This is an online version of the Audio Plug-Ins Designed with Faust workshop that took place at CCRMA (Stanford University) on July 6-10 2015. It aims at giving a broad introduction to digital signal processing and to the Faust programming language. By the end of this course, students should have a full collection of plug-ins usable in various DAWs and computer music environments. While previous experience in programming is helpful but not required to take this course, we expect students to have basic notions of algebra, calculus and to have some experience using computers to make music.
Additional information on the different topics treated in this course can be found in Julius Orion III Smith's book series:
For any question/comment/suggestion, please contact me.
1. Installing Faust
Linux
MacOSX
Windows
2. Introduction/What is Faust?
3. Basic Notions of Acoustics and Digital Audio Technologies
4. Faust Semantic Basics and Simple Audio Effects: Panner and Tremolo
Simple Gain Controller
Simple Panner
Tremolo
5. Ring Modulation (Amplitude Modulation)
Ring Modulator
Stereo Ring Modulation (Automated Panning)
6. Additive Synthesis
Simple Additive Synthesizer
7. Delay and Tour of oscillator.lib
Different Kinds of Delays in Faust
8. One Zero Lowpass/Highpass Filter, Feed-Forward Comb Filter and Simple Flanger
"One Zero" Lowpass / Highpass Filter
Feed-Forward Comb Filter
Simple Feed-Forward Flanger
9. Debugging and Generating Plug-ins
10. Iteration
Additive Synthesizer with Independent Frequency Control
11. Recursion: Echo Effect and Feedback Comb Filter
Echo Effect
Feedback Comb Filter
12. Introduction to Physical Modeling
Karplus-Strong: Virtual String
13. Other Kinds of Filters and Filter Bank
Filter Bank
14. Cubic Distortion
Cubic Distortion Plug-in
15. Vocoder and Conditions
Vocoder
16. Guest Lecture by Julius Smith: Signal Processing Libraries for Faust
17. Reverb
Simple Schroeder Reverberator: Freeverb
17. Dynamic Range Compression
Dynamic Range Compressor
If you encounter a problem during one of the following steps, feel free to send me an e-mail.
While the Faust compiler itself was designed to not rely on any external library, some of the tools gravitating around Faust do have a few dependencies. Thus you need to make sure that the following packages are installed on your system before we begin: git, llvm (3.4), llvm-dev (3.4), libmicrohttpd and qjackctl.
For the workshop, we will use some of the latest features of Faust so it is strongly recommended that you compile
and install Faust from the Git repository.
For that, open a terminal window, cd to the location where you want to download Faust's source code and
then type:
git clone git://git.code.sf.net/p/faudiostream/code faudiostream-code
cd faudiostream-code
git checkout faust2
make
sudo make installHopefully, everything went well and now if you type: faust and press the return key in your terminal,
the following message should be returned: ERROR : no files specified; for help type "faust --help".
However, if you get something like faust: command not found, something probably went wrong.
Next, we need to install FaustLive which is an "on-the-fly" compiler for Faust (we'll see during the workshop what
this means). It is a great tool that enables you
to compile Faust codes "almost in real-time" and that makes programming in Faust very smooth. While it is
strongly recommended that you compile it from the source, you can try to download and run this
pre-compiled version (64 bits only) directly. Otherwise, open a terminal window and cd to the location
where you want to download FaustLive's source code and type:
git clone git://git.code.sf.net/p/faudiostream/faustlive faudiostream-faustlive
cd faudiostream-faustlive
make
sudo make installYou should keep Faust's source code for the workshop: it will be probably helpful.
Hopefully, things went
well. To see if it worked just type: FaustLive & in your terminal and the FaustLive interface should
appear. If it does, hooray! You're now ready to code in Faust!
Optionally, you can also get the following packages: lv2-dev, ladspa-sdk,
libjack-jackd2-dev and libgtk2.0-dev as well as any dependency related to a specific Faust
architecture (to get a non-exhaustive list, just type faust2 in a terminal and then press the TAB key).
In order to install some of Faust's dependencies, you will have to use the terminal of your laptop that you can
find in the Applications folder. If you never used it before, keep in mind that you have to press the return key after
every command. Commands are indicated in this document using this font. This might sound stupid but don't
forget to press the return key to run each command :). If you encounter a problem during one of the
following steps, feel free to send me an e-mail.
The first thing you want to do is to install FaustLive. It is an "on-the-fly" compiler for Faust (we'll see during the workshop what this means) that will make your coding experience a very smooth one. For that, download FaustLive and follow the install instructions.
To make sure that it worked, open FaustLive and try to open one of the examples like "harpeautomation". If it makes sound, it's a good sign: you're now ready to start coding in Faust!
While FaustLive is great for prototyping, it's missing many "deeper" features of Faust which is why you should still install the Faust compiler. Before you do that, you should make sure that Xcode (developer and command line tools) is installed on your system.
For the workshop, we will use some of the latest features of Faust so it is strongly recommended that you compile
and install Faust from the Git repository.
For that, create a new folder in your home directory (or anywhere else) called faust. Open a terminal window,
and type: cd faust which means "go to the faust folder" from your home directory and press the return key.
After that, type the following commands to download, compile and install the Faust compiler:
git clone git://git.code.sf.net/p/faudiostream/code faudiostream-code
cd faudiostream-code
make
sudo make installYou should keep the faust folder for the workshop: it will be probably helpful. If things went well, Faust
should now be installed on your system. To make sure that it is the case, type:
faust and press the return key in your terminal, the following message should be returned:
ERROR : no files specified; for help type "faust --help". However, if you get something
like faust: command not found, something probably went wrong.
Lastly, you might want to install the VST SDK on your system if you want to generate VST plug-ins. For that,
download this file and unzip it. Open a new terminal window and cd to the
folder where you unziped the file and then type: sudo mv vstsdk2.4 /usr/local/include/. After this step,
you should be able to compile any Faust code into a VST plug-in by simply typing faust2vst yourFaustFile.dsp
in a terminal.
Optionally, you can also get any dependency related to a specific Faust
architecture (to get a non-exhaustive list, just type faust2 in a terminal and then press the TAB key). Feel
free to contact me if you have a question about that.
If you encounter a problem during one of the following steps, feel free to send me an e-mail.
Installing the Faust compiler on Windows can be complicated but fortunately, this is not the case of FaustLive that is an "on-the-fly" compiler for Faust (we'll see during the workshop what this means). For that, just download this file and follow the instructions. If you're a Jack user, you might want to check this version. Once this is done, you can try to run one of the examples like "harpeautomation". If it makes sound, it's a good sign: you're ready to code in Faust!
You should download the latest version of the source code of the Faust compiler by clicking on the "Download Snapshot" button on this page and leave it somewhere on your system as it will be very helpful during the workshop.
Lastly, if you don't already have one, you should download an "more advanced" text editor than notepad, etc. I recommend using Notepad++ for example.
import("filter.lib");
gainController = *(gain : smooth(tau2pole(interpTime)))
with{
gain = hslider("gain",0.5,0,1,0.1);
interpTime = hslider("Interpolation Time (s)",0.05,0,1,0.001);
};
process = gainController;import("filter.lib");
myPanner = _ <: *(1-pan),*(pan)
with{
pan = hslider("pan",0.5,0,1,0.01) : smooth(0.999);
};
process = myPanner;import("music.lib");
import("filter.lib");
tremolo = *(1-depth*(osc(freq)*0.5+0.5))
with{
freq = hslider("frequency",5,0.1,15,0.01) : smooth(0.999);
depth = hslider("depth",0,0,1,0.01) : smooth(0.999);
};
process = tremolo;For more theory around amplitude modulation, check this page on Julius Smith's website.
import("music.lib");
import("filter.lib");
ringMod = *(1-depth*(osc(freq)*0.5+0.5))
with{
freq = hslider("frequency",5,0.1,1000,0.01) : smooth(0.999);
depth = hslider("depth",0,0,1,0.01) : smooth(0.999);
};
process = ringMod <: _,_;import("music.lib");
import("filter.lib");
stereoRingMod = _ <: *(1-pan),*(pan)
with{
freq = hslider("frequency",5,0.01,1000,0.01) : smooth(0.999);
depth = hslider("depth",0,0,1,0.01) : smooth(0.999);
pan = 1-depth*(osc(freq)*0.5+0.5);
};
process = stereoRingMod;Check the Wikipedia page on additive synthesis to get more informations on this topic.
import("music.lib");
import("filter.lib");
freq = hslider("freq",300,20,2000,0.01) : smooth(0.999);
gain = hslider("gain",0.3,0,1,0.01) : smooth(0.999);
t = hslider("attRel (s)",0.1,0.001,2,0.001);
gate = button("gate") : smooth(tau2pole(t));
process = osc(freq),osc(freq*2),osc(freq*3) :> /(3) : *(gain)*gate;import("music.lib");
import("filter.lib");
_' // One sample delay
_@N // N samples delay
delay(NMax,N) // N samples variable delay with NMax, the maximum delay length as a power of 2
fdelay(NMax,N) // Fractional variable delay (allows N to be a decimal number)More informations about this type of filter can be found on Julius Smith's website.
oneZero = _ <: _,_'*b1 :> _
with{
b1 = hslider("b1",0,-1,1,0.01);
};
process = oneZero;More informations about this type of filter can be found on Julius Smith's website.
import("music.lib");
ffComb = _ <: _,delay(65536,N)*b1 :> _
with{
b1 = hslider("b1",0,-1,1,0.01);
N = hslider("N",1,1,500,1);
};
process = ffComb;More informations about flanging can be found on Julius Smith's website.
import("music.lib");
import("filter.lib");
myFlanger = _ <: _,(65536,N,_ : fdelay) :> _
with{
modFreq = hslider("Modulation Frequency",2,1,50,0.01) : smooth(0.999);
depth = hslider("Depth",1,1,100,0.1) : smooth(0.999);
N = osc(modFreq)*0.5+0.51 : *(depth);
};
process = myFlanger;import("music.lib");
mySine(n) = osc(freq)
with{
freq = hslider("freq %n",440,50,1000,0.01);
};
process = par(i,4,mySine(i)) :> *(0.25);import("music.lib");
import("filter.lib");
myEcho = _ <: *(dry), (+~(fdelay(65536,delLength)*feedback*-1) : *(1-dry)) :> _
with{
delLength = hslider("Time (ms)",250,0.1,1000,0.1)*0.001*SR : smooth(0.999);
feedback = hslider("Feedback",0,0,1,0.001) : smooth(0.999);
dry = hslider("Wet/Dry",1,0,1,0.01) : smooth(0.999);
};
process = myEcho;
More informations about this kind of filter can be found here on Julius Smith's website.
import("music.lib");
import("filter.lib");
myFbComb = +~(delay(2048,delLength)*(-a1))
with{
a1 = hslider("a1",0,-1,0.999,0.001) : smooth(0.999);
delLength = hslider("delLength",1,1,2000,1);
};
process = myFbComb;More informations about this algorithm can be found here on Julius Smith's website.
To learn more about physical modelling of musical instruments, check this online book by Julius Smith.
import("music.lib");
import("filter.lib");
myString(freq,feedback) = +~(fdelay4(1024,delLength) <: (_+_')/2 : *(feedback))
with{
delLength = SR/freq - 1;
};
frequency = hslider("freq",440,51,2000,0.01);
feedback = hslider("feedback",0.9,0.9,1,0.01);
gate = button("gate");
impulse = gate <: _,_' : - : >(0);
process = impulse : myString(frequency,feedback);import("music.lib");
import("filter.lib");
bandsNumber = 10;
oneBand(i) = vgroup("Band %i",peak_eq(Lfx,fx,B))
with{
highestBand = 10000;
currentFreq = highestBand*(i+1)/bandsNumber;
fx = hslider("[1]Freq[style:knob]",currentFreq,20,20000,0.1);
B = hslider("[2]Bdwth[style:knob]",100,1,5000,0.1);
Lfx = vslider("[3]Level",0,-90,10,0.1);
};
paramEqs(NBand) = hgroup("Filter Bank",seq(i,NBand,oneBand(i)));
process = paramEqs(bandsNumber);More informations on this topic can be found here on Julius Smith's website.
import("music.lib");
import("filter.lib");
distortion = +(offset) : *(pregain) : clip(-1,1) : cubic : dcblocker
with{
drive = hslider("Drive",0,0,1,0.01) : smooth(0.999);
offset = hslider("Offset",0,-0.1,0.1,0.01) : smooth(0.999);
pregain = pow(10,drive*2);
clip(lo,hi) = min(hi) : max(lo);
cubic(x) = x - x*x*x/3;
};
process = distortion;Check the Wikipedia page on Vocoders.
I also recommend How to Wreck a Nice Beach - the Vocoder from WWII to Hip-Hop by Dave Tompkins to get an overview of the history of vocoders.
import("music.lib");
import("filter.lib");
import("effect.lib");
import("oscillator.lib");
simpleVocoder(nBands,att,rel,BWRatio,excitation,source) = source <: par(i,nBands,oneVocoderBand(i,nBands,BWRatio,1) :
amp_follower_ud(att,rel) : _,excitation : oneVocoderBand(i,nBands,BWRatio)) :> _
with{
oneVocoderBand(band,bandsNumb,bwRatio,bandGain,x) = x : resonbp(bandFreq,bandQ,bandGain)
with{
bandFreq = 25*pow(2,(band+1)*(9/bandsNumb));
BW = (bandFreq - 25*pow(2,band*9/bandsNumb))*bwRatio;
bandQ = bandFreq/BW;
};
};
simpleVocoderDemo = hgroup("Simple Vocoder",lf_imptrain(freq)*gain,_ : simpleVocoder(bands,att,rel,BWRatio) <: _,_)
with{
bands = 32;
vocoderGroup(x) = vgroup("[0]Vocoder",x);
att = vocoderGroup(hslider("[0]Attack [style:knob]",5,0.1,100,0.1)*0.001);
rel = vocoderGroup(hslider("[1]Release [style:knob]",5,0.1,100,0.1)*0.001);
BWRatio = vocoderGroup(hslider("[2]BW [style:knob]",0.2,0.1,2,0.001));
excitGroup(x) = vgroup("[1]Excitation",x);
freq = excitGroup(hslider("[0]Frequency [style:knob]",140,50,2000,0.1));
gain = excitGroup(vslider("[1]Gain",0.5,0,1,0.01) : smooth(0.999));
};
process = simpleVocoderDemo;Check this section on artificial reverberation on Julius Smith's website to get more informations on the freeverb algorithm.
import("music.lib");
import("filter.lib");
monoFreeverb(fb1,fb2,damp,spread) = _ <: par(i,8,lpcf(combtuningL(i),fb1,damp)) :> seq(i,4,allpass_comb(1024,allpasstuningL(i),-fb2))
with{
lpcf(dt,fb,damp) = (+:delay(2048,dt))~ (*(1-damp) : (+ ~ *(damp)) : *(fb));
origSR = 44100;
cTuning = (1116,1188,1277,1356,1422,1491,1557,1617);
combtuningL(i) = take(i+1,cTuning)*SR/origSR : int : +(spread);
aTuning = (556,441,341,225);
allpasstuningL(i) = take(i+1,aTuning)*SR/origSR : int;
};
stereoFreeverb(fb1,fb2,damp,spread) = + <: monoFreeverb(fb1,fb2,damp,0),monoFreeverb(fb1,fb2,damp,spread);
freeverbDemo = _,_ <:
(*(g)*fixedgain,*(g)*fixedgain : stereoFreeverb(combfeed,allpassfeed,damping,spatSpread)),
*(1-g),*(1-g) :> _,_
with{
origSR = 44100;
scaleroom = 0.28;
offsetroom = 0.7;
allpassfeed = 0.5;
scaledamp = 0.4;
fixedgain = 0.1;
mainGroup(x) = vgroup("Freeverb",x);
knobsGroup(x) = mainGroup(hgroup("[0]",x));
combfeed = knobsGroup(hslider("[0]Room Size [style:knob]",0.5,0,1,0.01)*scaleroom*SR/origSR + offsetroom);
damping = knobsGroup(hslider("[1]Damping [style:knob]",0.5,0,1,0.01)*scaledamp*SR/origSR);
spatSpread = knobsGroup(hslider("[2]Spatial Spread [style:knob]",0.5,0,1,0.01)*46*SR/origSR : int);
g = mainGroup(hslider("[1]Dry/Wet",0.3,0,1,0.01));
};
process = freeverbDemo;More informations on this topic can be found here on Julius Smith's website.
import("music.lib");
import("filter.lib");
import("effect.lib");
simpleCompressor(ratio,thresh,att,rel,kneeAtt,gain) = _ <: _*(amp_follower_ud(att,rel) : linear2db : outminusindb : kneesmooth : visualizer : db2linear)*gain
with{
outminusindb(level) = max(level-thresh,0)*(1/ratio-1);
kneesmooth = smooth(tau2pole(kneeAtt));
visualizer = hbargraph("[1]Compressor Level [unit:dB]",-50,10);
};
simpleCompressorDemo = mainGroup(simpleCompressor(ratio,thresh,att,rel,kneeAtt,makeUpGain))
with{
mainGroup(x) = vgroup("Simple Compressor",x);
envelopesGroup(x) = hgroup("[0]Envelope",x);
att = envelopesGroup(hslider("[0]Attack [style:knob][unit:ms]",20,0,500,0.1)*0.001);
rel = envelopesGroup(hslider("[1]Release [style:knob][unit:ms]",20,0,500,0.1)*0.001);
kneeAtt = envelopesGroup(hslider("[2]Knee Attack [style:knob][unit:ms]",10,0,250,0.1)*0.001);
thresh = hslider("[2]Threshold [unit:dB]",-30,-60,4,0.1);
ratio = hslider("[3]Ratio",1,1,10,0.01);
makeUpGain = hslider("[4]Makeup Gain [unit:dB]",40,-96,96,0.1) : db2linear;
};
process = simpleCompressorDemo;