resample converts a mono audio file from one sample rate to another using the miniDSP polyphase sinc resampler. It reads any format supported by libsndfile (WAV, FLAC, AIFF, OGG, etc.) and writes WAV output (IEEE float).

Build

From the repository root:

make tools

This compiles tools/resample/resample (requires FFTW3 and libsndfile).

Usage

resample [-z N] [-b F] <input> <rate> <output.wav>

Argument	Description
<input>	Input audio file (must be mono)
<rate>	Target sample rate in Hz (positive integer)
<output.wav>	Output file path (WAV format only)

Options

Flag	Default	What it does
-z N	32	Number of sinc zero-crossings per side. More zero-crossings produce a sharper cutoff and better stopband rejection at the cost of speed.
-b F	10.0	Kaiser window beta parameter. Higher values widen the mainlobe but deepen stopband attenuation (10.0 gives >100 dB rejection).

The defaults work well for most use cases. Bump them up (e.g. -z 64 -b 14.0) for mastering-quality conversion where you want even cleaner anti-aliasing.

Examples

Upsample from 16 kHz to 48 kHz:

./tools/resample/resample speech.wav 48000 speech_48k.wav

Downsample to 8 kHz (telephone quality):

./tools/resample/resample recording.wav 8000 recording_8k.wav

The resampler automatically applies an anti-aliasing lowpass filter when downsampling — no separate filtering step is needed.

High-quality conversion with custom parameters:

./tools/resample/resample -z 64 -b 14.0 master.wav 96000 master_96k.wav

How it works

WAV/FLAC/... --> FIO_read_audio() --> float*
                                        |
                              float-to-double conversion
                                        |
                              MD_resample() (polyphase sinc)
                                        |
                              double-to-float conversion
                                        |
                              FIO_write_wav() --> output.wav

The tool reads the input file via libsndfile, converts the samples to double precision for the DSP pipeline, resamples using MD_resample(), converts back to float, and writes the output as IEEE float WAV.

Buffers are freed eagerly after each stage to keep peak memory at roughly 2x the larger of the input or output signal.

Same-rate detection: if the input is already at the target rate, the tool copies the audio directly and prints a note — no resampling overhead.

Constraints:

Input must be mono (single-channel). Use sox to split multi-channel files.
Output is always WAV format. A warning is printed if the output filename does not end in .wav.

The resampling core

The core algorithm is a 512-phase polyphase sinc interpolation filter with Kaiser windowing. For a detailed explanation of the math, including the Bessel function, normalised sinc, and filter design, see the Sample Rate Conversion tutorial.

    /* Resample */
    unsigned out_len = MD_resample_output_len(
        (unsigned)in_len, (double)in_rate, (double)target_rate);
 
    double *out_double = malloc(out_len * sizeof(double));
    if (out_double == NULL) {
        fprintf(stderr, "Error: failed to allocate memory\n");
        free(in_double);
        return 1;
    }
 
    MD_resample(in_double, (unsigned)in_len,
                out_double, out_len,
                (double)in_rate, (double)target_rate,
                zero_crossings, kaiser_beta);