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C library providing cross-platform audio input and output. The API is suitable for real-time software such as digital audio workstations as well as consumer software such as music players.

This library is an abstraction; however in the delicate balance between performance and power, and API convenience, the scale is tipped closer to the former. Features that only exist in some sound backends are exposed.

Features and Limitations

  • Supported operating systems:
    • Windows 7+
    • MacOS 10.10+
    • Linux 3.7+
  • Supported backends:
  • Exposes both raw devices and shared devices. Raw devices give you the best performance but prevent other applications from using them. Shared devices are default and usually provide sample rate conversion and format conversion.
  • Exposes both device id and friendly name. id you could save in a config file because it persists between devices becoming plugged and unplugged, while friendly name is suitable for exposing to users.
  • Supports optimal usage of each supported backend. The same API does the right thing whether the backend has a fixed buffer size, such as on JACK and CoreAudio, or whether it allows directly managing the buffer, such as on ALSA, PulseAudio, and WASAPI.
  • C library. Depends only on the respective backend API libraries and libc. Does not depend on libstdc++, and does not have exceptions, run-time type information, or setjmp.
  • Errors are communicated via return codes, not logging to stdio.
  • Supports channel layouts (also known as channel maps), important for surround sound applications.
  • Ability to monitor devices and get an event when available devices change.
  • Ability to get an event when the backend is disconnected, for example when the JACK server or PulseAudio server shuts down.
  • Detects which input device is default and which output device is default.
  • Ability to connect to multiple backends at once. For example you could have an ALSA device open and a JACK device open at the same time.
  • Meticulously checks all return codes and memory allocations and uses meaningful error codes.
  • Exposes extra API that is only available on some backends. For example you can provide application name and stream names which is used by JACK and PulseAudio.


Complete program to emit a sine wave over the default device using the best backend:

#include <soundio/soundio.h>

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>

static const float PI = 3.1415926535f;
static float seconds_offset = 0.0f;
static void write_callback(struct SoundIoOutStream *outstream,
        int frame_count_min, int frame_count_max)
    const struct SoundIoChannelLayout *layout = &outstream->layout;
    float float_sample_rate = outstream->sample_rate;
    float seconds_per_frame = 1.0f / float_sample_rate;
    struct SoundIoChannelArea *areas;
    int frames_left = frame_count_max;
    int err;

    while (frames_left > 0) {
        int frame_count = frames_left;

        if ((err = soundio_outstream_begin_write(outstream, &areas, &frame_count))) {
            fprintf(stderr, "%s\n", soundio_strerror(err));

        if (!frame_count)

        float pitch = 440.0f;
        float radians_per_second = pitch * 2.0f * PI;
        for (int frame = 0; frame < frame_count; frame += 1) {
            float sample = sinf((seconds_offset + frame * seconds_per_frame) * radians_per_second);
            for (int channel = 0; channel < layout->channel_count; channel += 1) {
                float *ptr = (float*)(areas[channel].ptr + areas[channel].step * frame);
                *ptr = sample;
        seconds_offset = fmodf(seconds_offset +
            seconds_per_frame * frame_count, 1.0f);

        if ((err = soundio_outstream_end_write(outstream))) {
            fprintf(stderr, "%s\n", soundio_strerror(err));

        frames_left -= frame_count;

int main(int argc, char **argv) {
    int err;
    struct SoundIo *soundio = soundio_create();
    if (!soundio) {
        fprintf(stderr, "out of memory\n");
        return 1;

    if ((err = soundio_connect(soundio))) {
        fprintf(stderr, "error connecting: %s", soundio_strerror(err));
        return 1;


    int default_out_device_index = soundio_default_output_device_index(soundio);
    if (default_out_device_index < 0) {
        fprintf(stderr, "no output device found");
        return 1;

    struct SoundIoDevice *device = soundio_get_output_device(soundio, default_out_device_index);
    if (!device) {
        fprintf(stderr, "out of memory");
        return 1;

    fprintf(stderr, "Output device: %s\n", device->name);

    struct SoundIoOutStream *outstream = soundio_outstream_create(device);
    outstream->format = SoundIoFormatFloat32NE;
    outstream->write_callback = write_callback;

    if ((err = soundio_outstream_open(outstream))) {
        fprintf(stderr, "unable to open device: %s", soundio_strerror(err));
        return 1;

    if (outstream->layout_error)
        fprintf(stderr, "unable to set channel layout: %s\n", soundio_strerror(outstream->layout_error));

    if ((err = soundio_outstream_start(outstream))) {
        fprintf(stderr, "unable to start device: %s", soundio_strerror(err));
        return 1;

    for (;;)

    return 0;

Backend Priority

When you use soundio_connect, libsoundio tries these backends in order. If unable to connect to that backend, due to the backend not being installed, or the server not running, or the platform is wrong, the next backend is tried.

  1. JACK
  2. PulseAudio
  3. ALSA (Linux)
  4. CoreAudio (OSX)
  5. WASAPI (Windows)
  6. Dummy

If you don't like this order, you can use soundio_connect_backend to explicitly choose a backend to connect to. You can use soundio_backend_count and soundio_get_backend to get the list of available backends.

API Documentation


Install the dependencies:

  • cmake
  • ALSA library (optional)
  • libjack2 (optional)
  • libpulseaudio (optional)
mkdir build
cd build
cmake ..
sudo make install

Building for Windows

You can build libsoundio with mxe. Follow the requirements section to install the packages necessary on your system. Then somewhere on your file system:

git clone
cd mxe
make MXE_TARGETS='x86_64-w64-mingw32.static i686-w64-mingw32.static' gcc

Then in the libsoundio source directory (replace "/path/to/mxe" with the appropriate path):

mkdir build-win32
cd build-win32
cmake .. -DCMAKE_TOOLCHAIN_FILE=/path/to/mxe/usr/i686-w64-mingw32.static/share/cmake/mxe-conf.cmake
mkdir build-win64
cd build-win64
cmake .. -DCMAKE_TOOLCHAIN_FILE=/path/to/mxe/usr/x86_64-w64-mingw32.static/share/cmake/mxe-conf.cmake


For each backend, do the following:

  1. Run the unit tests: ./unit_tests. To see test coverage, install lcov, run make coverage, and then view coverage/index.html in a browser.
  2. Run the example ./sio_list_devices and make sure it does not crash, and the output looks good. If valgrind is available, use it.
  3. Run ./sio_list_devices --watch and make sure it detects when you plug and unplug a USB microphone.
  4. Run ./sio_sine and make sure you hear a sine wave. For backends with raw devices, run ./sio_sine --device id --raw (where 'id' is a device id you got from sio_list_devices and make sure you hear a sine wave.
    • Use 'p' to test pausing, 'u' to test unpausing, 'q' to test cleanup.
    • 'c' for clear buffer. Clear buffer should not pause the stream and it should also not cause an underflow.
    • Use 'P' to test pausing from the callback, and then 'u' to unpause.
  5. Run ./underflow and read the testing instructions that it prints.
  6. Run ./sio_microphone and ensure that it is both recording and playing back correctly. If possible use the --in-device and --out-device parameters to test a USB microphone in raw mode.
  7. Run ./backend_disconnect_recover and read the testing instructions that it prints.
  8. Run ./latency and make sure the printed beeps line up with the beeps that you hear.

Building the Documentation

Ensure that doxygen is installed, then:

make doc

Then look at html/index.html in a browser.


C library for cross-platform real-time audio input and output







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