# libsoundio C99 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. The goal of this library is to be the only resource needed to implement top quality audio playback and capture on desktop and laptop systems. This includes detailed documentation explaining how audio works on each supported backend, how they are abstracted to provide the libsoundio API, and what assumptions you can and cannot make in order to guarantee consistent, reliable behavior on every platform. **This project is a work-in-progress.** ## Features and Limitations * 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) - Dummy (silence) - (planned) [WASAPI](https://msdn.microsoft.com/en-us/library/windows/desktop/dd371455%28v=vs.85%29.aspx) - (planned) [ASIO](http://www.asio4all.com/) * 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 or PulseAudio. * 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. ## Synopsis Complete program to emit a sine wave over the default device using the best backend: ```c #include #include #include #include #include #include __attribute__ ((cold)) __attribute__ ((noreturn)) __attribute__ ((format (printf, 1, 2))) static void panic(const char *format, ...) { va_list ap; va_start(ap, format); vfprintf(stderr, format, ap); fprintf(stderr, "\n"); va_end(ap); abort(); } 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))) panic("%s", soundio_strerror(err)); 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 += seconds_per_frame * frame_count; if ((err = soundio_outstream_end_write(outstream))) panic("%s", soundio_strerror(err)); frames_left -= frame_count; } } int main(int argc, char **argv) { int err; struct SoundIo *soundio = soundio_create(); if (!soundio) panic("out of memory"); if ((err = soundio_connect(soundio))) panic("error connecting: %s", soundio_strerror(err)); soundio_flush_events(soundio); int default_out_device_index = soundio_default_output_device_index(soundio); if (default_out_device_index < 0) panic("no output device found"); struct SoundIoDevice *device = soundio_get_output_device(soundio, default_out_device_index); if (!device) panic("out of memory"); 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))) panic("unable to open device: %s", soundio_strerror(err)); if (outstream->layout_error) fprintf(stderr, "unable to set channel layout: %s\n", soundio_strerror(outstream->layout_error)); if ((err = soundio_outstream_start(outstream))) panic("unable to start device: %s", soundio_strerror(err)); 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. ASIO (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. For complete API documentation, see `src/soundio.h`. ## Contributing libsoundio is programmed in a tiny subset of C++11: * No STL. * No `new` or `delete`. * No `class`. All fields in structs are `public`. * No constructors or destructors. * No exceptions or run-time type information. * No references. * No linking against libstdc++. Do not be fooled - this is a *C library*, not a C++ library. We just take advantage of a select few C++11 compiler features such as templates, and then link against libc. ### 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 gcc ``` Then in the libsoundio source directory (replace "/path/to/mxe" with the appropriate path): ``` mkdir build-win cd build-win cmake .. -DCMAKE_TOOLCHAIN_FILE=/path/to/mxe/usr/i686-w64-mingw32.static/share/cmake/mxe-conf.cmake make ``` #### Running the Tests ``` make test ``` For more detailed output: ``` make ./unit_tests ``` To see test coverage, install lcov, run `make coverage` and then view `coverage/index.html` in a browser. ## Roadmap 0. implement WASAPI (Windows) backend, get examples working 0. implement ASIO (Windows) backend, get examples working 0. Make sure PulseAudio can handle refresh devices crashing before block_until_have_devices 0. Do we really want `period_duration` in the API? 0. Integrate into libgroove and test with Groove Basin 0. clear buffer maybe could take an argument to say how many frames to not clear 0. Verify that JACK xrun callback context is the same as process callback. If not, might need to hav xrun callback set a flag and have process callback call the underflow callback. 0. Create a test for pausing and resuming input and output streams. 0. Create a test for the latency / synchronization API. - Input is an audio file and some events indexed at particular frame - when listening the events should line up exactly with a beat or visual indicator, even when the latency is large. - Play the audio file, have the user press an input right at the beat. Find out what the frame index it thinks the user pressed it at and make sure that is correct. 0. Create a test for input stream overflow handling. 0. Allow calling functions from outside the callbacks as long as they first call lock and then unlock when done. 0. Should pause/resume be callable from outside the callbacks? 0. clean up API and improve documentation - make sure every function which can return an error documents which errors it can return 0. use a documentation generator and host the docs somewhere 0. -fvisibility=hidden and then explicitly export stuff, or explicitly make the unexported stuff private 0. add len arguments to APIs that have char * - replace strdup with `soundio_str_dupe` 0. Support PulseAudio proplist properties for main context and streams 0. Expose JACK options in `jack_client_open` 0. custom allocator support 0. mlock memory which is accessed in the real time path 0. make rtprio warning a callback and have existing behavior be the default callback 0. write detailed docs on buffer underflows explaining when they occur, what state changes are related to them, and how to recover from them. 0. Consider testing on FreeBSD 0. In ALSA do we need to wake up the poll when destroying the in or out stream? 0. Detect PulseAudio server going offline and emit `on_backend_disconnect`. ## Planned Uses for libsoundio * [Genesis](https://github.com/andrewrk/genesis) * [libgroove](https://github.com/andrewrk/libgroove) ([Groove Basin](https://github.com/andrewrk/groovebasin))