# FM Pair Oscillator

This is a `sndkit`

algorithm. A more up-to-date version
can be found here.

### Introduction

This document outlines an implementation of a classic FM oscillator with frequency, C:M ratio, and modulation index control.

There is quite a lot to be said about FM synthesis, but a classic FM oscillator implementation is poetic tribute to the technique. Requiring only two oscillators (usually sine waves), an FM operator pair is capable of generating very rich and dynamic spectrums. For how little computation is required, these suckers can pack a punch!

The FM pair is the Bechamel sauce of the FM world. Creating one from scratch provides great sonic intution into the mostly unintuitive world of FM synthesis programming. It can be the building block for grokking more complex modulation configurations, such as those found on the DX7.

### Some words on FM synthesis

(I can't help but ramble a little bit about the topic of FM synthesis. Feel free to scroll/skip/page ahead.)

FM synthesis, or Frequently Modulation synthesis, was first implemented by John Chowning while doing research at the Stanford Artificial Intelligence Laboratory (SAIL, whose spritual successor would become CCRMA, or the Center for Computer Research in Music and Acoustics). As the legend goes, Chowning stumbled upon the technique by accident. Experimenting with different vibrato speeds, he found that when vibrato got fast enough, it started changing timbre and spectrum.

The humble beginings of FM synthesis started out with an FM pair configuration similar to this one. But can you imagine the surprise he got when he first heard that new FM sound?

To paint a picture:

At the time, computers were about the size of fridges. They were expensive devices owned by the University, and time spent using the computer was precious. Chowing no doubt had to fight to get his time slot. Being a composer amongst scientists and engineers, he probably could only work in late hours.

Typing away at a mainframe 3am, Chowning would have to had to wait at least a day for the computer to run his FM synthesis program. Assuming it successfully ran without any errors, Chowning would then have to ship the output of that program off somewhere to get it converted to magnetic reel-to-reel tape. Back then, digital-to-analog converers (DACs) were things that you had to get in a car and drive to. Now-a-days, audio DACs are things found on pretty much every electronic device you own.

So, if you're keeping track, that's at least a week from code to sound. That's an enormous amount of round-trip latency! To go that long, and to finally hear these new sounds must have been such a rewarding experience.

Chowing's original FM synthesis algorithm would eventually become one of Stanford University's highest grossing patents of all time. To this day, the royalties of this patent help to sustain the CCRMA facilities.

Famously, the patent was sold to Yamaha, and that gave us things like the DX7 synthesizer (and all the cheesy music that came with it). Due to the efficiency of the algorithm, it also found itself in a whole generation of sound cards at the time, used in video game consoles and arcade machines. The most famous of these chips was the YM2151, also known as the OPM chip.

### Overview of Algorithm and Implementation

An FM Pair uses two oscillators, sometimes referred two as
`operators`

. One of these operators synthesizes audble sound.
This known as the `carrier`

, and can be remembered because
it **carries** the sound signal to the speaker. The other
operator **modulates** the frequency of the carrier, and it
is known as the `modulator`

.

If you set the modulator to any frequency that is audio rate (give or take 20hz or higher), you will technically get some kind of FM synthesis. However, depending on the frequency of the carrier, the spectrum can range from being very pitched with harmonically related overtones, to something quite clangorous and not at all something you could easily play "Hot Cross Buns" on. So, to make an FM pair easier to use, three unit-less parameters are used to control the FM characteristics of the sound that work evenly across the spectrum: a carrier-to-modulator frequency ratio known as the C:M ratio, and something known as the modulation index, which can kind of be thought of as a thing that controls the amount of modulation.

The C and M parameters are values that primarily multiply the frequency value of the carrier and modulator operators, respectively.

The modulation index can be defined as having the following relationship (taken from the Computer Music Tutorial's chapter on Modulation synthesis, page 229):

$$ I = D/M $$

Where $I$ is the modulation index, $M$ is the modulation
frequency (which would be the oscillator frequency multiplied
by the modulator ratio), and $D$ is the amount of modulation
`depth`

. The equation can be reworked in terms of $D$:

$$ D = MI $$

This depth value can be thought of as the "amplitude" amount the modulator operator (in Hz). This will produce a signal that modulates the carrier frequency up the depth amount, and then down that same amount.

In this implemetation, two table-lookup oscillators will be used for the operators, using an algorithm similar to the table lookup oscillator, which employ a fixed-point phasor with linear interpolation. Details for the gory details of these oscillators are beyond the scope of this article, but are described in the implementation referenced above.

### Tangled Files

`fmpair.c`

contains the ANSI C code required for the
implementation.

*<<fmpair.c>>=*```
#include <stdint.h>
#include <math.h>
#define SK_FMPAIR_PRIV
#include "fmpair.h"
<<constants>>
<<funcs>>
```

`fmpair.h`

is the corresponding header file for the C file.
It contains all the forward declarations. If
`SK_FMPAIR_PRIV`

is defined, it exposes the structs.

*<<fmpair.h>>=*```
#ifndef SK_FMPAIR_H
#define SK_FMPAIR_H
#ifndef SKFLT
#define SKFLT float
#endif
<<typedefs>>
<<funcdefs>>
#ifdef SK_FMPAIR_PRIV
<<structs>>
#endif
#endif
```

### Struct

*<<typedefs>>=*`typedef struct sk_fmpair sk_fmpair;`

*<<structs>>=*```
struct sk_fmpair {
<<sk_fmpair>>
};
```

### Table-lookup oscillator data and constants

#### Constants

The constant `SK_FMPAIR_MAXLEN`

is the maximum length a
lookup table can be.

The constant `SK_FMPAIR_PHASEMASK`

is a bitmask used by
the fixed point phasor.

*<<constants>>=*```
#define SK_FMPAIR_MAXLEN 0x1000000L
#define SK_FMPAIR_PHASEMASK 0x0FFFFFFL
```

#### Struct Data

The FM oscillator implements two independent table-[ookup oscillators. Each one needs their own set of wavetables, incrementors, sizes, and cached phase positions.

The convention used will this: any variables used for the carrier oscillator will begin with 'c':

*<<sk_fmpair>>=*```
SKFLT *ctab;
int csz;
int clphs;
```

*<<init>>=*```
fmp->ctab = ctab;
fmp->csz = msz;
```

Any variables used for the modulator oscillator will begin with 'm':

*<<sk_fmpair>>=*```
SKFLT *mtab;
int msz;
int mlphs;
```

*<<init>>=*```
fmp->mtab = mtab;
fmp->msz = msz;
```

In order to be used by the fixed point phasor,
the initial phases need to be rescaled by `SK_FMPAIR_MAXLEN`

and then converted to an integer value. These are
stored in each respective variable keeping track
of phase.

*<<init>>=*```
fmp->clphs = floor(ciphs * SK_FMPAIR_MAXLEN);
fmp->mlphs = floor(miphs * SK_FMPAIR_MAXLEN);
```

A number of constants are used with these tables: `nlb`

,
`inlb`

, `mask`

, and `maxlens`

. These are beyond the scope
of this document, but are touched upon in osc.
Since these are all based on table size, two sets of
constants are derived for the carrier and modulator.

*<<sk_fmpair>>=*```
/* carrier constants */
uint32_t cnlb;
SKFLT cinlb;
uint32_t cmask;
/* modulator constants */
uint32_t mnlb;
SKFLT minlb;
uint32_t mmask;
SKFLT maxlens;
```

*<<init>>=*```
{
uint32_t tmp;
/* carrier */
tmp = SK_FMPAIR_MAXLEN / csz;
fmp->cnlb = 0;
while (tmp >>= 1) fmp->cnlb++;
/* modulator */
tmp = SK_FMPAIR_MAXLEN / msz;
fmp->mnlb = 0;
while (tmp >>= 1) fmp->mnlb++;
}
/* phase mask for dividing lower/upper bits */
fmp->cmask = (1<<fmp->cnlb) - 1;
fmp->mmask = (1<<fmp->mnlb) - 1;
/* constant used to convert to floating point */
fmp->cinlb = 1.0 / (1<<fmp->cnlb);
fmp->minlb = 1.0 / (1<<fmp->mnlb);
/* max table length in seconds */
/* used to convert cycles-per-second units to cycles */
fmp->maxlens = 1.0 * SK_FMPAIR_MAXLEN / sr;
```

### Initialization

The FM oscillator is initialize with `sk_fmpair_init`

.
It needs the sampling rate `sr`

, and two sets of
wavetables (`cwt`

and `mwt`

),
their sizes (`csz`

) and (`msz`

), and
initial phases (`ciphs`

and `miphs`

) for the carrier (c)
and modulator (m) oscillators.

In a classic FM pair, these would be identical, and the wavetable would contain a sampled period of a sine wave.

*<<funcdefs>>=*```
void sk_fmpair_init(sk_fmpair *fmp, int sr,
SKFLT *ctab, int csz, SKFLT ciphs,
SKFLT *mtab, int msz, SKFLT miphs);
```

*<<funcs>>=*```
void sk_fmpair_init(sk_fmpair *fmp, int sr,
SKFLT *ctab, int csz, SKFLT ciphs,
SKFLT *mtab, int msz, SKFLT miphs)
{
<<init>>
}
```

### Parameter Control

#### Frequency

Set with `sk_fmpair_freq`

.

*<<funcdefs>>=*`void sk_fmpair_freq(sk_fmpair *fmp, SKFLT freq);`

*<<funcs>>=*```
void sk_fmpair_freq(sk_fmpair *fmp, SKFLT freq)
{
fmp->freq = freq;
}
```

*<<sk_fmpair>>=*`SKFLT freq;`

A sensible default of A440.

*<<init>>=*`sk_fmpair_freq(fmp, 440);`

#### Carrier/Modulator Ratios

Set with `sk_fmpair_modulator`

and `sk_fmpair_carrier`

.

*<<funcdefs>>=*```
void sk_fmpair_modulator(sk_fmpair *fmp, SKFLT mod);
void sk_fmpair_carrier(sk_fmpair *fmp, SKFLT car);
```

*<<funcs>>=*```
void sk_fmpair_modulator(sk_fmpair *fmp, SKFLT mod)
{
fmp->mod = mod;
}
void sk_fmpair_carrier(sk_fmpair *fmp, SKFLT car)
{
fmp->car = car;
}
```

*<<sk_fmpair>>=*```
SKFLT car;
SKFLT mod;
```

A 1:1 ratio is a good strong and sensible default.

*<<init>>=*```
sk_fmpair_carrier(fmp, 1);
sk_fmpair_modulator(fmp, 1);
```

#### Modulation Index

Set with `sk_fmpair_modindex`

.

*<<funcdefs>>=*`void sk_fmpair_modindex(sk_fmpair *fmp, SKFLT index);`

*<<funcs>>=*```
void sk_fmpair_modindex(sk_fmpair *fmp, SKFLT index)
{
fmp->index = index;
}
```

*<<sk_fmpair>>=*`SKFLT index;`

A modulation index of 1 on with 1:1 C-to-M ratio produces a very warm and versatile initial sound.

*<<init>>=*`sk_fmpair_modindex(fmp, 1);`

### Computing a Sample

A single sample of audio is computed with `sk_fmpair_tick`

.

*<<funcdefs>>=*`SKFLT sk_fmpair_tick(sk_fmpair *fmp);`

*<<funcs>>=*```
SKFLT sk_fmpair_tick(sk_fmpair *fmp)
{
SKFLT out;
SKFLT cfreq, mfreq;
SKFLT modout;
int ipos;
SKFLT frac;
SKFLT x[2];
out = 0;
<<calculate_frequencies>>
<<table_lookup_for_modulator>>
<<scale_modulator_output>>
<<modulate_carrier_frequency>>
<<table_lookup_for_carrier>>
<<update_phase_positions>>
return out;
}
```

The carrier and modulator frequencies are calculated by multiplying their respective ratio values with the oscillator frequency.

*<<calculate_frequencies>>=*```
cfreq = fmp->freq * fmp->car;
mfreq = fmp->freq * fmp->mod;
```

The modulator oscillator does its table-lookup and linear interpolation with some bitwise magic. The details of this can be found in osc.

*<<table_lookup_for_modulator>>=*```
fmp->mlphs &= SK_FMPAIR_PHASEMASK;
ipos = fmp->mlphs >> fmp->mnlb;
x[0] = fmp->mtab[ipos];
if (ipos == fmp->msz - 1) {
x[1] = fmp->mtab[0];
} else {
x[1] = fmp->mtab[ipos + 1];
}
frac = (fmp->mlphs & fmp->mmask) * fmp->minlb;
modout = (x[0] + (x[1] - x[0]) * frac);
```

Before it can modulate the carrier, the output of the modulator needs to be appropriately scaled. This amplitude amount is the modulator frequency and modulation index multiplied together.

*<<scale_modulator_output>>=*`modout *= mfreq * fmp->index;`

The frequency of the carrier oscillator is now ready to be modulated. Modulation is an addition operation; The output of the scaled modulator is tacked on the carrier frequency.

*<<modulate_carrier_frequency>>=*`cfreq += modout;`

A table-lookup operation computation happens for the carrier oscillator, similar to what happened with the modulator oscillator. This output is what gets returned.

*<<table_lookup_for_carrier>>=*```
fmp->clphs &= SK_FMPAIR_PHASEMASK;
ipos = (fmp->clphs) >> fmp->cnlb;
x[0] = fmp->ctab[ipos];
if (ipos == fmp->csz - 1) {
x[1] = fmp->ctab[0];
} else {
x[1] = fmp->ctab[ipos + 1];
}
frac = (fmp->clphs & fmp->cmask) * fmp->cinlb;
out = (x[0] + (x[1] - x[0]) * frac);
```

To wrap things up, the phase increments and positions of both oscillators are updated, based on their respective frequencies.

increment value is derived by multiplying oscillator frequency (in units of cycles-per-second) to the maximum length of the phasor (in units of seconds). The seconds unit cancels, and the resulting output is cycles. This value, truncated to an integer value, becomes the phase increment value.

*<<update_phase_positions>>=*```
fmp->clphs += floor(cfreq * fmp->maxlens);
fmp->mlphs += floor(mfreq * fmp->maxlens);
```

### C:M Ratio tips

Some suggestions to get started with picking out good C,M and I parameters. For those starting out, these should help build some sonic intuition in FM.

Generally speaking, the more rational C:M ratios are (1:1, 1:2, 3:2, etc), the more harmonic and pitched the spectrum will be. The more irrational they are (1:1.6180339..., 1:3.1415926..., etc) the more clangorous and unpitched the spectrum will be.

Fractional or strange looking ratios that look irrational can sometimes be simple ratios in disguise. For example, 1:0.5 is harmonically similar to 2:1.

The carrier ratio value determines what the pitch will be. Whole integer values correspond with the harmonic series. A value of 1 is the base frequency, 2 is an octave, 3 is an octave and a fifth, etc.

The modulator ratio can be thought of as how spread-out the harmonic spectrum will be. A 1:1 ratio will give you a denser spectrum than a 1:7 ratio. These spread out spectrums were what made classic DX7 keyboard tine sounds work so well. Using ratios with high M values, they were able to cut through a mix while also leaving room for other instruments.

Adding small fractional values to the M ratio (ex: 1:1.007) can add some very unique spectral beating. FM pairs can be layered together with different fractional amounts to create thicker sounds.

In an FM pair, the behavior of the modulation index can be thought of as a kind of tone control, similar to a filter cutoff control that you'd find in a subtractive synthesizer. It's a gross simplification for all the amazing things happening to the spectrum, but it's close enough.

Controlling the modulation index of a 1:1 FM pair kind of feels like controlling a subtractive synthesizer, especially for lower modulation index values.

FM sounds can alias like crazy, especially when frequency, modulator, and modulation index values are high. When you crank up the modulation index crazy high, you can get some very interesting alias noises, but finding the sweet spots can be a bit of a treasure hunt. The plugin Mr.Alias is built around this concept.

Keyboard/frequency scaling the modulation index is a Good Idea. As frequencies get lower, the modulation index can get larger without risk of aliasing. More modulation index means more harmonic content, means more sonic beef for lower notes.

The modulation index spectrum behavior is determined by Bessel Functions of the First kind. The wiki page on this has a good chart of this that plots the amplitudes first 3 harmonics. You can see how ahey dip in and out. One should at least be passively aware that this happens. Changing the modulation index on a bass sound, for example, can sometimes cause the fundamental to drop out, which can produces thin patches of sound.