Audio Biquad EQ Calculator

Compute normalized biquad filter coefficients (b0, b1, b2, a1, a2) for peaking, low-shelf, and high-shelf EQ at any sample rate. Shows the frequency response curve.

Calculator Electronics Updated Apr 18, 2026
How to Use
  1. Pick filter type: peaking (bell EQ), low-shelf, high-shelf, low-pass, high-pass.
  2. Enter sample rate, center/cutoff frequency, gain in dB, and Q.
  3. Copy the normalized coefficients (b0, b1, b2, a1, a2) into your DSP code.
  4. The response plot shows magnitude in dB vs. frequency.
Input
Hz
Hz
dB
0.1–20
Presets
Frequency Response
b0
b1
b2
a1, a2

Coefficients

Enter values to compute coefficients.

Formulas (Robert Bristow-Johnson cookbook)

Difference Equation
y[n] = b0·x[n] + b1·x[n−1] + b2·x[n−2] − a1·y[n−1] − a2·y[n−2]
The direct-form-I biquad implementation.
Omega
ω₀ = 2π·fc / fs
Digital angular frequency.
Alpha
α = sin(ω₀) / (2Q)
Q-determined bandwidth factor.
Peaking b/a
A = √(10^(gain/40))
Gain scaling for EQ filters.
Normalization
Divide all by a0
Industry-standard: coefs are scaled so a0 = 1.
Stability
|poles| < 1
Required for a stable biquad; use standard Q ranges to guarantee.

About Biquad Filters

The biquad is the workhorse of digital audio processing. Robert Bristow-Johnson's "Audio EQ Cookbook" (1999) provided the standard formulas that nearly every EQ implementation uses — including Web Audio API, iOS Core Audio, and most DAW plugins. Five coefficients, one structure, decades of proven reliability.

Biquads are stable when poles are inside the unit circle. For Q values above ~20 at low frequencies, numerical precision matters — use double-precision floats or the transposed direct-form-II structure to avoid instability in 32-bit single precision. For most audio (20Hz-20kHz at 48kHz), direct-form-I in single precision is fine.

History of Digital Biquad Filters

The biquad (biquadratic) filter structure emerged from 1960s-70s digital signal processing research, when researchers at Bell Labs and elsewhere discovered that higher-order IIR filters could be factored into cascades of 2nd-order sections for better numerical stability. A 16th-order elliptic filter as a single monolithic difference equation is essentially unusable in fixed-point arithmetic; cascaded as 8 biquads, the same filter is robust and easily tunable.

Robert Bristow-Johnson's "Audio EQ Cookbook" (1999, circulated as a text file before any formal publication) gave the audio industry its lingua franca for digital EQ. Every plug-in developer, every DAW, every WebAudio implementation traces its biquad formulas back to this document. Before the cookbook, each company had its own slightly-different coefficient formulas, leading to subtle sonic differences when porting effects between platforms.

Today biquads run in the trillions of instances daily — every smartphone headphone EQ, every car audio tone control, every live-sound parametric rack unit is computing the same difference equation first formalized 60 years ago. WebAudio's BiquadFilterNode is a direct implementation of the cookbook; ARM's CMSIS-DSP and TI's C6000 DSP libraries both include optimized biquad kernels with bit-identical output.

About This Calculator

Pick filter type (peaking, low/high shelf, low/high pass, band-pass, notch), enter sample rate, center/cutoff frequency, gain in dB (for peaking and shelf), and Q. The tool applies Bristow-Johnson's cookbook formulas to compute b0, b1, b2, a1, a2 normalized with a0 = 1 — the exact same coefficient format used by WebAudio, iOS Core Audio, and most DSP libraries.

Copy the coefficients into your DSP code and implement with the direct-form-I difference equation: y[n] = b0·x[n] + b1·x[n-1] + b2·x[n-2] − a1·y[n-1] − a2·y[n-2]. For Q > 20 at low frequencies on embedded fixed-point DSPs, use transposed direct-form-II for better numerical stability. Everything runs client-side; no values leave your browser.

Frequently Asked Questions

What is a biquad filter?

A second-order IIR (infinite impulse response) filter, implemented as: y[n] = b0×x[n] + b1×x[n−1] + b2×x[n−2] − a1×y[n−1] − a2×y[n−2]. Five coefficients define its behavior. Audio DSP uses biquads as the basic building block for all EQ, filter, and shelf operations.

What is Q?

Q controls the bandwidth of peaking filters. Higher Q = narrower bump, lower Q = broader. For shelves, Q controls the slope transition sharpness. Common values: 0.707 (−3dB slope for shelves), 1.4 (musical bell), 5+ (notch/surgical).

What sample rate should I use?

Match your audio pipeline. CD/most consumer: 44.1 kHz. Pro: 48 kHz. Hi-res: 96 kHz or 192 kHz. Coefficients scale with sample rate — change the rate and recompute. Wrong sample rate at runtime produces detuned EQ.

What's the difference between peaking and shelf?

Peaking EQ boosts or cuts a narrow band around a center frequency — like a bell curve. Shelving boosts or cuts everything above (high-shelf) or below (low-shelf) a cutoff. Bass controls are usually low-shelf; treble is high-shelf. Parametric EQ mixes peaks in the mids.

Why are the coefficients normalized by a0?

The biquad difference equation has 6 coefficients; normalizing by a0 (dividing everything by it) reduces to 5 and simplifies the runtime math. All major DSP libraries (WebAudio, DSP textbooks, most embedded DSPs) use normalized forms.

Common Use Cases

Parametric EQ

Each band: one peaking biquad. Full 10-band EQ = 10 biquads in series. Tune center frequencies, Qs, and gains for any EQ curve.

Crossover Network

Low-pass + high-pass biquads create 2-way speaker crossover at a chosen frequency (Linkwitz-Riley: two cascaded Butterworths).

Tone Controls

Bass = low-shelf at ~100Hz. Treble = high-shelf at ~10kHz. Both with Q ≈ 0.707 for smooth response.

Notch / Anti-Hum

Very high Q peaking (Q > 10) with negative gain to remove 60Hz line hum or narrow-band noise.

Anti-Alias / DC Block

Low-pass biquad at fs/2 × 0.4 to prevent aliasing before downsampling; high-pass at ~20Hz to remove DC and sub-sonic content.

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