write a custom compressor and limiter for a PC game. Really great info. while on the topic of good books, I want to add two more that I've found very useful. Andy Farnell's "Designing Sound". It talks about the physics, math,

Re: [music-dsp] looking for tutorials

You bet! And apologies if i came off too harsh on your ideas. Passion is a good thing, and if you want to code all this stuff in assembly you'd get a lot of good experience working in both assembly and dsp stuff (: On Mon, Jun 13, 2016 at 9:17 AM, ty armour wrote: > Cool,

Re: [music-dsp] looking for tutorials

It would be ridiculous to code it all in assembly. The performance critical parts could be written in assembly, but only after profiling and finding that micro optimization would help. Assembly code is hard to write, hard to maintain, not portable, and you don't need it in situations where

Re: [music-dsp] R: Anyone using unums?

topic before the obvious problem comes >>> up. >>> >>> -Ethan >>> >>> >>> >>> On Fri, Apr 15, 2016 at 4:38 AM, Marco Lo Monaco >>> <marco.lomon...@teletu.it> wrote: >>>> >>>> I read his slides. Great

Re: [music-dsp] R: Anyone using unums?

ious problem comes >>> up. >>> >>> -Ethan >>> >>> >>> >>> On Fri, Apr 15, 2016 at 4:38 AM, Marco Lo Monaco < >>> marco.lomon...@teletu.it> wrote: >>> >>>> I read his slides. Great ideas but the best part is

Re: [music-dsp] list archives not updating?

listinfo/music-dsp > > > best, > douglas > > > > On Thu, Apr 14, 2016 at 5:29 PM, Alan Wolfe <alan.wo...@gmail.com> wrote: >> >> It looks like it stopped archiving messages last july: >> >> http://music.columbia.edu/pipermail/music-dsp/ >>

[music-dsp] Anyone using unums?

Apologies if this is a double post. I believe my last email was in HTML format so was likely rejected. I checked the list archives but they seem to have stopped updating as of last year, so posting again in plain text mode! I came across unums a couple weeks back, which seem to be a plausible

[music-dsp] list archives not updating?

It looks like it stopped archiving messages last july: http://music.columbia.edu/pipermail/music-dsp/ ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] Anyone using Chebyshev polynomials to approximate trigonometric functions in FPGA DSP

Chebyshev is indeed a decent way to approximate trig from what I've read. ( http://www.embeddedrelated.com/showarticle/152.php) Did you know that rational quadratic Bezier curves can exactly represent conic sections, and thus give you exact trig values? You essentially divide one quadratic

Re: [music-dsp] Inlined functions

Here's the standard response that you are likely to get a lot more of: You should profile before optimizing. If you are having a performance problem (including just wanting it to run faster in general), you should find out where the time is going and address the biggest time sinks specifically.

Re: [music-dsp] Inlined functions

lined methods. Usually it will be the render > callback, to reduce the function calls in that process. > > In some wave table oscillator code I have been looking at the main process > method that gets the next sample is inlined. > > Those were the reasons I was asking the questions > &g

Re: [music-dsp] sinc interp, higher orders

As far as the artifacts, it sounds like the information you are lacking is knowledge of bandlimiting and the nyquist frequency (: Check these out, I think they will help you, especially the second one, but the first one might have some info for you as well!

[music-dsp] what is GL_TEXTURE_2D_MULTISAMPLE??

-- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] recursive SIMD?

Right, with SIMD you buy in bulk so naive implementations of work where each sample needs processing before the next sample is problematic. Intuitively it seems like if you get creative, work out some math, and do some overlapping SIMD math, you might be able to do a factory line type of setup,

[music-dsp] Uses of Fourier Synthesis?

Hey Guys, I was wondering, does anyone know of any practical or interesting uses cases of Fourier synthesis for audio? I can already make bandlimited square, saw and triangle waves but was hoping for something like guitar strings or voice, or something along those lines. Someone shared

Re: [music-dsp] oversampled Fourier Transform

and of course, great discoveries often come from where people least expect them, and often trodden ground where others have walked before without noticing something major. Nothing wrong w/ fresh looks at old things, even if all it amounts to is someone getting a deeper understanding of things

Re: [music-dsp] R: Glitch/Alias free modulated delay

One thing to watch out for is to make sure you are not looking backwards AND forwards in time, but only looking BACK in time. When you say you have an LFO going from -1 to 1 that makes me think you might be going FORWARD in the buffer as well as backwards, which would definitely cause audible

Re: [music-dsp] Glitch/Alias free modulated delay

In case it helps, it isn't the delay buffer size that you need to modify, but rather just your read index into that delay buffer. If you have a flange for instance that can go from 0 to 500ms in the past, and is controlled by a sine wave, you always have a 500ms buffer that you put your output

Re: [music-dsp] Approximating convolution reverb with multitap?

not sure, if this can be used to identify tap positions, but my intuition tells me, it’d be starting point. Best, Steffan On 18.03.2015|KW12, at 18:11, Alan Wolfe alan.wo...@gmail.com wrote: Hey Guys, Let's say you have an impulse response recording of your favorite reverb location

[music-dsp] Approximating convolution reverb with multitap?

Hey Guys, Let's say you have an impulse response recording of your favorite reverb location. Are there any known algorithms for taking that impulse response and convert it to N taps for use in a multitap reverb implementation? I was trying to think about this and one hand it seems like maybe

Re: [music-dsp] Musical pitch detection by counting bitflips

Do you have a write up of this anywhere? I'd love to read more and have a place to point people to for more info. Also it would be neat to see how you extend this to higher dimensions, and also your log2 calculation is quite intriguing (: On Wed, Feb 4, 2015 at 7:52 AM, Peter S

Re: [music-dsp] magic formulae

You might check this out. An interesting tune made on Shadertoy.com where the audio is made with glsl https://www.shadertoy.com/view/ldfSW2 On Nov 27, 2014 8:28 AM, Michael Gogins michael.gog...@gmail.com wrote: I've experimented with this using LuaJIT, which has bitwise operations. I used a

Re: [music-dsp] entropy

For some reason, All I'm seeing are your emails Peter. not sure who you are chatting to or what they are saying in response :P On Wed, Oct 15, 2014 at 2:18 PM, Peter S peter.schoffhau...@gmail.com wrote: Academic person: There is no way you could do it! Impossibru!! Practical person: Hmm...

Re: [music-dsp] ANN: CDP is now a social enterprise

That's awesome! On Mon, Nov 4, 2013 at 2:58 PM, Richard Dobson richarddob...@blueyonder.co.uk wrote: [with apologies for any multiple posts] This is to announce that the Composers Desktop Project is now a UK social enterprise - a non profit-making limited company with (in the required legal

Re: [music-dsp] Functional Programming C code generation

fwiw, i have a DAW I work on, and on my todo list is the ability to export your creations to C++. One option would do generic C++ so you could drop it into whatever program you wanted (like, an fmod callback, or custom code etc). The other option to be generating code for a VST plugin. Just

Re: [music-dsp] basic trouble with signed and unsigned types

Just noticed in the archives that my response never went to the mailing list. People have covered most of it already but here ya go anyways... (: Well, you could certainly make a set of functions for the basic lego pieces of bit twiddling, and overload them for each type you want to support.

Re: [music-dsp] Efficiency of clear/copy/offset buffers

I'm sure it varies from hardware to hardware too, so always good to know your options On Thu, Mar 14, 2013 at 12:02 PM, jpff j...@cs.bath.ac.uk wrote: Ross == Ross Bencina rossb-li...@audiomulch.com writes: Ross I am suspicious about whether the mask is fast than the conditional for Ross a

Re: [music-dsp] Efficiency of clear/copy/offset buffers

RBJ's response would fit into that category I think Sampo (: On Thu, Mar 14, 2013 at 1:27 PM, Sampo Syreeni de...@iki.fi wrote: On 2013-03-14, jpff wrote: I did the comparison for Csound a few months ago. The loss in using modulus over mask was more than I could contemplate my users

Re: [music-dsp] Efficiency of clear/copy/offset buffers

interesting idea about rounding up and letting multiple buffers using the memory. Very nice. I just wanted to add on the front of enforcing powers of 2 sizes, the way you have it where you pass in an integer and it understand that as a power of 2 is nice but of course a little less intuitive to

Re: [music-dsp] Efficiency of clear/copy/offset buffers

Hey while we are on the topic of efficiency and the OP not knowing that division was slower... Often times in DSP you'll use circular buffers (like for delay buffers for instance). Those are often implemented by having an array, and an index into the array for where the next sample should go.

Re: [music-dsp] Efficiency of clear/copy/offset buffers

Quick 2 cents of my own to re-emphasize a point that Ross made - profile to find out which is fastest if you aren't sure (although it's good to ask too in case different systems have different oddities you don't know about) Also, if in the future you have performance issues, profile before acting

Re: [music-dsp] Thesis topic on procedural-audio in video games?

Howdy! I think kkrieger, the 96KB first person shooter uses procedural audio: http://www.youtube.com/watch?v=KgNfqYf_C_Q i work in games myself and was recently talking to an audio engineer (the non programming type of engineer) who has a couple decades of experience about procedural sound

Re: [music-dsp] Sound effects and Auditory illusions

I think it would be neat to show how wildly different wave forms can sound the same (like... you can make a square wave with sine or cosine. Using one looks like a square, using the other doesn't, but they both sound the same). Also it would probably be neat to budding audio folk / audio

Re: [music-dsp] Starting From The Ground Up

Heya, I'm a game programmer by trade who dabbles in DSP and audio programming. I have a handful of books on the subject but recently was turned onto one that was aimed at programmers. Reading it has been really enlightening and seeing things in code which previously i had only seen as complex

Re: [music-dsp] Calculating the gains for an XY-pad mixer

What you are trying to calculate is called barycentric coordinates, you might give them a google (: As far as them all adding to one (which barycentric coordinates do), I'm not sure if that's appropriate or not, because you have to remember that volume is linear, but the perception of that linear

Re: [music-dsp] Calculating the gains for an XY-pad mixer

of a simplex(a triangle, tetrahedron, etc.) - from wikipedia On Thu, Jan 17, 2013 at 9:22 PM, Ross Bencina rossb-li...@audiomulch.com wrote: On 18/01/2013 4:06 PM, Alan Wolfe wrote: What you are trying to calculate is called barycentric coordinates, Actually I don't think so. Barycentric

[music-dsp] Fwd: Programming ARC on a compressor

Hey Guys, I have a compressor that works by having an envelope follower (full rectifier with attack and release settings) following the uncompressed input, then applying the compression ratio to my input where the envelope follower samples are above the threshold (working in db for that part). I

Re: [music-dsp] Fwd: Programming ARC on a compressor

That sounds like a really interesting and not too perf intensive implementation. I can dig around in the archives, you already did all the hard work of creating it and sharing it hehe. Thanks a ton James! On Thu, Jan 10, 2013 at 4:16 PM, James C Chandler Jr jchan...@bellsouth.net wrote: On

Re: [music-dsp] Getting Started with PortAudio

Heya, You might want to ask the port audio mailing list, you're more likely to find better answers there :P http://www.portaudio.com/contacts.html On Thu, May 26, 2011 at 1:46 PM, resea...@ottomaneng.com resea...@ottomaneng.com wrote: Hello, I am trying to get started using PortAudio on

Re: [music-dsp] Sinewave generation - strange spectrum

Might want to ask this one on the PA list (: On Wed, Apr 27, 2011 at 2:47 PM, eu...@lavabit.com wrote: Hello, Today I tried compiling on Windows with MinGW MSYS, and everything works, the spectrum is clean at SR=44.1 kHz and even the LUT version is acceptable. Probably on linux I was

Re: [music-dsp] Sinewave generation - strange spectrum

just stabbing in the dark in case nobody else gives a more useful response but... #1 - what is the format of your output? If it's low in bitcount that could make the signal more dirty i believe (less resolution to make a more perfect sine wave) #2 - have you tried calculating via doubles? #3 -

Re: [music-dsp] Sinewave generation - strange spectrum

at 1:14 PM, Alan Wolfe alan.wo...@gmail.com wrote: just stabbing in the dark in case nobody else gives a more useful response but... #1 - what is the format of your output?  If it's low in bitcount that could make the signal more dirty i believe (less resolution to make a more perfect sine wave

Re: [music-dsp] Sinewave generation - strange spectrum

: Alan Wolfe Sent: Tuesday, April 26, 2011 9:14 PM To: A discussion list for music-related DSP Subject: Re: [music-dsp] Sinewave generation - strange spectrum just stabbing in the dark in case nobody else gives a more useful response but... #1 - what is the format of your output?  If it's low

Re: [music-dsp] Floating Point Division

i don't know that chip but have you thought about re-arranging your math? with limited precision, order of operations can matter. This is a bigger problem with fixed point (and integers) than floating point but it can still be a problem even in floating point. whats the specific error message?

Re: [music-dsp] resonance

Wolfe alan.wo...@gmail.com wrote: Thanks a bunch Ross (: On Sat, Jan 1, 2011 at 1:54 AM, Ross Bencina rossb-li...@audiomulch.com wrote: Alan Wolfe wrote: I have a future retro revolution (303 clone) and one of the knobs it has is resonance. Does anyone know what resonance

Re: [music-dsp] Merry Christmas - and some music

Very nice song (: Also thanks for bringing up the impulse train topic, i hadn't heard of that before. Im reading the link you sent but am i right in thinking that an impulse train is just a really narrow rectangle wave? ::continues to read:: On Fri, Dec 24, 2010 at 10:19 PM, Thor Harald

Re: [music-dsp] achieving chiptunes sounds

i tried the variable rectangle wave and that sounds A TON more like old early nes style game music. Even without sound degradation it really sounds a lot like a chiptune im going to try the quick arpeggios next (: On Tue, Dec 21, 2010 at 2:49 AM, Laurent de Soras laurent.de.so...@free.fr wrote:

Re: [music-dsp] Gate on/off clicks in analogue synths?

Someone else will surely chime in but i asked a similar question a couple months back and i remember one person suggested that internally, such devices probably have a capacitor (if my memory recalls correctly) that acts as a basic envelope by ramping up and down the volume at the begin and end.

Re: [music-dsp] Gate on/off clicks in analogue synths?

PM, Alan Wolfe alan.wo...@gmail.com wrote: Someone else will surely chime in but i asked a similar question a couple months back and i remember one person suggested that internally, such devices probably have a capacitor (if my memory recalls correctly) that acts as a basic envelope by ramping

Re: [music-dsp] Algorithms for finding seamless loops in audio

Agreed here (: in 2d graphics and skeletal animation, making tileable 2d art and seamless blends are basically the same problems. in both areas they MAKE the things seamless instead of trying to find how they could be seamless. in 2d graphics this comes up via texturing (probably obvious), and

Re: [music-dsp] Is beating the same thing as flanging?

i fear to post a question being the OP of this huge 100+ message thread but... it was mentioned here and in a previous email that for digital flangers you want to interpolate between samples for best results. Would you want to do this for all sampling digital effects such as delay and reverb