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A simple C++ library for reading and writing audio files
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A simple C++ library for reading and writing audio files.

Current supported formats:

  • WAV
  • AIFF


AudioFile is written and maintained by Adam Stark.


Create an AudioFile object:

#include "AudioFile.h"

AudioFile<double> audioFile;

Load an audio file:

audioFile.load ("/path/to/my/audiofile.wav");

Get some information about the loaded audio:

int sampleRate = audioFile.getSampleRate();
int bitDepth = audioFile.getBitDepth();

int numSamples = audioFile.getNumSamplesPerChannel();
double lengthInSeconds = audioFile.getLengthInSeconds();

int numChannels = audioFile.getNumChannels();
bool isMono = audioFile.isMono();
bool isStereo = audioFile.isStereo();

// or, just use this quick shortcut to print a summary to the console

Access the samples directly:

int channel = 0;
int numSamples = audioFile.getNumSamplesPerChannel();

for (int i = 0; i < numSamples; i++)
	double currentSample = audioFile.samples[channel][i];

Replace the AudioFile audio buffer with another

// 1. Create an AudioBuffer 
// (BTW, AudioBuffer is just a vector of vectors)

AudioFile<double>::AudioBuffer buffer;

// 2. Set to (e.g.) two channels
buffer.resize (2);

// 3. Set number of samples per channel
buffer[0].resize (100000);
buffer[1].resize (100000);

// 4. do something here to fill the buffer with samples, e.g.

#include <math.h> // somewhere earler (for M_PI and sinf())

// then...

int numChannels = 2;
int numSamplesPerChannel = 100000;
float sampleRate = 44100.f;
float frequency = 440.f;

for (int i = 0; i < numSamplesPerChannel; i++)
    float sample = sinf (2. * M_PI * ((float) i / sampleRate) * frequency) ;
    for (int channel = 0; channel < numChannels; channel++)
         buffer[channel][i] = sample * 0.5;

// 5. Put into the AudioFile object
bool ok = audioFile.setAudioBuffer (buffer);

Resize the audio buffer

// Set both the number of channels and number of samples per channel
audioFile.setAudioBufferSize (numChannels, numSamples);

// Set the number of samples per channel
audioFile.setNumSamplesPerChannel (numSamples);

// Set the number of channels
audioFile.setNumChannels (int numChannels);

Set bit depth and sample rate

audioFile.setBitDepth (24);
audioFile.setSampleRate (44100);

Save the audio file to disk

// Wave file (implicit) ("path/to/desired/audioFile.wav");

// Wave file (explicit) ("path/to/desired/audioFile.wav", AudioFileFormat::Wave);

// Aiff file ("path/to/desired/audioFile.aif", AudioFileFormat::Aiff);

A Note On Types

AudioFile is a template class and so it can be instantiated using floating point precision:

AudioFile<float> audioFile;

...or double precision:

AudioFile<double> audioFile;

This simply reflects the data type you would like to use to store the underlying audio samples. You can still read or write 8, 16 or 24-bit audio files, regardless of the type that you use (unless your system uses a precision for floats less than your desired bit depth).


1.0.3 - 28th October 2018
  • Bug fixes
  • Documentation updates
1.0.2 - 6th June 2017
  • Bug fixes


Copyright (c) 2017 Adam Stark

This program is free software: you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation, either version 3 of the License, or (at your option) any later version.

This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details.

You should have received a copy of the GNU General Public License along with this program. If not, see

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