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Recording audio

Recording to a sound buffer

The most common use for captured audio data is for it to be saved to a sound buffer (sf::SoundBuffer) so that it can either be played or saved to a file.

This can be achieved with the very simple interface of the sf::SoundBufferRecorder class:

// first check if an input audio device is available on the system
if (!sf::SoundBufferRecorder::isAvailable())
{
    // error: audio capture is not available on this system
    ...
}

// create the recorder
sf::SoundBufferRecorder recorder;

// start the capture
recorder.start();

// wait...

// stop the capture
recorder.stop();

// retrieve the buffer that contains the captured audio data
const sf::SoundBuffer& buffer = recorder.getBuffer();

The SoundBufferRecorder::isAvailable static function checks if audio recording is supported by the system. It if returns false, you won't be able to use the sf::SoundBufferRecorder class at all.

The start and stop functions are self-explanatory. The capture runs in its own thread, which means that you can do whatever you want between start and stop. After the end of the capture, the recorded audio data is available in a sound buffer that you can get with the getBuffer function.

With the recorded data, you can then:

  • Save it to a file

    buffer.saveToFile("my_record.ogg");
    
  • Play it directly

    sf::Sound sound(buffer);
    sound.play();
    
  • Access the raw audio data and analyze it, transform it, etc.

    const std::int16_t* samples = buffer.getSamples();
    std::size_t count = buffer.getSampleCount();
    doSomething(samples, count);
    

If you want to use the captured audio data after the recorder is destroyed or restarted, don't forget to make a copy of the buffer.

Selecting the input device

If you have multiple sound input devices connected to your computer (for example a microphone, a sound interface (external soundcard) or a webcam microphone) you can specify the device that is used for recording. A sound input device is identified by its name. A std::vector<std::string> containing the names of all connected devices is available through the static SoundBufferRecorder::getAvailableDevices() function. You can then select a device from the list for recording, by passing the chosen device name to the setDevice() method. It is even possible to change the device on the fly (i.e. while recording).

The name of the currently used device can be obtained by calling getDevice(). If you don't choose a device yourself, the default device will be used. Its name can be obtained through the static SoundBufferRecorder::getDefaultDevice() function.

Here is a small example of how to set the input device:

// get the available sound input device names
std::vector<std::string> availableDevices = sf::SoundRecorder::getAvailableDevices();

// choose a device
std::string inputDevice = availableDevices[0];

// create the recorder
sf::SoundBufferRecorder recorder;

// set the device
if (!recorder.setDevice(inputDevice))
{
    // error: device selection failed
    ...
}

// use recorder as usual

Custom recording

If storing the captured data in a sound buffer is not what you want, you can write your own recorder. Doing so will allow you to process the audio data while it is captured, (almost) directly from the recording device. This way you can, for example, stream the captured audio over the network, perform real-time analysis on it, etc.

To write your own recorder, you must inherit from the sf::SoundRecorder abstract base class. In fact, sf::SoundBufferRecorder is just a built-in specialization of this class.

You only have a single virtual function to override in your derived class: onProcessSamples. It is called every time a new chunk of audio samples is captured, so this is where you implement your specific stuff.

By default Audio samples are provided to the onProcessSamples method every 100 ms. You can change the interval by using the setProcessingInterval method. You may want to use a smaller interval if you want to process the recorded data in real time, for example. Note that this is only a hint and that the actual period may vary, so don't rely on it to implement precise timing.

There are also two additional virtual functions that you can optionally override: onStart and onStop. They are called when the capture starts/stops respectively. They are useful for initialization/cleanup tasks.

Here is the skeleton of a complete derived class:

class MyRecorder : public sf::SoundRecorder
{
    bool onStart() override // optional
    {
        // initialize whatever has to be done before the capture starts
        ...

        // return true to start the capture, or false to cancel it
        return true;
    }

    bool onProcessSamples(const std::int16_t* samples, std::size_t sampleCount) override
    {
        // do something useful with the new chunk of samples
        ...

        // return true to continue the capture, or false to stop it
        return true;
    }

    void onStop() override // optional
    {
        // clean up whatever has to be done after the capture is finished
        ...
    }
};

The isAvailable/start/stop functions are defined in the sf::SoundRecorder base, and thus inherited in every derived classes. This means that you can use any recorder class exactly the same way as the sf::SoundBufferRecorder class above.

if (!MyRecorder::isAvailable())
{
    // error...
}

MyRecorder recorder;
recorder.start();
...
recorder.stop();

Threading issues

Since recording is done in a separate thread, it is important to know what exactly happens, and where.

onStart will be called directly by the start function, so it is executed in the same thread that called it. However, onProcessSample and onStop will always be called from the internal recording thread that SFML creates.

If your recorder uses data that may be accessed concurrently in both the caller thread and in the recording thread, you have to protect it (with a mutex for example) in order to avoid concurrent access, which may cause undefined behavior -- corrupt data being recorded, crashes, etc.