REAPER JSFX Audio Effects

DSP algorithms and patterns for audio effects in JSFX. Covers filters, dynamics processors, delays, reverbs, and modulation effects.

Requires: reaper-jsfx-core for language fundamentals.

Rules

Rule	Description
filters	Biquad, SVF, ZDF, one-pole filters with coefficient calculation
dynamics	Compressor, limiter, gate, expander, envelope detection
delays	Delay lines, circular buffers, chorus, flanger, comb filters
reverb	Algorithmic reverb, allpass networks, FDN, early reflections
distortion	Waveshaping, soft/hard clipping, saturation, tube emulation
modulation	LFO patterns, tremolo, vibrato, phaser, ring modulation

Official Documentation

• JSFX Programming Reference
• JSFX Memory/FFT Functions

External References

• Audio EQ Cookbook - Biquad filter coefficients
• Tale's JSFX Libraries - Reference implementations
• musicdsp.org - DSP algorithms and code

Key Principles

1. Coefficient Calculation in @slider

Always calculate filter coefficients when parameters change, not per-sample:

@slider
omega = 2 * $pi * cutoff / srate;
sin_omega = sin(omega);
cos_omega = cos(omega);
alpha = sin_omega / (2 * q);

// Lowpass coefficients (RBJ cookbook)
a0 = 1 + alpha;
b0 = ((1 - cos_omega) / 2) / a0;
b1 = (1 - cos_omega) / a0;
b2 = b0;
a1 = (-2 * cos_omega) / a0;
a2 = (1 - alpha) / a0;

@sample
// Only apply coefficients here
out = b0*in + b1*z1 + b2*z2 - a1*z1_out - a2*z2_out;

2. State Variable Preservation

Filter and delay states must persist between samples:

@init
// Initialize filter states to zero
z1 = z2 = 0;
z1_out = z2_out = 0;

@sample
// Update states after processing
z2 = z1;
z1 = input;
z2_out = z1_out;
z1_out = output;

3. Gain Staging

Maintain proper gain throughout the signal chain:

@init
// Keep track of gain changes
input_gain = 1;
output_gain = 1;

@slider
// Convert dB to linear
input_gain = 10^(input_db / 20);
output_gain = 10^(output_db / 20);

@sample
// Apply input gain
sample = spl0 * input_gain;

// Process
processed = do_something(sample);

// Apply output gain and makeup
spl0 = processed * output_gain;

4. Oversampling for Nonlinear Processing

Distortion and saturation benefit from oversampling to reduce aliasing:

@init
oversample = 2;  // 2x oversampling

@sample
// Upsample (simple interpolation)
s1 = spl0;
s2 = (spl0 + last_input) * 0.5;
last_input = spl0;

// Process at higher rate
s1 = saturate(s1);
s2 = saturate(s2);

// Downsample (simple averaging)
spl0 = (s1 + s2) * 0.5;

Quick Reference

Use Case	Rule
EQ / Filter	filters
Compressor / Limiter	dynamics
Delay / Echo	delays
Reverb	reverb
Distortion / Saturation	distortion
Chorus / Flanger / Phaser	modulation

Common Effect Patterns

Stereo Processing

@sample
// Process left and right identically
spl0 = process(spl0);
spl1 = process(spl1);

// Or use mid-side
mid = (spl0 + spl1) * 0.5;
side = (spl0 - spl1) * 0.5;
mid = process_mid(mid);
side = process_side(side);
spl0 = mid + side;
spl1 = mid - side;

Dry/Wet Mix

@slider
dry = 1 - wet_amount;
wet = wet_amount;

@sample
dry_signal = spl0;
wet_signal = process(spl0);
spl0 = dry_signal * dry + wet_signal * wet;

Bypass with Smoothing

@slider
bypass_target = bypass ? 0 : 1;

@sample
// Smooth bypass transition
bypass_smooth += 0.001 * (bypass_target - bypass_smooth);

dry_signal = spl0;
wet_signal = process(spl0);
spl0 = dry_signal * (1 - bypass_smooth) + wet_signal * bypass_smooth;