PySide6.QtMultimedia.QAudioSink¶
- class QAudioSink¶
The
QAudioSinkclass provides an interface for sending audio data to an audio output device. More…Synopsis¶
Methods¶
def
__init__()def
bufferSize()def
bytesFree()def
elapsedUSecs()def
error()def
format()def
framesFree()def
isNull()def
processedUSecs()def
reset()def
resume()def
setBufferSize()def
setVolume()def
start()def
state()def
stop()def
suspend()def
volume()
Signals¶
def
stateChanged()
Note
This documentation may contain snippets that were automatically translated from C++ to Python. We always welcome contributions to the snippet translation. If you see an issue with the translation, you can also let us know by creating a ticket on https:/bugreports.qt.io/projects/PYSIDE
Detailed Description¶
Warning
This section contains snippets that were automatically translated from C++ to Python and may contain errors.
You can construct an audio output with the system’s default audio output device. It is also possible to create
QAudioSinkwith a specificQAudioDevice. When you create the audio output, you should also send in theQAudioFormatto be used for the playback (see theQAudioFormatclass description for details).QAudioSinkcan be used in two different modes:Using a QIODevice from an application thread
Using a callback-based interface from the audio thread
Starting to play an audio stream is simply a matter of calling
start()with a QIODevice.QAudioSinkwill 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.OpenModeFlag.ReadOnly) format = QAudioFormat() # Set up the format, eg. format.setSampleRate(44100) format.setChannelCount(1) format.setSampleFormat(QAudioFormat.Int16) info = QAudioDevice(QMediaDevices.defaultAudioOutput()) if not info.isFormatSupported(format): qWarning() << "Raw audio format not supported by backend, cannot play audio." return audio = QAudioSink(format, self) audio.connect(QAudioSink::stateChanged, self.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:
def stopAudioOutput(self): audio.stop() sourceFile.close() del audio
At any given time, the
QAudioSinkwill be in one of four states: active, suspended, stopped, or idle. These states are described by theStateenum.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 withbytesFree(). If the ringbuffer runs out of data, the audio thread will send silence to the audio device and the state will change toIdleStateand resume toActiveStatewhen more data is available from the QIODevice.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. format = QAudioFormat() # Set up the format, eg. format.setSampleRate(44100) format.setChannelCount(2) format.setSampleFormat(QAudioFormat.Float) info = QAudioDevice(QMediaDevices.defaultAudioOutput()) if not info.isFormatSupported(format): qWarning() << "Raw audio format not supported by backend, cannot play audio." return audio = QAudioSink(format, self) 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 sampleCount = interleavedAudioBuffer.size() / 2 # Stereo, so divide by 2 for i in range(0, sampleCount): 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 not 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
QAudioSinkcan only be in the states active, suspendend and stopped. ThesetBufferSize()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 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 aplay/pausebutton. You request a state change directly withsuspend(),stop(),reset(),resume(), andstart().The
QAudioSinkwill enter theStoppedStatewhen an error is encountered. Theerror typecan be retrieved with theerror()function. Please see theErrorenum for a description of the possible errors that are reported. Callingstop()orreset()will reset the error state toNoError.You can check for errors by connecting to the
stateChanged()signal:def handleStateChanged(self, newState): match newState: case QtAudio.IdleState: # Finished playing (no more data) AudioOutputExample::stopAudioOutput() case QtAudio.StoppedState: # Stopped for other reasons if audio.error() != QtAudio.NoError: # Error handling case _: # ... other cases as appropriate
See also
- __init__([format=QAudioFormat()[, parent=None]])¶
- Parameters:
format –
QAudioFormatparent –
QObject
Construct a new audio output and attach it to
parent. The default audio output device is used with the outputformatparameters. Ifformatis default-initialized, the format will be set to the preferred format of the audio device.- __init__(audioDeviceInfo[, format=QAudioFormat()[, parent=None]])
- Parameters:
audioDeviceInfo –
QAudioDeviceformat –
QAudioFormatparent –
QObject
Construct a new audio output and attach it to
parent. The device referenced byaudioDeviceis used with the outputformatparameters. Ifformatis default-initialized, the format will be set to the preferred format ofaudioDevice.- bufferFrameCount()¶
- Return type:
int
Returns the audio buffer size in frames.
If called before
start(), returns platform default value. If called beforestart()butsetBufferSize()orsetBufferFrameCount()was called prior, returns value set bysetBufferSize()orsetBufferFrameCount(). If called afterstart(), returns the actual buffer size being used. This may not be what was set previously bysetBufferSize()orsetBufferFrameCount().See also
- bufferSize()¶
- Return type:
int
Returns the audio buffer size in bytes.
If called before
start(), returns platform default value. If called beforestart()butsetBufferSize()orsetBufferFrameCount()was called prior, returns value set bysetBufferSize()orsetBufferFrameCount(). If called afterstart(), returns the actual buffer size being used. This may not be what was set previously bysetBufferSize()orsetBufferFrameCount().See also
- bytesFree()¶
- Return type:
int
Returns the number of free bytes available in the audio buffer.
Note
The returned value is only valid while in
ActiveStateorIdleStatestate, otherwise returns zero.See also
- elapsedUSecs()¶
- Return type:
int
Returns the microseconds since
start()was called, including time in Idle and Suspend states.Returns the error state.
- format()¶
- Return type:
Returns the
QAudioFormatbeing used.- framesFree()¶
- Return type:
int
Returns the number of free frames available in the audio buffer.
Note
The returned value is only valid while in
ActiveStateorIdleStatestate, otherwise returns zero.See also
- isNull()¶
- Return type:
bool
Returns
trueis theQAudioSinkinstance isnull, otherwise returnsfalse.- processedUSecs()¶
- Return type:
int
Returns the amount of audio data processed since
start()was called (in microseconds).- 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
- resume()¶
Resumes processing audio data after a
suspend().Sets
state()to the state the sink had whensuspend()was called, and setserror()to QAudioError::NoError. This function does nothing if the audio sink’s state is notSuspendedState.- setBufferFrameCount(framesCount)¶
- Parameters:
framesCount – int
Sets the audio buffer size to
valuein frame count.Note
This function can be called anytime before
start(). Calls to this are ignored afterstart(). It should not be assumed that the buffer size set is the actual buffer size used - callbufferFrameCount()anytime afterstart()to return the actual buffer size being used.See also
- setBufferSize(bytes)¶
- Parameters:
bytes – int
Sets the audio buffer size to
valuein bytes.Note
This function can be called anytime before
start(). Calls to this are ignored afterstart(). It should not be assumed that the buffer size set is the actual buffer size used - callbufferSize()anytime afterstart()to return the actual buffer size being used.See also
- setVolume(volume)¶
- Parameters:
volume – float
Sets the output volume to
volume.The volume is scaled linearly from
0.0(silence) to1.0(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
convertVolume()for more details.See also
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
QAudioSinkis able to access the system’s audio device,state()returnsIdleState,error()returnsNoErrorand thestateChanged()signal is emitted.If a problem occurs during this process,
error()returnsOpenError,state()returnsStoppedStateand thestateChanged()signal is emitted.See also
QIODevice interface- start(device)
- Parameters:
device –
QIODevice
Starts transferring audio data from the
deviceto the system’s audio output. Thedevicemust have been opened in the ReadOnly or ReadWrite modes.If the
QAudioSinkis able to successfully output audio data,state()returnsActiveState,error()returnsNoErrorand thestateChanged()signal is emitted.If a problem occurs during this process,
error()returnsOpenError,state()returnsStoppedStateand thestateChanged()signal is emitted.See also
QIODevice interfaceReturns the state of audio processing.
This signal is emitted when the device
statehas 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::Stateas the parameter type:connect(source, SIGNAL(stateChanged(QAudio::State)), ...);- stop()¶
Stops the audio output, detaching from the system resource.
Sets
error()toNoError,state()toStoppedStateand emitstateChanged()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
resetinstead.See also
- suspend()¶
Stops processing audio data, preserving buffered audio data.
Sets
error()toNoError,state()toSuspendedStateand emitsstateChanged()signal.- volume()¶
- Return type:
float
Returns the volume between 0.0 and 1.0 inclusive.
See also