Wave Equation for Ideal Strings

The ideal string results in an oscillation without losses. The differential wave-equation for this process is defined as follows. The velocity \(c\) determines the propagation speed of the wave and this the frequency of the oscillation.

A solution for the different equation without losses is given by d'Alembert (1746). The oscillation is composed of two waves - one left-traveling and one right traveling component.

• \(y^+\) = left traveling wave

• \(y^-\) = right traveling wave

Tuning the String

The velocity \(c\) depends on tension \(K\) and mass-density \(\epsilon\) of the string:

With tension \(K\), cross sectional area \(S\) and density \(\rho\) in \({\frac{g}{cm^3}}\).

Frequency \(f\) of the vibrating string depends on the velocity and the string length:

Make it Discrete

For an implementation in digital systems, both time and space have to be discretized. This is the discrete version of the above introduced solution:

For the time, this discretization is bound to the sampling frequency \(f_s\). Spatial sample distance \(X\) depends on sampling-rate \(f_s = \frac{1}{T}\) and velocity \(c\).

• \(t = \ nT\)

• \(x = \ mX\)

• \(X = cT\)

Using MIDI CC

Using MIDI in Faust requires only minor additions to the code and compiler arguments. For first steps it can be helpful to control single synth parameters with MIDI controllers. This can be configured via the UI elements. The following example uses MIDI controller number 48 to control the frequency of a sine wave by adding [midi:ctrl 48] to the hslider parameters.

// midi-example.dsp
//
// Control a sine wave frequency with a MIDI controller.
//
// Henrik von Coler
// 2020-05-17

import("stdfaust.lib");

freq = hslider("frequency[midi:ctrl 48]",100,20,1000,0.1) : si.smoo;

process = os.osc(freq) <: _,_ ;

CC 48 has been chosen since it is the first slider on the AKAI APC mini. If the controller numbers for other devices are not known, they can be found using the PD patch reverse_midi.pd.

Compiling with MIDI

In order to enable the MIDI functions, the compiler needs to be called with an additional flag -midi:

$ faust2xxxx -midi midi_example.dsp

This flag can also be combined with the -osc flag to make synths listen to both MIDI and OSC.

Note Handling & Polyphony

Typical monophonic and polyphonic synth control can be added to Faust programs by defining and mapping three parameters:

• freq

• gain

• gate

When used like in the following example, they will be linked to the parameters of MIDI note on and note off events with a frequency and a velocity.

// midi_trigger.dsp
//
// Henrik von Coler
// 2020-05-17

import("stdfaust.lib");
freq    = nentry("freq",200,40,2000,0.01) : si.polySmooth(gate,0.999,2);
gain   = nentry("gain",1,0,1,0.01) : si.polySmooth(gate,0.999,2);
gate   = button("gate") : si.smoo;

process = vgroup("synth",os.sawtooth(freq)*gain*gate <: _,_);

Compiling Polyphonic Code

$ faust2xxxx -midi -nvoices 12 midi_trigger.dsp

MIDI on Linux

Faust programs use Jack MIDI, whereas MIDI controllers usually connect via ALSA MIDI. In order to control the synth with an external controller, a bridge is nedded:

$ a2jmidi_bridge

The MIDI controller can now connect to the a2j_bridge input, which is then connected to the synth input.

Splitting a Signal

import("stdfaust.lib");

// a source signal
signal = os.imptrain(5);

// split signal to stereo in process function:
process = signal <: _,_;

process = signal <: _,_,_,_,_,_,_,_;

Merging Signals

import("stdfaust.lib");

// create four sine waves
// with arbitrary frequencies
s1 = 0.2*os.osc(120);
s2 = 0.2*os.osc(340);
s3 = 0.2*os.osc(1560);
s4 = 0.2*os.osc(780);

// merge them to two signals
process = s1,s2,s3,s4 :> _;

// merge them to two signals
process = s1,s2,s3,s4 :> _,_;

Early Mechanical

Early use of the Fourier representation, respectively additive synthesis, for modeling musical sounds has been made by Hermann von Helmholtz. He built mechanical devices, based on tuning forks, resonant tubes and electromagnetic excitation for additive synthesis. Von Helmholtz used these devices for investigating various aspects of harmonic sounds, including spectral distribution and relative phases.

Tuning forks with resonant tubes (von Helmholtz, 1870, p.183).

Early Analog

The history of Elektronische Musik started with additive synthesis. In his composition Studie II, Karlheinz Stockhausen composed timbres by superimposing sinusoidal components. In that era this was realized through single sine wave oscillators, tuned to the desired frequency and recorded on tape.

Studie II is the attempt to fully compose music on a timbral level in a rigid score. Stockhausen therefor generated tables with frequencies and mixed tones for creating source material. Fig.1 shows an excerpt from the timeline, which was used to arrange the material. The timbres are recognizable through their vertical position in the upper system, whereas the lower system represents articulation, respectively fades and amplitudes.

Fig.1: From the score of Studie II.

Early Digital

Max Mathews

As mentioned in the Introduction, Max Mathews used additive synthesis to generate the first digitally synthesized pieces of music in the 1950s. In the early 1960s, Mathews had advanced the method to synthesize dynamic timbres, as in Bycicle Built for Two:

Iannis Xenakis

In his Electroacoustic compositions, Iannis Xenakis made use of the UPIC system for additive synthesis (Di Scipio, 1998), as for example in his Mycenae-Alpha (1977).

Follow this link for more information on the UPIC system (and many more instruments) 120 Years.

Anti-Aliasing Filters

In analog-to-digital conversion, simple anti-aliasing filters can be used to band-limit the input and discard signal components above the Nyquist frequency. In case of digital synthesis, however, this principle can not be applied. When generating a square wave signal with an infinite number of harmonics, aliasing happens instantaneously and can not be removed, afterwards.

Band Limited Generators

In order to avoid the aliasing, band-limited signal generators are provided in most audio programming languages and environments.

Breadboard Circuit

The breadboard circuit is the same as in the first Arduino sensor example:

Arduino Code

For the SC example, serial data is sent in a simple way. The additional scaling is optional, but makes it easier to process the data in SuperCollider.

void setup() {

   Serial.begin(9600);
}

void loop() {

 int sensorValue = analogRead(A0);

 // scale to 0..1
 float voltage   = sensorValue/1024.0 ;

 Serial.println(voltage);

}

SC Code

On Linux, the Arduino's serial interface can be found in the terminal:

$ ls -l /dev/ttyACM*

On the SC receiver end, a serial port object is initialized with the matching serial interface:

(
p = SerialPort(
  "/dev/ttyACM0",
  baudrate: 9600,
  crtscts: true);
)

A control rate bus is used to visualize the received data and make it accessible to other nodes:

~sensorBUS = Bus.control(s,1);
~sensorBUS.scope;

The actual receiving and decoding of the data happens inside a routine with an infinite loop. It appends incoming characters, until a return character (13) is received. In this case, the assembled string is converted to a Float and written to the sensor bus:

(
r= Routine({
    var byte, str, res;
    inf.do{|i|
        if(p.read==10, {
            str = "";
            while({byte = p.read; byte !=13 }, {
                str= str++byte.asAscii;
            });
            res= str.asFloat;

            // ("read value:"+res).postln;

                    ~sensorBUS.set(res);
        });
    };
}).play;
)

External Resources

The SuperCollider Tutorial by Eli Fieldsteel shows a similar solution for getting Arduino sensors into SuperCollider via USB.

Exercise

NeoWs

At https://api.nasa.gov/, the NASA offers various APIs. This example uses data from the 'Asteroids - NeoWs' RESTful web service, which contains data of near earth Asteroids.

JSON Data Structure

The JSON data is arraned as an array, featuring the data on 20 celestial bodies, accessible via index:

links {…}
page  {…}
near_earth_objects
  0   {…}
  1   {…}
  2   {…}
  3   {…}
  4   {…}
  5   {…}
  6   {…}
  7   {…}
  8   {…}
  9   {…}
  10  {…}
  11  {…}
  12  {…}
  13  {…}
  14  {…}
  15  {…}
  16  {…}
  17  {…}
  18  {…}
  19  {…}

Harmonic Sonification

orbital_period       = sine wave frequency
absolute_magnitude_h = sine wave  amplitude

The Result

A Partial SynthDef

A SynthDef for a single partial with amplitude and frequency as arguments. In addition, the output bus can be set. The sample rate is considered to avoid aliasing for high partial frequencies.

(
SynthDef(\spatial_additive,

      {
              |outbus = 16, freq=100, amp=1|

              // anti aliasing safety
              var gain = amp*(freq<(SampleRate.ir*0.5));

              var sine = gain*SinOsc.ar(freq);

              Out.ar(outbus, sine);

      }
).send;
)

The Partial Synths

Create an array with 40 partial Synths, using integer multiple frequencies of 100 Hz. Their amplitude decreases towards higher partials. An audio bus with 40 channels receives all partial signals separately. All synths are added to a dedicated group to ease control over the node order.

~partial_GROUP = Group(s);

~npart         = 40;

~partial_BUS   = Bus.audio(s,~npart);

(
~partials = Array.fill(40,
{ arg i;
  Synth(\spatial_additive, [\outbus,~partial_BUS.index+i, \freq, 100*(i+1),\amp, 1/(1+i*~npart*0.1)],~partial_GROUP)
});
)

s.scope(16,~partial_BUS.index);

The Encoder SynthDef

A simple encoder SynthDef with dynamic input bus and the control parameters azimuth and elevation.

~ambi_BUS      = Bus.audio(s,16);

(
SynthDef(\encoder,
      {
              |inbus=0, azim=0, elev=0|

              Out.ar(~ambi_BUS,HOAEncoder.ar(3,In.ar(inbus),azim,elev));
      }
).send;
)

The Encoder Synths

An array of 16 3rd order decoders is created in a dedicated encoder group. This group is added after the partial group to ensure the correct order of the synths. Each encoder synth receives a single partial from the partial bus. All 16 encoded signals are sent to a 16-channel audio bus.

~encoder_GROUP = Group(~partial_GROUP,addAction:'addAfter');

(
~encoders = Array.fill(~npart,
      {arg i;
              Synth(\encoder,[\inbus,~partial_BUS.index+i,\azim, i*0.1],~encoder_GROUP)
});
)

~ambi_BUS.scope

The Decoder Synth

A decoder is added after the encoder group and fed with the encoded Ambisonics signal. The binaural output is routed to outputs 0,1 - left and right.

// load binaural IRs for the decoder
HOABinaural.loadbinauralIRs(s);

(
~decoder = {
Out.ar(0, HOABinaural.ar(3, In.ar(~ambi_BUS.index, 16)));
}.play;
)


~decoder.moveAfter(~encoder_GROUP);

References