- Qt 6.12
- Qt Multimedia
- C++ Classes
- QtMultimedia
- QAudioSink
- Qt 6.12.0 Reference Documentation

# QAudioSink Class

The QAudioSink class provides an interface for sending audio data to an audio output device. More...

Header: | `#include <QAudioSink>`
CMake: | `find_package(Qt6 REQUIRED COMPONENTS Multimedia)` `target_link_libraries(mytarget PRIVATE Qt6::Multimedia)`
qmake: | `QT += multimedia`
Inherits: | QObject

- List of all members, including inherited members

## Public Functions

| QAudioSink(const QAudioFormat &format = QAudioFormat(), QObject *parent = nullptr)
| QAudioSink(const QAudioDevice &audioDevice, const QAudioFormat &format = QAudioFormat(), QObject *parent = nullptr)
virtual | ~QAudioSink() override
`(since 6.10)` qsizetype | bufferFrameCount() const
qsizetype | bufferSize() const
qsizetype | bytesFree() const
qint64 | elapsedUSecs() const
QtAudio::Error | error() const
QAudioFormat | format() const
`(since 6.10)` qsizetype | framesFree() const
bool | isNull() const
`(since 6.12)` int | nativePeriodFrameCount() const
qint64 | processedUSecs() const
void | reset()
void | resume()
`(since 6.10)` void | setBufferFrameCount(qsizetype value)
void | setBufferSize(qsizetype value)
`(since 6.12)` void | setNativePeriodFrameCount(int frameCount)
void | setVolume(qreal volume)
QIODevice * | start()
`(since 6.11)` void | start(Callback &&cb)
void | start(QIODevice *device)
QtAudio::State | state() const
void | stop()
void | suspend()
qreal | volume() const

## Signals

void | stateChanged(QtAudio::State state)

## Detailed Description

You can construct an audio output with the system's default audio output device. It is also possible to create QAudioSink with a specific QAudioDevice. When you create the audio output, you should also send in the QAudioFormat to be used for the playback (see the QAudioFormat class description for details).

QAudioSink can be used in two different modes:

- Using a QIODevice from an application thread
- Using a callback-based interface from the audio thread

### QIODevice interface

Starting to play an audio stream is simply a matter of calling start() with a QIODevice. QAudioSink will then fetch the data it needs from the io device. So playing back an audio file is as simple as:

     QFile sourceFile;  // class member.
     QAudioSink* audio; // class member.
     {
         sourceFile.setFileName("/tmp/test.raw");
         sourceFile.open(QIODevice::ReadOnly);

         QAudioFormat format;
         // Set up the format, eg.
         format.setSampleRate(44100);
         format.setChannelCount(1);
         format.setSampleFormat(QAudioFormat::Int16);

         QAudioDevice info(QMediaDevices::defaultAudioOutput());
         if (!info.isFormatSupported(format)) {
             qWarning() << "Raw audio format not supported by backend, cannot play audio.";
             return;
         }

         audio = new QAudioSink(format, this);
         connect(audio, QAudioSink::stateChanged, this, &AudioInputExample::handleStateChanged);
         audio->start(&sourceFile);
     }

The file will start playing assuming that the audio system and output device support it. If you run out of luck, check what's up with the error() function.

After the file has finished playing, we need to stop the device:

     void AudioOutputExample::stopAudioOutput()
     {
         audio->stop();
         sourceFile.close();
         delete audio;
     }

At any given time, the QAudioSink will be in one of four states: active, suspended, stopped, or idle. These states are described by the QtAudio::State enum.

#### Threading model and buffering

The QIODevice interface is designed to be used from the application thread. A wait-free ringbuffer is used to communicate to the audio thread. The size of this ringbuffer can be configured with setBufferSize() and defaults to 250ms. The state of this buffer can be queried with bytesFree(). If the ringbuffer runs out of data, the audio thread will send silence to the audio device and the state will change to QtAudio::IdleState and resume to QtAudio::ActiveState when more data is available from the QIODevice.

### Callback interface

The preferred way to achieve low audio latency is to use the callback-based interface. It allows you to write audio data directly to the audio device without having to go through a QIODevice. This is done by calling start() with a callback function that will be called from the audio thread. This callback function will be called with a QSpan<SampleType> whenever the audio backend requires data.

     QAudioSink* audio; // class member.
     float phase;       // class member.
     {
         QAudioFormat format;
         // Set up the format, eg.
         format.setSampleRate(44100);
         format.setChannelCount(2);
         format.setSampleFormat(QAudioFormat::Float);

         QAudioDevice info(QMediaDevices::defaultAudioOutput());
         if (!info.isFormatSupported(format)) {
             qWarning() << "Raw audio format not supported by backend, cannot play audio.";
             return;
         }

         audio = new QAudioSink(format, this);
         float phaseIncrement = 2 * M_PI * 220.0 / format.sampleRate(); // 220 Hz sine wave
         audio->start([&phase, phaseIncrement] (QSpan<float> interleavedAudioBuffer) {
             // The audio callback should not call any functions that may potentially be blocking

             // Fill the audio buffer with a sine wave
             const int sampleCount = interleavedAudioBuffer.size() / 2; // Stereo, so divide by 2
             for (int i = 0; i < sampleCount; ++i) {
                 float sample = std::sin(phase);
                 interleavedAudioBuffer[i * 2] = sample;     // Left channel
                 interleavedAudioBuffer[i * 2 + 1] = sample; // Right channel
                 phase += phaseIncrement;                    // Increment phase for next sample
             }
         });

         if (!audio->error() == QtAudio::Error::NoError) {
             // in addition to the other start() signatures, starting the audio callback will fail if
             // * the backend does not implement callback-based IO (the API is available on all major
             //   platforms)
             // * the signature of the audio callback does not match format.sampleFormat()

             qWarning() << "Error starting audio output:" << audio->errorString();
         }
     }

Unlike the QIODevice-based interface, the QAudioSink can only be in the states active, suspendend and stopped. The setBufferSize() API is not available when using the callback, the size of the callback argument is determined by the audio backend.

Note: This API is only available on platforms that support the callback API: Apple's CoreAudio (macOS, iOS, etc), Windows, Linux (using the PulseAudio or PipeWire backend) and Android.

Note: The callback will be called on a soft-realtime audio thread. It is important to ensure that the callback does not block, as this can cause audio glitches or dropouts. This includes performing blocking IO, locking mutexes, allocating memories or any other operations that may block. For best practices consult Ross Bencina's article Real-time audio programming 101: time waits for nothing. Also consider using clang's Realtime sanitizer to validate the audio callback.

### State and error handling

State changes are reported through the stateChanged() signal. You can use this signal to, for instance, update the GUI of the application; the mundane example here being changing the state of a `play/pause` button. You request a state change directly with suspend(), stop(), reset(), resume(), and start().

The QAudioSink will enter the StoppedState when an error is encountered. The error type can be retrieved with the error() function. Please see the QtAudio::Error enum for a description of the possible errors that are reported. Calling stop() or reset() will reset the error state to NoError.

You can check for errors by connecting to the stateChanged() signal:

     void AudioOutputExample::handleStateChanged(QtAudio::State newState)
     {
         switch (newState) {
             case QtAudio::IdleState:
                 // Finished playing (no more data)
                 AudioOutputExample::stopAudioOutput();
                 break;

             case QtAudio::StoppedState:
                 // Stopped for other reasons
                 if (audio->error() != QtAudio::NoError) {
                     // Error handling
                 }
                 break;

             default:
                 // ... other cases as appropriate
                 break;
         }
     }

See also QAudioSource and QAudioDevice.

## Member Function Documentation

### `[explicit]` QAudioSink::QAudioSink(const QAudioFormat &format = QAudioFormat(), QObject *parent = nullptr)

Construct a new audio output and attach it to parent. The default audio output device is used with the output format parameters. If format is default-initialized, the format will be set to the preferred format of the audio device.

### `[explicit]` QAudioSink::QAudioSink(const QAudioDevice &audioDevice, const QAudioFormat &format = QAudioFormat(), QObject *parent = nullptr)

Construct a new audio output and attach it to parent. The device referenced by audioDevice is used with the output format parameters. If format is default-initialized, the format will be set to the preferred format of audioDevice.

### `[override virtual noexcept]` QAudioSink::~QAudioSink()

Destroys this audio output.

This will release any system resources used and free any buffers.

### `[since 6.10]` qsizetype QAudioSink::bufferFrameCount() const

Returns the audio buffer size in frames.

If called before start(), returns platform default value. If called before `start()` but setBufferSize() or setBufferFrameCount() was called prior, returns value set by `setBufferSize()` or `setBufferFrameCount()`. If called after `start()`, returns the actual buffer size being used. This may not be what was set previously by `setBufferSize()` or `setBufferFrameCount()`.

This function was introduced in Qt 6.10.

See also setBufferFrameCount() and bufferSize.

### qsizetype QAudioSink::bufferSize() const

Returns the audio buffer size in bytes.

If called before start(), returns platform default value. If called before `start()` but setBufferSize() or setBufferFrameCount() was called prior, returns value set by `setBufferSize()` or `setBufferFrameCount()`. If called after `start()`, returns the actual buffer size being used. This may not be what was set previously by `setBufferSize()` or `setBufferFrameCount()`.

See also setBufferSize() and bufferFrameCount.

### qsizetype QAudioSink::bytesFree() const

Returns the number of free bytes available in the audio buffer.

Note: The returned value is only valid while in QtAudio::ActiveState or QtAudio::IdleState state, otherwise returns zero.

See also framesFree.

### qint64 QAudioSink::elapsedUSecs() const

Returns the microseconds since start() was called, including time in Idle and Suspend states.

### QtAudio::Error QAudioSink::error() const

Returns the error state.

### QAudioFormat QAudioSink::format() const

Returns the QAudioFormat being used.

### `[since 6.10]` qsizetype QAudioSink::framesFree() const

Returns the number of free frames available in the audio buffer.

Note: The returned value is only valid while in QtAudio::ActiveState or QtAudio::IdleState state, otherwise returns zero.

This function was introduced in Qt 6.10.

See also bytesFree.

### bool QAudioSink::isNull() const

Returns `true` is the QAudioSink instance is `null`, otherwise returns `false`.

### `[since 6.12]` int QAudioSink::nativePeriodFrameCount() const

Returns the native period frame count.

Returns -1 if not set, otherwise returns the value previously set with setNativePeriodFrameCount().

This function was introduced in Qt 6.12.

See also setNativePeriodFrameCount.

### qint64 QAudioSink::processedUSecs() const

Returns the amount of audio data processed since start() was called (in microseconds).

### void QAudioSink::reset()

Immediately halts audio output and discards any audio data currently in the buffers. All pending audio data pushed to QIODevice is ignored.

See also stop().

### void QAudioSink::resume()

Resumes processing audio data after a suspend().

Sets state() to the state the sink had when suspend() was called. This function does nothing if the audio sink's state is not QtAudio::SuspendedState.

### `[since 6.10]` void QAudioSink::setBufferFrameCount(qsizetype value)

Sets the audio buffer size to value in frame count.

Note: This function can be called anytime before start(). Calls to this are ignored after start(). It should not be assumed that the buffer size set is the actual buffer size used - call bufferFrameCount() anytime after start() to return the actual buffer size being used.

The bufferFrameCount() and bufferSize() properties denote the ringbuffer size when using the QIODevice APIs. The native period size is controlled via setNativePeriodFrameCount().

This function was introduced in Qt 6.10.

See also bufferFrameCount() and setBufferSize.

### void QAudioSink::setBufferSize(qsizetype value)

Sets the audio buffer size to value in bytes.

Note: This function can be called anytime before start(). Calls to this are ignored after start(). It should not be assumed that the buffer size set is the actual buffer size used - call bufferSize() anytime after start() to return the actual buffer size being used.

The bufferFrameCount() and bufferSize() properties denote the ringbuffer size when using the QIODevice APIs. The native period size is controlled via setNativePeriodFrameCount().

See also bufferSize() and setBufferFrameCount.

### `[since 6.12]` void QAudioSink::setNativePeriodFrameCount(int frameCount)

Sets the native period frame count to frameCount.

Tunes the audio buffer size used by the operating system and hardware. By default (when not set), the system uses a platform-dependent value, typically 1024 frames (~23ms at 44100 Hz).

Use smaller values (64-256) to reduce latency, or larger values (2048-4096) to reduce the risk of buffer underruns (dropouts).

Valid values are powers of 2 between 32 and 4096 (inclusive), or -1 to unset. Can only be called when the audio sink is stopped.

Maps to platform-specific settings: `kAudioDevicePropertyBufferFrameSize` (macOS), `IAudioClient::bufferDuration` (Windows), `PW_KEY_NODE_FORCE_QUANTUM` (PipeWire/Linux), and `AAudioStreamBuilder_setBufferCapacityInFrames` (Android).

Note: This is separate from the ringbuffer configured with setBufferSize() or setBufferFrameCount().

This function was introduced in Qt 6.12.

See also nativePeriodFrameCount.

### void QAudioSink::setVolume(qreal volume)

Sets the output volume to volume.

The volume is scaled linearly from `0.0` (silence) to `1.0` (full volume). Values outside this range will be clamped.

The default volume is `1.0`.

Note: Adjustments to the volume will change the volume of this audio stream, not the global volume.

UI volume controls should usually be scaled non-linearly. For example, using a logarithmic scale will produce linear changes in perceived loudness, which is what a user would normally expect from a volume control. See QtAudio::convertVolume() for more details.

See also volume().

### QIODevice *QAudioSink::start()

Returns a pointer to the internal QIODevice being used to transfer data to the system's audio output. The device will already be open and write() can write data directly to it.

Note: The pointer will become invalid after the stream is stopped or if you start another stream.

If the QAudioSink is able to access the system's audio device, state() returns QtAudio::IdleState, error() returns QtAudio::NoError and the stateChanged() signal is emitted.

If a problem occurs during this process, error() returns QtAudio::OpenError, state() returns QtAudio::StoppedState and the stateChanged() signal is emitted.

See also QIODevice and QIODevice interface.

### `[since 6.11]` template <typename Callback> requires QtAudio::if_audio_sink_callback<Callback> void QAudioSink::start(Callback &&cb)

Starts the QAudioSink with a callback function that will be called on a soft-realtime audio thread. The callback is a callable that takes a QSpan<SampleType> as an argument, SampleType has to match the QAudioFormat::SampleFormat of the QAudioSink's format. The span needs to be filled with interleaved audio data.

If the QAudioSink is able to successfully start, error() returns QtAudio::NoError.

If a problem occurs during this process, error() returns QtAudio::OpenError, state() returns QtAudio::StoppedState and the stateChanged() signal is emitted.

Note: This API is only available on platforms that support the callback API: Apple's CoreAudio (macOS, iOS, etc), Windows, Linux (using the PulseAudio or PipeWire backend) and Android.

Note: The callback will be called on a soft-realtime audio thread. It is important to ensure that the callback does not block, as this can cause audio glitches or dropouts. This includes performing blocking IO, locking mutexes, allocating memories or any other operations that may block. For best practices consult Ross Bencina's article Real-time audio programming 101: time waits for nothing. Also consider using clang's Realtime sanitizer to validate the audio callback.

This function was introduced in Qt 6.11.

See also Callback interface.

### void QAudioSink::start(QIODevice *device)

Starts transferring audio data from the device to the system's audio output. The device must have been opened in the ReadOnly or ReadWrite modes.

If the QAudioSink is able to successfully output audio data, state() returns QtAudio::ActiveState, error() returns QtAudio::NoError and the stateChanged() signal is emitted.

If a problem occurs during this process, error() returns QtAudio::OpenError, state() returns QtAudio::StoppedState and the stateChanged() signal is emitted.

See also QIODevice and QIODevice interface.

### QtAudio::State QAudioSink::state() const

Returns the state of audio processing.

### `[signal]` void QAudioSink::stateChanged(QtAudio::State state)

This signal is emitted when the device state has changed. This is the current state of the audio output.

Note: The QtAudio namespace was named QAudio up to and including Qt 6.6. String-based connections to this signal have to use `QAudio::State` as the parameter type: `connect(source, SIGNAL(stateChanged(QAudio::State)), ...);`

### void QAudioSink::stop()

Stops the audio output, detaching from the system resource.

Sets error() to QtAudio::NoError, state() to QtAudio::StoppedState and emit stateChanged() signal.

Note: On Linux, and Darwin, this operation synchronously drains the underlying audio buffer, which may cause delays accordingly to the buffer payload. To reset all the buffers immediately, use the method reset instead.

See also reset().

### void QAudioSink::suspend()

Stops processing audio data, preserving buffered audio data.

Sets state() to QtAudio::SuspendedState and emits stateChanged() signal.

### qreal QAudioSink::volume() const

Returns the volume between 0.0 and 1.0 inclusive.

See also setVolume().