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https://github.com/andrewrk/libsoundio

C library for cross-platform real-time audio input and output
https://github.com/andrewrk/libsoundio

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

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# libsoundio



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:

   - [JACK](http://jackaudio.org/)

   - [PulseAudio](http://www.freedesktop.org/wiki/Software/PulseAudio/)

   - [ALSA](http://www.alsa-project.org/)

   - [CoreAudio](https://developer.apple.com/library/mac/documentation/MusicAudio/Conceptual/CoreAudioOverview/Introduction/Introduction.html)

   - [WASAPI](https://msdn.microsoft.com/en-us/library/windows/desktop/dd371455%28v=vs.85%29.aspx)

   - Dummy (silence)

 * 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](http://latentcontent.net/2007/12/05/libpng-worst-api-ever/).

 * 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.



## Synopsis



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

backend:



```c

#include 



#include 

#include 

#include 

#include 



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));

            exit(1);

        }



        if (!frame_count)

            break;



        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));

            exit(1);

        }



        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;

    }



    soundio_flush_events(soundio);



    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 (;;)

        soundio_wait_events(soundio);



    soundio_outstream_destroy(outstream);

    soundio_device_unref(device);

    soundio_destroy(soundio);

    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.



 0. JACK

 0. PulseAudio

 0. ALSA (Linux)

 0. CoreAudio (OSX)

 0. WASAPI (Windows)

 0. 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](http://libsound.io/doc)



### Building



Install the dependencies:



 * cmake

 * ALSA library (optional)

 * libjack2 (optional)

 * libpulseaudio (optional)



```

mkdir build

cd build

cmake ..

make

sudo make install

```



### Building for Windows



You can build libsoundio with [mxe](http://mxe.cc/). Follow the

[requirements](http://mxe.cc/#requirements) section to install the

packages necessary on your system. Then somewhere on your file system:



```

git clone https://github.com/mxe/mxe

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

make

```



```

mkdir build-win64

cd build-win64

cmake .. -DCMAKE_TOOLCHAIN_FILE=/path/to/mxe/usr/x86_64-w64-mingw32.static/share/cmake/mxe-conf.cmake

make

```



### Testing



For each backend, do the following:



 0. Run the unit tests: `./unit_tests`. To see test coverage, install lcov, run

   `make coverage`, and then view `coverage/index.html` in a browser.

 0. Run the example `./sio_list_devices` and make sure it does not crash, and

    the output looks good. If valgrind is available, use it.

 0. Run `./sio_list_devices --watch` and make sure it detects when you plug and

    unplug a USB microphone.

 0. 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.

 0. Run `./underflow` and read the testing instructions that it prints.

 0. 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.

 0. Run `./backend_disconnect_recover` and read the testing instructions that

    it prints.

 0. Run `./latency` and make sure the printed beeps line up with the beeps that

    you hear.



### Building the Documentation



Ensure that [doxygen](http://www.stack.nl/~dimitri/doxygen/) is installed,

then:



```

make doc

```



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