A small C library of DSP (Digital Signal Processing) routines for audio applications.
Read the full documentation -- API reference, tutorials, and interactive examples.
- GCC-PHAT -- estimate the time delay between two microphone signals using Generalized Cross-Correlation with Phase Transform. This is the core of acoustic source localisation.
- Magnitude spectrum -- compute |X(k)| from a real signal using the FFT; the foundation of frequency-domain analysis.
- Power spectral density -- compute |X(k)|^2 / N (periodogram); shows how signal power distributes across frequencies.
- Phase spectrum -- compute arg(X(k)) in radians; reveals the timing of each frequency component and is a prerequisite for phase-vocoder effects.
- Spectrogram (STFT) -- sliding-window FFT producing a time-frequency magnitude matrix; the standard tool for visualising time-varying audio.
- Signal measurements -- energy, power, power in dB, normalised entropy.
- Scaling & AGC -- linear range mapping, automatic gain control.
- Hanning window -- smooth windowing function for FFT analysis.
- Sine wave generator -- pure tone at a given frequency and amplitude; the "hello world" of DSP.
Seven classic audio filter types, all based on Robert Bristow-Johnson's Audio EQ Cookbook:
- Low-pass, High-pass, Band-pass, Notch
- Peaking EQ, Low shelf, High shelf
- Read audio files in any format supported by libsndfile (WAV, FLAC, AIFF, OGG, etc.)
- Write audio to WAV (IEEE float for lossless DSP round-trips)
- Write feature vectors in NumPy
.npyformat (for Python interop) - Write feature vectors in safetensors format (for ML pipelines)
- Write feature vectors in HTK binary format (deprecated)
- Record from the microphone and play back to speakers via PortAudio
- Non-blocking API with callback support
Install the following libraries before building:
| Library | Purpose | Debian/Ubuntu | macOS (Homebrew) |
|---|---|---|---|
| FFTW3 | Fast Fourier Transform | apt install libfftw3-dev |
brew install fftw |
| PortAudio | Live audio I/O | apt install portaudio19-dev |
brew install portaudio |
| libsndfile | Audio file reading | apt install libsndfile1-dev |
brew install libsndfile |
| Doxygen | API docs generation (optional) | apt install doxygen |
brew install doxygen |
| Apple container | Linux container testing (optional) | — | macOS 26+ built-in |
The Makefiles auto-detect Homebrew paths on macOS (both Apple Silicon and Intel).
On Ubuntu, GCC 14 or later is required for -std=c23 support. Ubuntu 24.04 ships GCC 13 by default, so install gcc-14 explicitly (apt install gcc-14).
make # builds libminidsp.amake test # builds and runs all testsTo verify the library builds and passes all tests on Linux (Ubuntu 24.04 with GCC 14):
make container-test # builds image, then runs make test inside the containerThis requires the Apple container CLI on macOS 26+.
make docs # generates HTML docs in docs/htmlA pre-push hook is included that runs make test and make container-test before allowing pushes to main:
make install-hooks#include "minidsp.h"
/* Two 4096-sample signals captured by spatially separated microphones */
double mic_a[4096], mic_b[4096];
/* Estimate the delay in samples (+/- 50 sample search window) */
int delay = MD_get_delay(mic_a, mic_b, 4096, NULL, 50, PHAT);
printf("Signal B is %d samples behind signal A\n", delay);
/* Clean up FFTW resources when done */
MD_shutdown();#include "minidsp.h"
double signal[1024];
// ... fill signal with audio samples ...
unsigned num_bins = 1024 / 2 + 1; /* 513 unique frequency bins */
double *mag = malloc(num_bins * sizeof(double));
MD_magnitude_spectrum(signal, 1024, mag);
/* mag[k] = |X(k)|, where frequency = k * sample_rate / 1024 */
free(mag);
MD_shutdown();A full example with Hanning windowing is in examples/magnitude_spectrum.c.
Run it to generate an interactive HTML plot (Plotly.js + D3.js):
make -C examples plot
open examples/magnitude_spectrum.html # interactive: zoom, pan, hover for valuesFor a step-by-step walkthrough of the DSP concepts, see the Magnitude Spectrum tutorial.
#include "minidsp.h"
double signal[1024];
// ... fill signal with audio samples ...
unsigned num_bins = 1024 / 2 + 1; /* 513 unique frequency bins */
double *psd = malloc(num_bins * sizeof(double));
MD_power_spectral_density(signal, 1024, psd);
/* psd[k] = |X(k)|^2 / N (power at frequency k * sample_rate / 1024) */
free(psd);
MD_shutdown();A full example with Hanning windowing and one-sided PSD conversion is in
examples/power_spectral_density.c.
See the PSD tutorial
for a detailed explanation.
#include "minidsp.h"
double signal[32000];
// ... fill signal with 2 s of audio at 16 kHz ...
unsigned N = 512; /* 32 ms window */
unsigned hop = 128; /* 8 ms hop (75% overlap) */
unsigned num_frames = MD_stft_num_frames(32000, N, hop); /* 247 */
unsigned num_bins = N / 2 + 1; /* 257 */
double *mag = malloc(num_frames * num_bins * sizeof(double));
MD_stft(signal, 32000, N, hop, mag);
/* mag[f * num_bins + k] = |X_f(k)|
* Time of frame f: time_s = (double)(f * hop) / 16000.0
* Frequency of bin k: freq_hz = (double)k * 16000.0 / N */
free(mag);
MD_shutdown();A full example generating an interactive HTML heatmap is in
examples/spectrogram.c.
See the Spectrogram tutorial
for a step-by-step explanation.
#include "biquad.h"
/* Create a 1 kHz low-pass filter at 44.1 kHz sample rate, 1-octave bandwidth */
biquad *lpf = BiQuad_new(LPF, 0.0, 1000.0, 44100.0, 1.0);
/* Process each audio sample */
for (int i = 0; i < num_samples; i++) {
output[i] = BiQuad(input[i], lpf);
}
free(lpf);The test suite (tests/test_minidsp.c) covers every public function:
- Dot product -- orthogonal vectors, known values, self-dot
- Energy / Power / dB -- known signals, sine wave power, dB floor
- Scaling -- endpoints, midpoint, vector scaling, fit-within-range
- AGC -- target dB level achievement
- Entropy -- uniform, spike, zero, clip/no-clip modes
- Hanning window -- endpoints, peak, symmetry, range
- Magnitude spectrum -- single sine, two sines, DC signal, zeros, impulse (flat spectrum), Parseval's theorem, FFT plan re-caching, non-negativity
- Power spectral density -- single sine, two sines, DC signal, zeros, impulse (flat PSD), Parseval's theorem, FFT plan re-caching, non-negativity
- Spectrogram (STFT) -- frame count formula, silence, pure tone peak, hop=N non-overlapping frames, non-negativity, Parseval's theorem per frame, plan re-caching across window sizes
- GCC-PHAT -- positive/negative/zero delays, SIMP vs PHAT weighting, multi-signal delays, FFT plan caching
- Biquad filters -- LPF, HPF, BPF, Notch, PEQ, Low shelf, High shelf, DC rejection
- File I/O writers -- .npy round-trip, safetensors round-trip, WAV round-trip
See TODO.md for planned features -- FFT spectrum analysis, signal generators, FIR filters, window functions, simple effects, pitch detection, and more.