Re: [music-dsp] Transient shaping - differential envelope?
Hi Danijel, You might be interested in these papers: http://www.aes.org/e-lib/browse.cfm?elib=12912 Avendano, Carlos; Goodwin, Michael "Enhancement of Audio Signals Using Transient Detection and Modification” AES 117, 2004 http://c4dm.eecs.qmul.ac.uk/audioengineering/transientmodification/transient_shaping_report.pdf -David > On Jul 6, 2016, at 7:24 AM, Danijel Domazet > wrote: > > Hi music-dsp, > How does one implement an envelope adjustment algorithm that is triggered > only on transients, rather than on a loudness threashold which is used in > conventional compressors? Something similar to Voxengo's TransGainer or SPL's > Transient Designer. Transient Designer uses “differential envelope”. There is > some info in the tech-talk section of their manual > (spl.info/fileadmin/user_upload/anleitungen/english/TransientDesigner4_9842_OM_E.pdf). > Could someone elaborate on this concept of level-independent envelope > processing which doesn't need conventional “threshold” control? > > Thanks, > Danijel > > > ___ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] Will Pirkle's "Designing Software Synthesizer Plug-Ins in C++"
Hi, I just purchased Will Pirkle’s textbook "Designing Software Synthesizer Plug-Ins in C++” and wanted to give a huge thumbs up. It demystifies so many state-of-the-art things about virtual analog, including filters (delay-free loops!), band-limited oscillators, envelope generators, modulation matrices, etc. And also goes into heavy detail on the ins and outs of AU and VST (and his own platform RAFX). As a fledgling music-dsp coder, I really wish I’d had a practical manual such as this! I was frustrated in university that I could write algorithms and DSP code but didn’t know how to package it into a plugin, similarly to how I could design crazy hardware/software on paper and breadboards but didn’t know how to make a PCB or surface-mount solder to put it in a box. I also look forward to perusing his other book on Digital Audio effects. -David ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a family of simple polynomial windows and waveforms - DINisnoise
> On Jun 12, 2016, at 3:04 AM, Andy Farnell wrote: > > I did some experiments with Bezier after being hugely inspired by > the sounds Jagannathan Sampath got with his DIN synth. > (http://dinisnoise.org/) DIN is not just an additive synth? appears to be so looking at the prominent and low-res FFT display everywhere. > Jag told me that he had a cute method for matching the endpoints > of the segment (you can see in the code), and listening, sounds > seem to be alias free, but we never could arrive at a proof of > that. there’s no way those naive (linear) saw and pulse waveforms could be alias free. arriving at proof would be as easy as an FFT with more resolution? -David signature.asc Description: Message signed with OpenPGP using GPGMail ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] LFO Flanger.
Hi Darwin,
The LFO looks to be not implemented properly... appears like you are trying
to make a lookup-table, which would be an array of values. But you will still
need to interpolate between integer indexes in order to match up with your
float phasor value. Check out this article to see how to do it:
http://www.mochima.com/articles/LUT/LUT.html
You might also want to look at the STK library for nice C++ classes for
oscillators and delay lines: https://ccrma.stanford.edu/software/stk/
In the meantime, you can make a phasor and use the sine function:
float normalized_freq = lfo_freq/samplerate;
float phase = 0.f;
// naive phasor oscillator, this part goes in your process loop
phase += normalized_freq;
if (phase >= 1.0) {
phase -= 1.0; // wrap
}
float lfo = sin( 2*M_PI * phase );
you’ll need these global variables:
float lfo_freq, center_freq, mod_depth, feedback, wetness; // FX parameters
float last_output;
float flanger_fx( float input ) {
//… paste in LFO code here …
float flange_delay = 0.5 * samplerate / ( center_freq + mod_depth* lfo); //
this is the number of samples to delay
set_delay( flange_delay ); // see
https://ccrma.stanford.edu/~jos/doppler/Variable_Delay_Line_Software.html
output = input + wetness*delay_line(input) - feedback*last_output;
last_output = output; // store the state for the feedback
return output;
}
good luck!
-David
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Re: [music-dsp] LFO Flanger.
Hi Darwin,
Check out the diagram here: http://www.donreiman.com/Flanger/Flanger.htm
A flanger works like this: y[n] = x[n] + wetness*x[n-delay] + feedback*y[n-1]
where delay = 0.5*sample_rate / ( center_frequency + mod_depth* LFO[phasor] )
(This will be a float so you need to make an interpolated delay line for
sub-sample delays)
your LFO is a lookup table, so you will need a phasor (sawtooth oscillator
between scaled between 0 and 1) to read it.
It is also common to use a triangle wave LFO rather than a sine wave.
-David
On Jun 25, 2014, at 2:48 AM, Darwin Sese wrote:
> I am writing a basic flanger. i successfully create an delay or echo
> effects.
>
> i am coding an LFO. but i dont know how to do it. here is my code.
>
>
> LFO = (float) sin( ((double)i++/(double)TABLE_SIZE) * M_PI * 2. );
>
>
> here is my full code functions.
>
> float flanger( float input )
> {
>what is the correct value for LFO?
>LFO = (float) sin( ((double)i++/(double)TABLE_SIZE) * M_PI * 2.
> );
>
>
> //how i can put the lfo here?
>echo_buffer[echooffset]=input*0.5;
>if (echooffset++>=ECHO-1)echooffset=0;
>output+=echo_buffer[echooffset]*0.4;// amount to mix into dry
>if (echooffsetplay++>=ECHO-1)echooffsetplay=0;
>
>
>
>
>
>
>return output;
>
>
>
> }
>
>
>
> Thanks in advance.
> Darwin
>
>
>
>
>
> --
> Darwin Sese
>
>
>
Re: [music-dsp] Best way to do sine hard sync?
On Mar 21, 2014, at 5:26 AM, Emanuel Landeholm wrote: > Also, David Lowenfels, would you like to share your pd implementation? Sure. I was able to dust it off and get it running, but there are some external dependencies which I need to remove in order to make it portable. It will take me a few days to clean it up. I played around with it this morning, and there is some audible aliasing in the higher octave, especially when using the more rectangular-ish shaped windows (which is no big surprise). Oversampling would help for sure. My patch uses an envelope follower on a drum loop to modulate the master/slave ratio. It does sound pretty cool with a sine wave, kind of like one of those plastic whirly tubes. -David signature.asc Description: Message signed with OpenPGP using GPGMail -- 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] 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 parameter of control. http://music.columbia.edu/pipermail/music-dsp/2001-October/045626.html http://dl.dropbox.com/u/285841/hardsync.tar.bz2 -David On Mar 10, 2014, at 5:02 PM, Marco Lo Monaco wrote: > Hello Tobias, > You should also have a look at the BLOO method, explained in this thread a > long time ago > http://music.columbia.edu/pipermail/music-dsp/2009-june/067853.html > That thread is quite long and the discussion pretty animated but you could > get some new ideas from the the paper of George at the link > http://s1gnals.blogspot.it/2008/12/bloo_6897.html and look for additonal > interpretations of in in the thread itself. > > Hope to have helped > > Marco > >> -Messaggio originale- >> Da: music-dsp-boun...@music.columbia.edu [mailto:music-dsp- >> boun...@music.columbia.edu] Per conto di Tobias Münzer >> Inviato: martedì 25 febbraio 2014 15:54 >> A: music-dsp@music.columbia.edu >> Oggetto: [music-dsp] Best way to do sine hard sync? >> >> Hi, >> >> I would like to implement a hard-synced sine oscillator in my synth and I > am >> wondering which is the best way to do so. >> I read the paper 'Generation of bandlimited sync transitions for sine >> waveforms' by Vadim Zavalishin which compares several approaches. >> Are there any better ways then the 'frequency shifting method' described > in >> the paper? (Better in terms of less aliasing, faster,..) >> >> Thanks a lot >> >> Best Regards >> Tobias >> -- >> 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 > > -- > 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 signature.asc Description: Message signed with OpenPGP using GPGMail -- 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] filter smoothly changeable from LP<->BP<->HP? SVF
I second this idea, it's something that I've played with myself (and sounds good enough to me) but never analyzed the frequency response. I recall Urs Heckmann released a plugin years ago that had a SVF filter like this, with a polar control to crossfade between LP/BP/HP/BR/LP, with AP in the middle of the circle. -David On Feb 11, 2013, at 11:02 AM, andy butler wrote: > > > Wouldn't the obvious thing to try first be a state variable filter? > > Taking the Lo-pass, Band-pass and Hi-Pass outputs. > > Then the parameter would in the first half of it's range crossfade from LP to > BP, and in the second > half from BP to HP. > > I can confirm that mixing the outputs from an SVF sounds > musical enough...to me anyway, but don't know if > there are phase issues that mean it's not really doing a good morph between > the different shapes. > (are there?) > > andy -- 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] need help with gnuplot and octave on my mac.
Did you ever get this working? I recommend installing everything with homebrew http://mxcl.github.com/homebrew/ -D On Oct 27, 2012, at 8:35 AM, robert bristow-johnson wrote: > On 10/27/12 2:25 AM, gwenhwyfaer wrote: >> On 27/10/2012, gwenhwyfaer wrote: >>> On 26/10/2012, robert bristow-johnson wrote: say, any among you using a Mac and Octave and gnuplot? i used to be able to plot with Octave, it would start up X11 and if i set the variable GNUTERM=x11 before starting Octave this would work. now it doesn't :-( anybody know what i'm doing wrong? i could use some help. thanks for any. >>> Hmm. I don't know Macs, but from the error messages, it looks as >>> though something installed a new version of fontconfig without >>> updating freetype to match. Have you installed anything since last >>> using Octave/gnuplot that could have autoupgraded (or autodowngraded, >>> or otherwise mucked about with) half, but not all, of your X11 >>> subsystem? >> The other possibility is that there's more than one version of either >> library on your system, and somehow the DYLD_LIBRARY_PATH inherited by >> gnuplot points to the wrong one. I bet that's going to be fun to >> debug. >> >> Shared libraries - Satan's preferred method of software reuse... > > well, i agree. i have to confess i was much more productive and comfortable > back in the OS 9 days and using CodeWarrior instead of Xcode. i really > believe that computers should serve people and not the other way around. > > we'll probably have to worry about this later. > > -- > > r b-j r...@audioimagination.com > > "Imagination is more important than knowledge." > > > > -- > 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 -- 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
[music-dsp] audio plugins with Ruby and the JVM
Hi Everyone, I found this the other day and thought some people here might find this interesting. IMHO the ruby part is a bit of a mess under the hood, but the idea is really exciting to me (since Ruby is my current favorite). You can also use Mirah (f.k.a. Duby) to compile direct to JVM for better performance, or go hybrid JRuby / Java. Endpoints are cross-platform: LADSPA, VST or even AU! http://blog.logeek.fr/2009/11/17/how-to-prototype-vst-audio-plugins-with-jruby-and-java https://github.com/thbar/opaz-plugdk > We’ve been working iteratively using Git, and a few things came out: > • a DSL that allows to use declarative syntax for parameters handling > (which usually involves a lot of boilerplate code) > • the possibility to mix and match Java (for performance) and JRuby > (for conciseness and declarativeness) as needed > • a set of rake tasks to make it easier to bundle the plugins for the 3 > platforms (Windows, Mac OS X, Linux) > • some simple but working, portable plugins > • other promising things like UI DSL, IRB console for live code hacking > (!) > • a lot of fun :) I also found another cool audio project using the JVM, this time with clojure: http://overtone.github.com/ -David -- 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] maintaining musicdsp.org
adding recaptcha to an existing site would not be too difficult, and would get the job done. if you decide to overhaul... I'm partial to Rails, it's pretty awesome. (disclosure I'm a ruby/rails developer as my day job) there are a handful of CMS solutions for rails 3, here are two options that look decent... I could help customize http://refinerycms.com/ http://www.browsercms.org/ -D On Apr 5, 2012, at 5:57 PM, Kevin Dixon wrote: > I would vote for a CAPTCHA... specifically recaptcha > http://www.google.com/recaptcha/whyrecaptcha > > -Kevin > > On Thu, Apr 5, 2012 at 1:03 PM, Bastian Schnuerle > wrote: >> just did wordpress for a friend .. looks nice .. +1 .. >> >> Am 05.04.2012 um 21:50 schrieb douglas repetto: >> >> >>> >>> I think even Wordpress would work very well for the content on >>> musicdsp.org. I agree a full drupal site seems like overkill! >>> >>> douglas >>> >>> On 4/5/12 10:05 AM, Bjorn Roche wrote: On Apr 5, 2012, at 4:53 AM, Ross Bencina wrote: > Hey Bjorn, > > On 5/04/2012 1:52 AM, Bjorn Roche wrote: >> >> Any thoughts about modernizing the whole thing with a fresh CMS? >> I think it would be easier to maintain, have built-in spam >> filters, and it would be easier to have multiple people do the >> work. Plus it would look more attractive. I don't think it would >> take much effort to redo the whole thing in, say, drupal. > > > Have you ever set up a Drupal site? I have. It is not for > small-time, non-commercial, low-maintenance overhead projects > imho. Yes. Quite a few. > Imho it would be a huge job to port the current site to Drupal and > there is a lot of ongoing maintenance required to keep security > patches up to date etc etc. Yes. The biggest problem is security updates. You are right: major PITA factor. This can be mitigated by a hosted solution, or a multi-site install where someone is already monitoring the site for security updates. But, at the end of the day, that might not be realistic. > Doing the theme port alone would be a lot of work. I would not dream of porting the existing theme, but rather use a new, or built-in theme. > Unless I'm completely out of touch it is really non-trivial to set > up something like musicdsp.org in Drupal with adequate spam > filtering. The standard Drupal capcha solution (Mollom) is not > great -- in my experience it flags a lot of false positives (spam > that isn't spam). Mollom sucks. Captchas alone catch the vast majority of spam. The rest can be handled with moderation. > Anyway, this is really just a vote against Drupal for musicdsp.org, > not against using a CMS. > > I actually think the current ad-hoc php solution is not so bad -- > but Bram knows more about these things than me. Recaptcha could be added to the existing site with fairly little effort, but there are other advantages to a CMS: they are easier to team-manage, organize, and they have a number of potentially useful features like taxonomies (giving the ability to tag and categorize algos by language and purpose for example.) bjorn - Bjorn Roche http://www.xonami.com Audio Collaboration http://blog.bjornroche.com -- 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 >>> >>> -- >>> ... http://artbots.org >>> .douglas.irving http://dorkbot.org >>> .. http://music.columbia.edu/cmc/music-dsp >>> ...repetto. http://music.columbia.edu/organism >>> ... http://music.columbia.edu/~douglas >>> >>> -- >>> 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 >> >> >> -- >> 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 > -- > 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 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list
Re: [music-dsp] maintaining musicdsp.org
hmm... what about a CAPTCHA? -D -- 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] Orfanidis-style filter design
I have the PDF. The cool thing about this paper which is not mentioned in the abstract is the least-squares fit to the spectrum magnitude of an arbitrary 2-pole filter, giving a very nice biquad with no need for oversampling to get a good nyquist response. I have some matlab/octave code for this somewhere... I made an EQ using this many years ago. -D On Nov 27, 2011, at 2:29 PM, robert bristow-johnson wrote: > On 11/27/11 3:17 PM, Dominique Würtz wrote: >> Any ideas? >>> Knud Christensen "A Generalization of the Biquadratic Parametric" >>> http://www.aes.org/e-lib/browse.cfm?elib=12429 >>> >> Hmm, reading the abstract I'm not 100% sure if it really addresses what >> I'm aiming at. Sorry for being sceptical, but before I shell out 20$ for >> this, can you confirm me that it actually considers the gain correction >> at Nyquist? >> > what it does is inverse-map the 5 coefficients of the biquad filter to 5 > parameters that users might find useful. there *are* only 5 degrees of > freedom, so it really is only an issue for how you want those 5 control knobs > defined. > > why not put out a request for anyone here at music-dsp who has free access to > the full AES archives (i don't, sorry) to send you the pdf? maybe there is a > similar paper by the author living out there in the internet. maybe you can > find and contact the author. i have a paper copy of the preprint > *somewhere*, but i am not sure where. > > L8r, > > -- > > r b-j r...@audioimagination.com > > "Imagination is more important than knowledge." > > > > -- > 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 -- 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] Trapezoidal and other integration methods applied tomusical resonant filters
thanks for sharing this Andy, also to Vadim and RBJ for contributing. I'm following this thread with great interest! -David -- 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] quasi-bandlimited sawtooth and pulse waveforms
Hi Kalle, Thanks for sharing your experimentation. After a second glance, this appears to me quite similar to minBLEP (minimum-phase bandlimited step) synthesis, in that you have a pre-integrated lookup table. This avoids the DC-leak from BLIT, which RBJ brought up, and IMHO makes BLIT untenable for sawtooth (though bipolar blit for square waves does not have this problem). I'm a bit rusty in csound, so it's not clear to me from the code how you sweep the mainlobe frequency... is this just changing the exponent? The cool thing about using min-phase is that you can do stuff like hard-sync, because it doesn't require a priori knowledge about the phase. Best, -David On Mar 26, 2011, at 5:30 PM, kalle@helsinki.fi wrote: > Hi, > > this might be of interest: > > http://www.csounds.com/node/1475 > > Kalle Aho > > > > -- > 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 -- 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] Modular synthesis percussion?
Hi Alan, The most famous analog drum machines are the Roland TR 808 and 909. Look at the block diagrams in the service manuals: http://ericarcher.net/wp-content/uploads/2009/11/808-svc-man.pdf http://privat.bahnhof.se/wb447909/dinsync/service_manuals/TR-909.pdf This is a great analysis of the 909 circuits: http://hardware.freepage.de/raf909/ For building actual circuits, this book is great: "The Electronic Drum Cookbook" http://www.magsmoke.com/thomas_henry_books.asp -David On Feb 17, 2011, at 9:51 AM, Alan Wolfe wrote: > Hey Guys, > > Does anyone know how to do percussion sounds with modular synthesis? > > I'm talking about just using VCO, VCA, EG etc, not an actual > percussion module (: > > I've been looking around and all the info i can find shows people > using percussion modules which isn't so helpful if you only have the > basic tools at your disposal hehe. > > Or i guess, info about percussion synthesis in the DSP realm at all > would be nice too if anyone can point me at any of that. > > Thank you!! > Alan > -- > 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 -- 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
