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

/* Update phase, rollover at 1.0 */ data-phase[0] += (data-frequency[0] / SAMPLE_RATE); if(data-phase[0] 1.0f) data-phase[0] -= 2.0f; data-phase[1] += (data-frequency[1] / SAMPLE_RATE); if(data-phase[1] 1.0f) data-phase[1] -= 2.0f; You haven't shown us the declarations/data types for

Re: [music-dsp] FM Synthesis

The sad news is that FM with feedback cannot be done the naïve way. You need to account for aliasing. Someone upthread suggested adding noise instend of feedback, this is probably a good idea. But it will not make your FM synthesis engine sound like the real thing. -- dupswapdrop -- the music-dsp

Re: [music-dsp] stereo-wide pan law?

Lol! Actually, when reading the OP the first thought that went through my head was Hilbert transformer. So I scanned the thread, and sure enough... It would seem that once you have an analytic signal, all you need to do is to apply a simple complex rotation with a phase offset for the second

Re: [music-dsp] stereo-wide pan law?

On Wed, Feb 8, 2012 at 5:25 PM, Theo Verelst theo...@tiscali.nl wrote: left = cos 0 * Re - sin 0 * Im = Re right = sin theta * Re + cos theta * Im It sometimes amazes me where people learn all this ..., though I partially know the answer. How can you take the Real and imaginary part of a

Re: [music-dsp] stereo-wide pan law?

On Tue, Feb 14, 2012 at 12:11 AM, Sampo Syreeni de...@iki.fi wrote: On 2012-02-09, Emanuel Landeholm wrote: I was just reacting to the oxymoronic juxtaposition of two blatant opposites. And no.. I wasn't thinking in the Fourier domain. It's a complex rotation in time domain analytic signal

Re: [music-dsp] stereo-wide pan law?

Mr Syreeni, this is like the worst cliff hanger for me. Please sort this out asap! On Tue, Feb 14, 2012 at 2:24 AM, Emanuel Landeholm emanuel.landeh...@gmail.com wrote: On Tue, Feb 14, 2012 at 12:11 AM, Sampo Syreeni de...@iki.fi wrote: On 2012-02-09, Emanuel Landeholm wrote: I was just

Re: [music-dsp] a little about myself

Adam, I am also curious as to how you interface with a development environment. Moreover, what are you working on? Also, wasn't there a blind contributor around a few years ago? Your name does not ring a bell for me, so I can only guess that you are number two (minimum)! Also, isn't this a

Re: [music-dsp] a little about myself

Well. I need to start using csound. To actually do things in the real world instead of just solving idle mind puzzles. On Tue, Feb 21, 2012 at 10:02 PM, Victor victor.lazzar...@nuim.ie wrote: i have been running csound in realtime since about 1998, which makes it what? about fourteen years,

Re: [music-dsp] a little about myself

For example, the strings are made from a few sawtooth waves starting at random phases, then having pitch and amplitude randomly modulated. The random modulation is absolutely essential for avoiding that harsh, metallic sound, but I suspect that it also has the side effect of reducing the

Re: [music-dsp] google's non-sine

NURBS should do the trick. On Thu, Feb 23, 2012 at 3:53 PM, Didier Dambrin di...@skynet.be wrote: There's also the fact that it's not easy to draw a sinewave in existing tools out there. Those who have drawn GUIs here and had to show waveforms know what I mean, I remember I've ended up with

Re: [music-dsp] Boulez

While raw speed does reduce the risk of missing deadlines, you need an infinitely fast computer to guarantee hard realtime performance with code that isn't designed for it. Also, theoretically, not even that helps, unless you also have a realtime OS. And then there's I/O, synchronization and

Re: [music-dsp] a little about myself

Yeah, no shit just hit the fan... When you least expect it... On Sun, Feb 26, 2012 at 2:26 AM, robert bristow-johnson r...@audioimagination.com wrote: On 2/20/12 10:28 AM, douglas repetto wrote: Hi Adam, Welcome to the list. It's slow right now, but no doubt it'll flare up again soon!

Re: [music-dsp] Boulez

It certainly helps when you can do interesting stuff in suboptimal ways, and still end up using only a few percent of one of your many CPU cores. :-) Actually, this is my routine for determining whether or not I'm living in the future: look up suboptimal in the dictionary. If it isn't there,

Re: [music-dsp] guitar physical model

       http://music.columbia.edu/~brad/music/mp3/Rough_Raga_Riffs.mp3 This. I just listened to it and it put me in a good mood! -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links

Re: [music-dsp] ANN: ChipSound 0.1.0 under zlib license

Nice one! Will definitely check out and thanks for sharing. I'm aiming to have something to share myself later this spring, so this is really good for my morale. On Tue, Mar 13, 2012 at 5:51 PM, David Olofson da...@olofson.net wrote:        ChipSound 0.1.0 under zlib license        

Re: [music-dsp] Mastering correction by FFT-based filtering followed by 1 octave or 1/10 octave equalizer

Continuing off topic... On correction; it's an interesting philosophical concept. I listen to lots of audio books, and the program material comes with all kinds of problems. Noise, too low volume due to spurious noise+peak-limiting, too much dynamics etc. My typical listening device is my

Re: [music-dsp] Mastering correction by FFT-based filtering followed by 1 octave or 1/10 octave equalizer

audible but neutrally rendered speech. cheers, On Thu, Mar 6, 2014 at 8:52 PM, Theo Verelst theo...@theover.org wrote: Emanuel Landeholm wrote: Continuing off topic... On correction; it's an interesting philosophical concept. I listen to lots of audio books, and the program material comes

Re: [music-dsp] The Uncertainties in Frequency Recognition

You have to be careful about presumptions of the kind in the main theory to begin with. *if* you sampled properly, *and* either it is given you have certain sinusoidal components, or your sample row and analysis length is long enough (probably seconds for high q audio), there's only one

Re: [music-dsp] The Uncertainties in Frequency Recognition

right at the edge of my dragon territory, Here be dragons? Lol. cheers, On Wed, Mar 12, 2014 at 2:36 AM, Sampo Syreeni de...@iki.fi wrote: On 2014-03-12, Emanuel Landeholm wrote: The intepolation filter only needs to be infinitely long if you need infinite precision. In practice, any

Re: [music-dsp] Best way to do sine hard sync?

FYI Emanuel Landeholm also had a cool method using windows to suppress aliasing. It sounded pretty good to my ear, though I never did any spectral measurements. It works with any slave oscillator waveform, including sine. I implemented it in PD with a Kaiser-Bessel window for an extra

Re: [music-dsp] Best way to do sine hard sync?

and slave rates can be set independently, but you have to watch out for DC when master frequency slave frequency. cheers, E On Fri, Mar 21, 2014 at 10:28 PM, David Lowenfels david.lowenf...@gmail.com wrote: On Mar 21, 2014, at 5:26 AM, Emanuel Landeholm emanuel.landeh...@gmail.com wrote: Also

Re: [music-dsp] Best way to do sine hard sync?

Frank Sheeran, From my reading of wikipedias page on phase distortion synthesis, my method is definitely related. The main differences are that I use two modulators (master oscillators), and a cos^2 window instead of a triangular wave form. I wouldn't be at all surprised if Casio CZ synthesis was

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

tl;dr version: The justification for DSP (equi-distant samples) is the Whittaker-Shannon interpolation formula, which follows from the Poisson summation formula plus some hand-waving about distributions (dirac delta theory). Am I right? On Fri, Mar 28, 2014 at 4:50 AM, Ethan Duni

Re: [music-dsp] Dither video and articles

Dither theory is way cool. The problem with quantization noise is that it's correlated to the signal. This is the reason it sounds so horrible. When you're doing 1 bit dsp, dither (and noise shaping) is an absolute requirement. When rendering to 8 bits you definitely benefit from dithering. 16

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

) (thus the distributional hand waving requirement). This is what I meant by PSF + hand waving. I think we're on the same page, basically. cheers, E On Fri, Mar 28, 2014 at 1:32 PM, robert bristow-johnson r...@audioimagination.com wrote: On 3/28/14 4:25 AM, Emanuel Landeholm wrote: tl;dr

Re: [music-dsp] Dither video and articles

First, it's meaningless to talk about bit depth alone I agree with the points you raise and I'd like to add that you can also trade bandwidth for bits. On Fri, Mar 28, 2014 at 8:31 PM, Sampo Syreeni de...@iki.fi wrote: On 2014-03-28, robert bristow-johnson wrote: 14 bits??? i seriously

Re: [music-dsp] Nyquista?Shannon sampling theorem

Possibly on topic: Some people like to apply insane compression with a lazy attack/release to their bass drums. Then they amplitude modulate the rest of the mix with that. They call it house music. On Fri, Mar 28, 2014 at 5:13 PM, Sampo Syreeni de...@iki.fi wrote: On 2014-03-28, Charles Z

Re: [music-dsp] a weird but salient, LTI-relevant question

Datapoint: I just tried repeating a ~1 sec brown noise clip in Audacity and I'm not sure if I get that choo choo effect. It sounds pretty continuous to me. However, I think this requires ABX testing in order to make sure. On Thu, May 8, 2014 at 8:14 PM, Nigel Redmon earle...@earlevel.com wrote:

Re: [music-dsp] On the theoretical foundations of BLEP, BLAMP etc

Drunk me just signing in to say: this thread is epic. The discussion here captures everything, from the basic dsp stuff to the esoteric. Pls continue! On Thu, Jul 3, 2014 at 6:09 PM, Nigel Redmon earle...@earlevel.com wrote: On Jul 3, 2014, at 1:36 AM, Vadim Zavalishin

Re: [music-dsp] Instant frequency recognition

Haven't really been following the thread but I wonder if the sinusoid model is really that good. Don't we actually want to match something like SUM(k,1,N) e^jwkt and might not harmonics help us from falling down to the noise floor? On Mon, Aug 4, 2014 at 10:25 AM, Vadim Zavalishin

Re: [music-dsp] Instant frequency recognition

SUM(k,1,N) a_k e^jwkt even On Wed, Aug 6, 2014 at 11:00 PM, Emanuel Landeholm emanuel.landeh...@gmail.com wrote: Haven't really been following the thread but I wonder if the sinusoid model is really that good. Don't we actually want to match something like SUM(k,1,N) e^jwkt and might

Re: [music-dsp] Instant frequency recognition

Sorry, meant to say SUM(k,1,N) a_k e^jwk(t+p_k) It would seem that phase should be important, especially if instaneous frequency is desired . On Wed, Aug 6, 2014 at 11:01 PM, Emanuel Landeholm emanuel.landeh...@gmail.com wrote: SUM(k,1,N) a_k e^jwkt even On Wed, Aug 6, 2014 at 11:00 PM

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

Glitch/Alias free non-LTI (exemplified by modulated delay line) is not going to happen. It's a pipe dream. What you should be looking for is tolerance (in dB or whatever). On Sat, Mar 21, 2015 at 3:47 PM, Theo Verelst theo...@theover.org wrote: Nuno Santos wrote: Hi, I’m trying to implement

Re: [music-dsp] minBLEP parameters: grain design and duration?

My advice is this: don't get too caught up in the theory! I tend to do that myself, so... Just implement something quick and dirty and *listen* to the results. Also, I really don't think the choice of window matters very much. You could probably get away with Hamming. And 150 samples at 44k1 is

Re: [music-dsp] Low cost DSPs

If you target modern GPU:s you will have a truly huge platform with massive computational power. Just a thought. On Thu, 16 Feb 2017 at 05:03, Pablo Riera wrote: > Hi, > > I am collecting information on how to accomplish DSP projects (mainly > synths, only output) with

Re: [music-dsp] Help with "Sound Retainer"/Sostenuto Effect

Simple OLA will produce warbles. I recommend a phase vocoder. ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] Help with "Sound Retainer"/Sostenuto Effect

Essentially, what you want is a "sustain" effect? ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] Bandlimited morphable waveform generation

> How do I detect discontinuities? It is easy to see when printed visually but I do not see how I can approach this with code. Do I need the ‘complete’ function at once and check or can I do it in runtime for each sample. I think so since you suggest that I can jump around within the function

Re: [music-dsp] Help with "Sound Retainer"/Sostenuto Effect

Fabian (and list), This looks reality interesting. Are you estimating PSD FIR-coeffs using that Burg algorithm? I have seen something similar that produces second order sections for IIR. I believe it's called CELP. /Emanuel ___ dupswapdrop: music-dsp

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

Sorry for being a slowpoke! Is this an efficient implementation of STFT (short time fourier transform)? On Thu, Jun 25, 2020 at 8:49 AM STEFFAN DIEDRICHSEN wrote: > I think, Robert had its morning coffee after his reply …. ;-) > > Steffan > > On 24.06.2020|KW26, at 23:03, Zhiguang Eric Zhang >