Web Audio API

Introduction

Audio on the web has been fairly primitive up to this point and until very recently has had to be delivered through plugins such as Flash and QuickTime. The introduction of the audio element in HTML5 is very important, allowing for basic streaming audio playback. But, it is not powerful enough to handle more complex audio applications. For sophisticated web-based games or interactive applications, another solution is required. It is a goal of this specification to include the capabilities found in modern game audio engines as well as some of the mixing, processing, and filtering tasks that are found in modern desktop audio production applications.

The APIs have been designed with a wide variety of use cases [webaudio-usecases] in mind. Ideally, it should be able to support any use case which could reasonably be implemented with an optimized C++ engine controlled via script and run in a browser. That said, modern desktop audio software can have very advanced capabilities, some of which would be difficult or impossible to build with this system. Apple’s Logic Audio is one such application which has support for external MIDI controllers, arbitrary plugin audio effects and synthesizers, highly optimized direct-to-disk audio file reading/writing, tightly integrated time-stretching, and so on. Nevertheless, the proposed system will be quite capable of supporting a large range of reasonably complex games and interactive applications, including musical ones. And it can be a very good complement to the more advanced graphics features offered by WebGL. The API has been designed so that more advanced capabilities can be added at a later time.

Features

The API supports these primary features:

Modular routing for simple or complex mixing/effect architectures, including multiple sends and submixes.
High dynamic range, using 32bits floats for internal processing.
Sample-accurate scheduled sound playback with low latency for musical applications requiring a very high degree of rhythmic precision such as drum machines and sequencers. This also includes the possibility of dynamic creation of effects.
Automation of audio parameters for envelopes, fade-ins / fade-outs, granular effects, filter sweeps, LFOs etc.
Flexible handling of channels in an audio stream, allowing them to be split and merged.
Processing of audio sources from an audio or video media element.
Processing live audio input using a MediaStream from getUserMedia().
Integration with WebRTC
- Processing audio received from a remote peer using a MediaStreamTrackAudioSourceNode and [webrtc].
- Sending a generated or processed audio stream to a remote peer using a MediaStreamAudioDestinationNode and [webrtc].
Audio stream synthesis and processing directly using scripts.
Spatialized audio supporting a wide range of 3D games and immersive environments:
- Panning models: equalpower, HRTF, pass-through
- Distance Attenuation
- Sound Cones
- Obstruction / Occlusion
- Source / Listener based
A convolution engine for a wide range of linear effects, especially very high-quality room effects. Here are some examples of possible effects:
- Small / large room
- Cathedral
- Concert hall
- Cave
- Tunnel
- Hallway
- Forest
- Amphitheater
- Sound of a distant room through a doorway
- Extreme filters
- Strange backwards effects
- Extreme comb filter effects
Dynamics compression for overall control and sweetening of the mix
Efficient real-time time-domain and frequency analysis / music visualizer support.
Efficient biquad filters for lowpass, highpass, and other common filters.
A Waveshaping effect for distortion and other non-linear effects
Oscillators

Modular Routing

Modular routing allows arbitrary connections between different AudioNode objects. Each node can have inputs and/or outputs. A source node has no inputs and a single output. A destination node has one input and no outputs. Other nodes such as filters can be placed between the source and destination nodes. The developer doesn’t have to worry about low-level stream format details when two objects are connected together; the right thing just happens. For example, if a mono audio stream is connected to a stereo input it should just mix to left and right channels appropriately.

In the simplest case, a single source can be routed directly to the output. All routing occurs within an AudioContext containing a single AudioDestinationNode:

Illustrating this simple routing, here’s a simple example playing a single sound:

var context = new AudioContext();

function playSound() {
  var source = context.createBufferSource();
  source.buffer = dogBarkingBuffer;
  source.connect(context.destination);
  source.start(0);
}

Here’s a more complex example with three sources and a convolution reverb send with a dynamics compressor at the final output stage:

modular routing2 — A more complex example of modular routing.

var context = 0;var compressor = 0;var reverb = 0;var source1 = 0;var source2 = 0;var source3 = 0;var lowpassFilter = 0;var waveShaper = 0;var panner = 0;var dry1 = 0;var dry2 = 0;var dry3 = 0;var wet1 = 0;var wet2 = 0;var wet3 = 0;var masterDry = 0;var masterWet = 0;function setupRoutingGraph () {  context = new AudioContext();  // Create the effects nodes.  lowpassFilter = context.createBiquadFilter();  waveShaper = context.createWaveShaper();  panner = context.createPanner();  compressor = context.createDynamicsCompressor();  reverb = context.createConvolver();  // Create master wet and dry.  masterDry = context.createGain();  masterWet = context.createGain();  // Connect final compressor to final destination.  compressor.connect(context.destination);  // Connect master dry and wet to compressor.  masterDry.connect(compressor);  masterWet.connect(compressor);  // Connect reverb to master wet.  reverb.connect(masterWet);  // Create a few sources.  source1 = context.createBufferSource();  source2 = context.createBufferSource();  source3 = context.createOscillator();  source1.buffer = manTalkingBuffer;  source2.buffer = footstepsBuffer;  source3.frequency.value = 440;  // Connect source1  dry1 = context.createGain();  wet1 = context.createGain();  source1.connect(lowpassFilter);  lowpassFilter.connect(dry1);  lowpassFilter.connect(wet1);  dry1.connect(masterDry);  wet1.connect(reverb);  // Connect source2  dry2 = context.createGain();  wet2 = context.createGain();  source2.connect(waveShaper);  waveShaper.connect(dry2);  waveShaper.connect(wet2);  dry2.connect(masterDry);  wet2.connect(reverb);  // Connect source3  dry3 = context.createGain();  wet3 = context.createGain();  source3.connect(panner);  panner.connect(dry3);  panner.connect(wet3);  dry3.connect(masterDry);  wet3.connect(reverb);  // Start the sources now.  source1.start(0);  source2.start(0);  source3.start(0);}

Modular routing also permits the output of AudioNodes to be routed to an AudioParam parameter that controls the behavior of a different AudioNode. In this scenario, the output of a node can act as a modulation signal rather than an input signal.

modular routing3 — Modular routing illustrating one Oscillator modulating the frequency of another.

function setupRoutingGraph() {  var context = new AudioContext();  // Create the low frequency oscillator that supplies the modulation signal  var lfo = context.createOscillator();  lfo.frequency.value = 1.0;  // Create the high frequency oscillator to be modulated  var hfo = context.createOscillator();  hfo.frequency.value = 440.0;  // Create a gain node whose gain determines the amplitude of the modulation signal  var modulationGain = context.createGain();  modulationGain.gain.value = 50;  // Configure the graph and start the oscillators  lfo.connect(modulationGain);  modulationGain.connect(hfo.detune);  hfo.connect(context.destination);  hfo.start(0);  lfo.start(0);}

API Overview

The interfaces defined are:

An AudioContext interface, which contains an audio signal graph representing connections between AudioNodes.
An AudioNode interface, which represents audio sources, audio outputs, and intermediate processing modules. AudioNodes can be dynamically connected together in a modular fashion. AudioNodes exist in the context of an AudioContext.
An AnalyserNode interface, an AudioNode for use with music visualizers, or other visualization applications.
An AudioBuffer interface, for working with memory-resident audio assets. These can represent one-shot sounds, or longer audio clips.
An AudioBufferSourceNode interface, an AudioNode which generates audio from an AudioBuffer.
An AudioDestinationNode interface, an AudioNode subclass representing the final destination for all rendered audio.
An AudioParam interface, for controlling an individual aspect of an AudioNode's functioning, such as volume.
An AudioListener interface, which works with a PannerNode for spatialization.
An AudioWorklet interface representing a factory for creating custom nodes that can process audio directly using scripts.
An AudioWorkletGlobalScope interface, the context in which AudioWorkletProcessor processing scripts run.
An AudioWorkletNode interface, an AudioNode representing a node processed in an AudioWorkletProcessor.
An AudioWorkletProcessor interface, representing a single node instance inside an audio worker.
A BiquadFilterNode interface, an AudioNode for common low-order filters such as:
- Low Pass
- High Pass
- Band Pass
- Low Shelf
- High Shelf
- Peaking
- Notch
- Allpass
A ChannelMergerNode interface, an AudioNode for combining channels from multiple audio streams into a single audio stream.
A ChannelSplitterNode interface, an AudioNode for accessing the individual channels of an audio stream in the routing graph.
A ConstantSourceNode interface, an AudioNode for generating a nominally constant output value with an AudioParam to allow automation of the value.
A ConvolverNode interface, an AudioNode for applying a real-time linear effect (such as the sound of a concert hall).
A DelayNode interface, an AudioNode which applies a dynamically adjustable variable delay.
A DynamicsCompressorNode interface, an AudioNode for dynamics compression.
A GainNode interface, an AudioNode for explicit gain control. Because inputs to AudioNodes support multiple connections (as a unity-gain summing junction), mixers can be easily built with GainNodes.
An IIRFilterNode interface, an AudioNode for a general IIR filter.
A MediaElementAudioSourceNode interface, an AudioNode which is the audio source from an audio, video, or other media element.
A MediaStreamAudioSourceNode interface, an AudioNode which is the audio source from a MediaStream such as live audio input, or from a remote peer.
A MediaStreamTrackAudioSourceNode interface, an AudioNode which is the audio source from a MediaStreamTrack.
A MediaStreamAudioDestinationNode interface, an AudioNode which is the audio destination to a MediaStream sent to a remote peer.
A PannerNode interface, an AudioNode for spatializing / positioning audio in 3D space.
A PeriodicWave interface for specifying custom periodic waveforms for use by the OscillatorNode.
An OscillatorNode interface, an AudioNode for generating a periodic waveform.
A StereoPannerNode interface, an AudioNode for equal-power positioning of audio input in a stereo stream.
A WaveShaperNode interface, an AudioNode which applies a non-linear waveshaping effect for distortion and other more subtle warming effects.

There are also several features that have been deprecated from the Web Audio API but not yet removed, pending implementation experience of their replacements:

A ScriptProcessorNode interface, an AudioNode for generating or processing audio directly using scripts.
An AudioProcessingEvent interface, which is an event type used with ScriptProcessorNode objects.

1. The Audio API

1.1. The `BaseAudioContext` Interface

This interface represents a set of AudioNode objects and their connections. It allows for arbitrary routing of signals to an AudioDestinationNode. Nodes are created from the context and are then connected together.

BaseAudioContext is not instantiated directly, but is instead extended by the concrete interfaces AudioContext (for real-time rendering) and OfflineAudioContext (for offline rendering).

enum AudioContextState {
  "suspended",
  "running",
  "closed"
};

Enumeration description
"`suspended`"	This context is currently suspended (context time is not proceeding, audio hardware may be powered down/released).
"`running`"	Audio is being processed.
"`closed`"	This context has been released, and can no longer be used to process audio. All system audio resources have been released.

callback DecodeErrorCallback = void (DOMException error);

callback DecodeSuccessCallback = void (AudioBuffer decodedData);

[Exposed=Window]
interface BaseAudioContext : EventTarget {
  readonly attribute AudioDestinationNode destination;
  readonly attribute float sampleRate;
  readonly attribute double currentTime;
  readonly attribute AudioListener listener;
  readonly attribute AudioContextState state;
  [SameObject, SecureContext]
  readonly attribute AudioWorklet audioWorklet;
  attribute EventHandler onstatechange;

  AnalyserNode createAnalyser ();
  BiquadFilterNode createBiquadFilter ();
  AudioBuffer createBuffer (unsigned long numberOfChannels, unsigned long length, float sampleRate);
  AudioBufferSourceNode createBufferSource ();
  ChannelMergerNode createChannelMerger (optional unsigned long numberOfInputs = 6);
  ChannelSplitterNode createChannelSplitter (optional unsigned long numberOfOutputs = 6);
  ConstantSourceNode createConstantSource ();
  ConvolverNode createConvolver ();
  DelayNode createDelay (optional double maxDelayTime = 1.0);
  DynamicsCompressorNode createDynamicsCompressor ();
  GainNode createGain ();
  IIRFilterNode createIIRFilter (sequence<double> feedforward, sequence<double> feedback);
  OscillatorNode createOscillator ();
  PannerNode createPanner ();
  PeriodicWave createPeriodicWave (sequence<float> real, sequence<float> imag, optional PeriodicWaveConstraints constraints);
  ScriptProcessorNode createScriptProcessor(optional unsigned long bufferSize = 0,
                                            optional unsigned long numberOfInputChannels = 2,
                                            optional unsigned long numberOfOutputChannels = 2);
  StereoPannerNode createStereoPanner ();
  WaveShaperNode createWaveShaper ();

  Promise<AudioBuffer> decodeAudioData (ArrayBuffer audioData,
                                        optional DecodeSuccessCallback? successCallback,
                                        optional DecodeErrorCallback? errorCallback);
  Promise<void> resume ();
};

1.1.1. Attributes

audioWorklet, of type AudioWorklet, readonly

Allows access to the Worklet object that can import a script containing AudioWorkletProcessor class definitions via the algorithms defined by [worklets-1] and AudioWorklet.

currentTime, of type double, readonly

This is the time in seconds of the sample frame immediately following the last sample-frame in the block of audio most recently processed by the context’s rendering graph. If the context’s rendering graph has not yet processed a block of audio, then currentTime has a value of zero.

In the time coordinate system of currentTime, the value of zero corresponds to the first sample-frame in the first block processed by the graph. Elapsed time in this system corresponds to elapsed time in the audio stream generated by the BaseAudioContext, which may not be synchronized with other clocks in the system. (For an OfflineAudioContext, since the stream is not being actively played by any device, there is not even an approximation to real time.)

All scheduled times in the Web Audio API are relative to the value of currentTime.

When the BaseAudioContext is in the "running" state, the value of this attribute is monotonically increasing and is updated by the rendering thread in uniform increments, corresponding to one render quantum. Thus, for a running context, currentTime increases steadily as the system processes audio blocks, and always represents the time of the start of the next audio block to be processed. It is also the earliest possible time when any change scheduled in the current state might take effect.

currentTime MUST be read atomically on the control thread before being returned.

destination, of type AudioDestinationNode, readonly

An AudioDestinationNode with a single input representing the final destination for all audio. Usually this will represent the actual audio hardware. All AudioNodes actively rendering audio will directly or indirectly connect to destination.

listener, of type AudioListener, readonly

An AudioListener which is used for 3D spatialization.

onstatechange, of type EventHandler

A property used to set the EventHandler for an event that is dispatched to BaseAudioContext when the state of the AudioContext has changed (i.e. when the corresponding promise would have resolved). An event of type Event will be dispatched to the event handler, which can query the AudioContext’s state directly. A newly-created AudioContext will always begin in the suspended state, and a state change event will be fired whenever the state changes to a different state. This event is fired before the oncomplete event is fired.

sampleRate, of type float, readonly

The sample rate (in sample-frames per second) at which the BaseAudioContext handles audio. It is assumed that all AudioNodes in the context run at this rate. In making this assumption, sample-rate converters or "varispeed" processors are not supported in real-time processing. The Nyquist frequency is half this sample-rate value.

state, of type AudioContextState, readonly

Describes the current state of the AudioContext, on the control thread.

1.1.2. Methods

createAnalyser()

Factory method for an AnalyserNode.

No parameters.

Return type: AnalyserNode

createBiquadFilter()

Factory method for a BiquadFilterNode representing a second order filter which can be configured as one of several common filter types.

No parameters.

Return type: BiquadFilterNode

createBuffer(numberOfChannels, length, sampleRate)

Creates an AudioBuffer of the given size. The audio data in the buffer will be zero-initialized (silent). A NotSupportedError exception MUST be thrown if any of the arguments is negative, zero, or outside its nominal range.

Arguments for the BaseAudioContext.createBuffer() method.
Parameter	Type	Nullable	Optional	Description
`numberOfChannels`	unsigned long	✘	✘	Determines how many channels the buffer will have. An implementation MUST support at least 32 channels.
`length`	unsigned long	✘	✘	Determines the size of the buffer in sample-frames. This MUST be at least 1.
`sampleRate`	float	✘	✘	Describes the sample-rate of the linear PCM audio data in the buffer in sample-frames per second. An implementation MUST support sample rates in at least the range 8000 to 96000.

Return type: AudioBuffer

createBufferSource()

Factory method for a AudioBufferSourceNode.

No parameters.

Return type: AudioBufferSourceNode

createChannelMerger(numberOfInputs)

Factory method for a ChannelMergerNode representing a channel merger. An IndexSizeError exception MUST be thrown if numberOfInputs is less than 1 or is greater than the number of supported channels.

Arguments for the BaseAudioContext.createChannelMerger(numberOfInputs) method.
Parameter	Type	Nullable	Optional	Description
`numberOfInputs`	unsigned long	✘	✔	Determines the number of inputs. Values of up to 32 MUST be supported. If not specified, then `6` will be used.

Return type: ChannelMergerNode

createChannelSplitter(numberOfOutputs)

Factory method for a ChannelSplitterNode representing a channel splitter. An IndexSizeError exception MUST be thrown if numberOfOutputs is less than 1 or is greater than the number of supported channels.

Arguments for the BaseAudioContext.createChannelSplitter(numberOfOutputs) method.
Parameter	Type	Nullable	Optional	Description
`numberOfOutputs`	unsigned long	✘	✔	The number of outputs. Values of up to 32 MUST be supported. If not specified, then `6` will be used.

Return type: ChannelSplitterNode

createConstantSource()

Factory method for a ConstantSourceNode.

No parameters.

Return type: ConstantSourceNode

createConvolver()

Factory method for a ConvolverNode.

No parameters.

Return type: ConvolverNode

createDelay(maxDelayTime)

Factory method for a DelayNode. The initial default delay time will be 0 seconds.

Arguments for the BaseAudioContext.createDelay(maxDelayTime) method.
Parameter	Type	Nullable	Optional	Description
`maxDelayTime`	double	✘	✔	Specifies the maximum delay time in seconds allowed for the delay line. If specified, this value MUST be greater than zero and less than three minutes or a `NotSupportedError` exception MUST be thrown. If not specified, then `1` will be used.

Return type: DelayNode

createDynamicsCompressor()

Factory method for a DynamicsCompressorNode.

No parameters.

Return type: DynamicsCompressorNode

createGain()

Factory method for GainNode.

No parameters.

Return type: GainNode

createIIRFilter(feedforward, feedback)

Arguments for the BaseAudioContext.createIIRFilter() method.
Parameter	Type	Nullable	Optional	Description
`feedforward`	sequence<double>	✘	✘	An array of the feedforward (numerator) coefficients for the transfer function of the IIR filter. The maximum length of this array is 20. If all of the values are zero, an `InvalidStateError` MUST be thrown. A `NotSupportedError` MUST be thrown if the array length is 0 or greater than 20.
`feedback`	sequence<double>	✘	✘	An array of the feedback (denominator) coefficients for the transfer function of the IIR filter. The maximum length of this array is 20. If the first element of the array is 0, an `InvalidStateError` MUST be thrown. A `NotSupportedError` MUST be thrown if the array length is 0 or greater than 20.

Return type: IIRFilterNode

createOscillator()

Factory method for an OscillatorNode.

No parameters.

Return type: OscillatorNode

createPanner()

Factory method for a PannerNode.

No parameters.

Return type: PannerNode

createPeriodicWave(real, imag, constraints)

Factory method to create a PeriodicWave.

When calling this method, execute these steps:

If real and imag are not of the same length, an IndexSizeError MUST be thrown.
Let o be a new object of type PeriodicWaveOptions.
Respectively set the real and imag parameters passed to this factory method to the attributes of the same name on o.
Set the disableNormalization attribute on o to the value of the disableNormalization attribute of the constraints attribute passed to the factory method.
Construct a new PeriodicWave p, passing the BaseAudioContext this factory method has been called on as a first argument, and o.
Return p.

Arguments for the BaseAudioContext.createPeriodicWave() method.
Parameter	Type	Nullable	Optional	Description
`real`	sequence<float>	✘	✘	A sequence of cosine parameters. See its `real` constructor argument for a more detailed description.
`imag`	sequence<float>	✘	✘	A sequence of sine parameters. See its `imag` constructor argument for a more detailed description.
`constraints`	PeriodicWaveConstraints	✘	✔	If not given, the waveform is normalized. Otherwise, the waveform is normalized according the value given by `constraints`.

Return type: PeriodicWave

createScriptProcessor(bufferSize, numberOfInputChannels, numberOfOutputChannels)

Factory method for a ScriptProcessorNode. This method is DEPRECATED, as it is intended to be replaced by AudioWorkletNode. Creates a ScriptProcessorNode for direct audio processing using scripts. An IndexSizeError exception MUST be thrown if bufferSize or numberOfInputChannels or numberOfOutputChannels are outside the valid range.

It is invalid for both numberOfInputChannels and numberOfOutputChannels to be zero. In this case an IndexSizeError MUST be thrown.

Arguments for the BaseAudioContext.createScriptProcessor(bufferSize, numberOfInputChannels, numberOfOutputChannels) method.
Parameter	Type	Nullable	Optional	Description
`bufferSize`	unsigned long	✘	✔	The `bufferSize` parameter determines the buffer size in units of sample-frames. If it’s not passed in, or if the value is 0, then the implementation will choose the best buffer size for the given environment, which will be constant power of 2 throughout the lifetime of the node. Otherwise if the author explicitly specifies the bufferSize, it MUST be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. This value controls how frequently the `onaudioprocess` event is dispatched and how many sample-frames need to be processed each call. Lower values for `bufferSize` will result in a lower (better) latency. Higher values will be necessary to avoid audio breakup and glitches. It is recommended for authors to not specify this buffer size and allow the implementation to pick a good buffer size to balance between latency and audio quality. If the value of this parameter is not one of the allowed power-of-2 values listed above, an `IndexSizeError` MUST be thrown.
`numberOfInputChannels`	unsigned long	✘	✔	This parameter determines the number of channels for this node’s input. Values of up to 32 must be supported. A `NotSupportedError` must be thrown if the number of channels is not supported.
`numberOfOutputChannels`	unsigned long	✘	✔	This parameter determines the number of channels for this node’s output. Values of up to 32 must be supported. A `NotSupportedError` must be thrown if the number of channels is not supported.

Return type: ScriptProcessorNode

createStereoPanner()

Factory method for a StereoPannerNode.

No parameters.

Return type: StereoPannerNode

createWaveShaper()

Factory method for a WaveShaperNode representing a non-linear distortion.

No parameters.

Return type: WaveShaperNode

decodeAudioData(audioData, successCallback, errorCallback)

Asynchronously decodes the audio file data contained in the ArrayBuffer. The ArrayBuffer can, for example, be loaded from an XMLHttpRequest’s response attribute after setting the responseType to "arraybuffer". Audio file data can be in any of the formats supported by the audio element. The buffer passed to decodeAudioData() has its content-type determined by sniffing, as described in [mimesniff].

Although the primary method of interfacing with this function is via its promise return value, the callback parameters are provided for legacy reasons. The system shall ensure that the AudioContext is not garbage collected before the promise is resolved or rejected and any callback function is called and completes.

When decodeAudioData is called, the following steps MUST be performed on the control thread:

Let promise be a new promise.
If the operation IsDetachedBuffer (described in [ECMASCRIPT]) on audioData is false, execute the following steps:
1. Detach the audioData ArrayBuffer. This operation is described in [ECMASCRIPT].
2. Queue a decoding operation to be performed on another thread.
Else, execute the following steps:
1. Let error be a DataCloneError.
2. Reject promise with error.
3. Queue a task to invoke errorCallback with error.
Return promise.

When queuing a decoding operation to be performed on another thread, the following steps MUST happen on a thread that is not the control thread nor the rendering thread, called the decoding thread.

Note: Multiple decoding threads can run in parallel to service multiple calls to decodeAudioData.

Attempt to decode the encoded audioData into linear PCM.
If a decoding error is encountered due to the audio format not being recognized or supported, or because of corrupted/unexpected/inconsistent data, then queue a task to execute the following step, on the control thread’s event loop:
1. Let error be a DOMException whose name is EncodingError.
2. Reject promise with error.
3. If errorCallback is not missing, invoke errorCallback with error.
Otherwise:
1. Take the result, representing the decoded linear PCM audio data, and resample it to the sample-rate of the AudioContext if it is different from the sample-rate of audioData.
2. Queue a task on the control thread’s event loop to execute the following steps:
  
  1.Let buffer be an AudioBuffer containing the final result (after possibly sample-rate conversion).
  1. Resolve promise with buffer.
  2. If successCallback is not missing, invoke successCallback with buffer.

Arguments for the BaseAudioContext.decodeAudioData() method.
Parameter	Type	Nullable	Optional	Description
`audioData`	ArrayBuffer	✘	✘	An ArrayBuffer containing compressed audio data.
`successCallback`	DecodeSuccessCallback?	✔	✔	A callback function which will be invoked when the decoding is finished. The single argument to this callback is an AudioBuffer representing the decoded PCM audio data.
`errorCallback`	DecodeErrorCallback?	✔	✔	A callback function which will be invoked if there is an error decoding the audio file.

Return type: Promise<AudioBuffer>

resume()

Resumes the progression of the BaseAudioContext's currentTime when it has been suspended.

When resume is called, execute these steps:

Let promise be a new Promise.
If the control thread state flag on the BaseAudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
If the BaseAudioContext is not allowed to start, append promise to pendingResumePromises and abort these steps, returning promise.
Set the control thread state flag on the BaseAudioContext to running.
Queue a control message to resume the BaseAudioContext.
Return promise.

Running a control message to resume an BaseAudioContext means running these steps on the rendering thread:

Attempt to acquire system resources.
Set the rendering thread state flag on the BaseAudioContext to running.
Start rendering the audio graph.
In case of failure, queue a task on the control thread to execute the following, and abort these steps:
1. Reject all promises from pendingResumePromises in order, then clear pendingResumePromises.
2. Reject promise.
Queue a task on the control thread’s event loop, to execute these steps:
1. Resolve all promises from pendingResumePromises in order, then clear pendingResumePromises.
2. Resolve promise.
3. If the state attribute of the BaseAudioContext is not already "running":
  1. Set the state attribute of the BaseAudioContext to "running".
  2. Queue a task to fire a simple event named statechange at the BaseAudioContext.

No parameters.

Return type: Promise<void>

1.1.3. Callback `DecodeSuccessCallback()` Parameters

decodedData, of type AudioBuffer: The AudioBuffer containing the decoded audio data.

1.1.4. Callback `DecodeErrorCallback()` Parameters

error, of type DOMException: The error that occurred while decoding.

1.1.5. Lifetime

Once created, an AudioContext will continue to play sound until it has no more sound to play, or the page goes away.

1.1.6. Lack of Introspection or Serialization Primitives

The Web Audio API takes a fire-and-forget approach to audio source scheduling. That is, source nodes are created for each note during the lifetime of the AudioContext, and never explicitly removed from the graph. This is incompatible with a serialization API, since there is no stable set of nodes that could be serialized.

Moreover, having an introspection API would allow content script to be able to observe garbage collections.

1.1.7. System Resources Associated with `BaseAudioContext` Subclasses

The subclasses AudioContext and OfflineAudioContext should be considered expensive objects. Creating these objects may involve creating a high-priority thread, or using a low-latency system audio stream, both having an impact on energy consumption. It is usually not necessary to create more than one AudioContext in a document.

Constructing or resuming a BaseAudioContext subclass involves acquiring system resources for that context. For AudioContext, this also requires creation of a system audio stream. These operations return when the context begins generating output from its associated audio graph.

Additionally, a user-agent can have an implementation-defined maximum number of AudioContexts, after which any attempt to create a new AudioContext will fail, throwing NotSupportedError.

suspend and close allow authors to release system resources, including threads, processes and audio streams. Suspending a BaseAudioContext permits implementations to release some of its resources, and allows it to continue to operate later by invoking resume. Closing an AudioContext permits implementations to release all of its resources, after which it cannot be used or resumed again.

Note: For example, this can involve waiting for the audio callbacks to fire regularly, or to wait for the hardware to be ready for processing.

1.2. The `AudioContext` Interface

This interface represents an audio graph whose AudioDestinationNode is routed to a real-time output device that produces a signal directed at the user. In most use cases, only a single AudioContext is used per document.

An AudioContext is said to be allowed to start if the user agent and the system allow audio output in the current context. In other words, if the AudioContext control thread state is allowed to transition from suspended to running.

Note: For example, a user agent could require that an AudioContext control thread state change to running is triggered by user activation (as described in [HTML]).

enum AudioContextLatencyCategory {
    "balanced",
    "interactive",
    "playback"
};

Enumeration description
"`balanced`"	Balance audio output latency and power consumption.
"`interactive`"	Provide the lowest audio output latency possible without glitching. This is the default.
"`playback`"	Prioritize sustained playback without interruption over audio output latency. Lowest power consumption.

[Exposed=Window,
 Constructor (optional AudioContextOptions contextOptions)]
interface AudioContext : BaseAudioContext {
    readonly attribute double baseLatency;
    readonly attribute double outputLatency;
    AudioTimestamp getOutputTimestamp ();
    Promise<void> suspend ();
    Promise<void> close ();
    MediaElementAudioSourceNode createMediaElementSource (HTMLMediaElement mediaElement);
    MediaStreamAudioSourceNode createMediaStreamSource (MediaStream mediaStream);
    MediaStreamTrackAudioSourceNode createMediaStreamTrackSource (MediaStreamTrack mediaStreamTrack);
    MediaStreamAudioDestinationNode createMediaStreamDestination ();
};

1.2.1. Constructors

AudioContext(contextOptions)

When creating an AudioContext, execute these steps:

Set a control thread state to suspended on the AudioContext.
Set a rendering thread state to suspended on the AudioContext.
Let pendingResumePromises be an empty ordered list of promises.
If contextOptions is given, apply the options:
1. Set the internal latency of this AudioContext according to contextOptions.latencyHint, as described in latencyHint.
2. If contextOptions.sampleRate is specified, set the sampleRate of this AudioContext to this value. Otherwise, use the sample rate of the default output device. If the selected sample rate differs from the sample rate of the output device, this AudioContext MUST resample the audio output to match the sample rate of the output device.
  
  Note: If resampling is required, the latency of the AudioContext may be affected, possibly by a large amount.
If the AudioContext is not allowed to start, abort these steps.
Send a control message to start processing.

Sending a control message to start processing means executing the following steps:

Attempt to acquire system resources.
In case of failure, abort these steps.
Set the rendering thread state to running on the AudioContext.
Queue a task on the control thread event loop, to execute these steps:
1. Set the state attribute of the AudioContext to "running".
2. Queue a task to fire a simple event named statechange at the AudioContext.

Note: It is unfortunately not possible to programatically notify authors that the creation of the AudioContext failed. User-Agents are encouraged to log an informative message if they have access to a logging mechanism, such as a developer tools console.

Arguments for the AudioContext.AudioContext() method.
Parameter	Type	Nullable	Optional	Description
`contextOptions`	AudioContextOptions	✘	✔	User-specified options controlling how the `AudioContext` should be constructed.

1.2.2. Attributes

baseLatency, of type double, readonly

This represents the number of seconds of processing latency incurred by the AudioContext passing the audio from the AudioDestinationNode to the audio subsystem. It does not include any additional latency that might be caused by any other processing between the output of the AudioDestinationNode and the audio hardware and specifically does not include any latency incurred the audio graph itself.

For example, if the audio context is running at 44.1 kHz and the AudioDestinationNode implements double buffering internally and can process and output audio each render quantum, then the processing latency is \((2\cdot128)/44100 = 5.805 \mathrm{ ms}\), approximately.

outputLatency, of type double, readonly

The estimation in seconds of audio output latency, i.e., the interval between the time the UA requests the host system to play a buffer and the time at which the first sample in the buffer is actually processed by the audio output device. For devices such as speakers or headphones that produce an acoustic signal, this latter time refers to the time when a sample’s sound is produced.

The outputLatency attribute value depends on the platform and the connected hardware audio output device. The outputLatency attribute value does not change for the context’s lifetime as long as the connected audio output device remains the same. If the audio output device is changed the outputLatency attribute value will be updated accordingly.

1.2.3. Methods

close()

Closes the AudioContext, releasing the system resources it’s using. This will not automatically release all AudioContext-created objects, but will suspend the progression of the AudioContext's currentTime, and stop processing audio data.

When close is called, execute these steps:

Let promise be a new Promise.
If the control thread state flag on the AudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
If the state attribute of the AudioContext is already "closed", resolve promise, return it, and abort these steps.
Set the control thread state flag on the AudioContext to closed.
Queue a control message to the AudioContext.
Return promise.

Running a control message to close an AudioContext means running these steps on the rendering thread:

Attempt to release system resources.
Set the rendering thread state to suspended.
Queue a task on the control thread’s event loop, to execute these steps:
1. Resolve promise.
2. If the state attribute of the AudioContext is not already "closed":
  1. Set the state attribute of the AudioContext to "closed".
  2. Queue a task to fire a simple event named statechange at the AudioContext.

When an AudioContext is closed, any MediaStreams and HTMLMediaElements that were connected to an AudioContext will have their output ignored. That is, these will no longer cause any output to speakers or other output devices. For more flexibility in behavior, consider using HTMLMediaElement.captureStream().

Note: When an AudioContext has been closed, implementation can choose to aggressively release more resources than when suspending.

No parameters.

Return type: Promise<void>

createMediaElementSource(mediaElement)

Creates a MediaElementAudioSourceNode given an HTMLMediaElement. As a consequence of calling this method, audio playback from the HTMLMediaElement will be re-routed into the processing graph of the AudioContext.

Arguments for the AudioContext.createMediaElementSource() method.
Parameter	Type	Nullable	Optional	Description
`mediaElement`	HTMLMediaElement	✘	✘	The media element that will be re-routed.

Return type: MediaElementAudioSourceNode

createMediaStreamDestination()

Creates a MediaStreamAudioDestinationNode

No parameters.

Return type: MediaStreamAudioDestinationNode

createMediaStreamSource(mediaStream)

Creates a MediaStreamAudioSourceNode.

Arguments for the AudioContext.createMediaStreamSource() method.
Parameter	Type	Nullable	Optional	Description
`mediaStream`	MediaStream	✘	✘	The media stream that will act as source.

Return type: MediaStreamAudioSourceNode

createMediaStreamTrackSource(mediaStreamTrack)

Creates a MediaStreamTrackAudioSourceNode.

Arguments for the AudioContext.createMediaStreamTrackSource() method.
Parameter	Type	Nullable	Optional	Description
`mediaStreamTrack`	MediaStreamTrack	✘	✘	The MediaStreamTrack that will act as source. The value of its `kind` attribute must be equal to `"audio"`, or an `InvalidStateError` exception MUST be thrown.

Return type: MediaStreamTrackAudioSourceNode

getOutputTimestamp()

Returns a new AudioTimestamp instance containing two correlated context’s audio stream position values: the contextTime member contains the time of the sample frame which is currently being rendered by the audio output device (i.e., output audio stream position), in the same units and origin as context’s currentTime; the performanceTime member contains the time estimating the moment when the sample frame corresponding to the stored contextTime value was rendered by the audio output device, in the same units and origin as performance.now() (described in [hr-time-2]).

If the context’s rendering graph has not yet processed a block of audio, then getOutputTimestamp call returns an AudioTimestamp instance with both members containing zero.

After the context’s rendering graph has started processing of blocks of audio, its currentTime attribute value always exceeds the contextTime value obtained from getOutputTimestamp method call.

The value returned from getOutputTimestamp method can be used to get performance time estimation for the slightly later context’s time value:

function outputPerformanceTime(contextTime) {
  var timestamp = context.getOutputTimestamp();
  var elapsedTime = contextTime - timestamp.contextTime;
  return timestamp.performanceTime + elapsedTime * 1000;
}

In the above example the accuracy of the estimation depends on how close the argument value is to the current output audio stream position: the closer the given contextTime is to timestamp.contextTime, the better the accuracy of the obtained estimation.

Note: The difference between the values of the context’s currentTime and the contextTime obtained from getOutputTimestamp method call cannot be considered as a reliable output latency estimation because currentTime may be incremented at non-uniform time intervals, so outputLatency attribute should be used instead.

No parameters.

Return type: AudioTimestamp

suspend()

Suspends the progression of AudioContext's currentTime, allows any current context processing blocks that are already processed to be played to the destination, and then allows the system to release its claim on audio hardware. This is generally useful when the application knows it will not need the AudioContext for some time, and wishes to temporarily release system resource associated with the AudioContext. The promise resolves when the frame buffer is empty (has been handed off to the hardware), or immediately (with no other effect) if the context is already suspended. The promise is rejected if the context has been closed.

When suspend is called, execute these steps:

Let promise be a new Promise.
If the control thread state flag on the AudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
Set the control thread state flag on the AudioContext to suspended.
Queue a control message to suspend the AudioContext.
Return promise.

Running a control message to suspend an AudioContext means running these steps on the rendering thread:

Attempt to release system resources.
Set the rendering thread state on the AudioContext to suspended.
Queue a task on the control thread’s event loop, to execute these steps:
1. Resolve promise.
2. If the state attribute of the AudioContext is not already "suspended":
  1. Set the state attribute of the AudioContext to "suspended".
  2. Queue a task to fire a simple event named statechange at the AudioContext.

While an AudioContext is suspended, MediaStreams will have their output ignored; that is, data will be lost by the real time nature of media streams. HTMLMediaElements will similarly have their output ignored until the system is resumed. AudioWorkletNodes and ScriptProcessorNodes will cease to have their processing handlers invoked while suspended, but will resume when the context is resumed. For the purpose of AnalyserNode window functions, the data is considered as a continuous stream - i.e. the resume()/suspend() does not cause silence to appear in the AnalyserNode's stream of data. In particular, calling AnalyserNode functions repeatedly when a AudioContext is suspended MUST return the same data.

No parameters.

Return type: Promise<void>

1.2.4. `AudioContextOptions`

The AudioContextOptions dictionary is used to specify user-specified options for an AudioContext.

dictionary AudioContextOptions {
  (AudioContextLatencyCategory or double) latencyHint = "interactive";
  float sampleRate;
};

1.2.4.1. Dictionary `AudioContextOptions` Members

latencyHint, of type (AudioContextLatencyCategory or double), defaulting to "interactive"

Identify the type of playback, which affects tradeoffs between audio output latency and power consumption.

The preferred value of the latencyHint is a value from AudioContextLatencyCategory. However, a double can also be specified for the number of seconds of latency for finer control to balance latency and power consumption. It is at the browser’s discretion to interpret the number appropriately. The actual latency used is given by AudioContext’s baseLatency attribute.

sampleRate, of type float

Set the sampleRate to this value for the AudioContext that will be created. The supported values are the same as the sample rates for an AudioBuffer. A NotSupportedError exception MUST be thrown if the specified sample rate is not supported.

If sampleRate is not specified, the preferred sample rate of the output device for this AudioContext is used.

1.2.5. `AudioTimestamp`

dictionary AudioTimestamp {
  double contextTime;
  DOMHighResTimeStamp performanceTime;
};

1.2.5.1. Dictionary `AudioTimestamp` Members

contextTime, of type double: Represents a point in the time coordinate system of BaseAudioContext’s currentTime.
performanceTime, of type DOMHighResTimeStamp: Represents a point in the time coordinate system of a Performance interface implementation (described in [hr-time-2]).

1.3. The `OfflineAudioContext` Interface

OfflineAudioContext is a particular type of BaseAudioContext for rendering/mixing-down (potentially) faster than real-time. It does not render to the audio hardware, but instead renders as quickly as possible, fulfilling the returned promise with the rendered result as an AudioBuffer.

[Exposed=Window,
 Constructor (OfflineAudioContextOptions contextOptions),
 Constructor (unsigned long numberOfChannels, unsigned long length, float sampleRate)]
interface OfflineAudioContext : BaseAudioContext {
  Promise<AudioBuffer> startRendering();
  Promise<void> suspend(double suspendTime);
  readonly attribute unsigned long length;
  attribute EventHandler oncomplete;
};

1.3.1. Constructors

OfflineAudioContext(contextOptions)

Let c be a new OfflineAudioContext object. Initialize c as follows:

Set the control thread state for c to "suspended".
Set the rendering thread state for c to "suspended".
Construct an AudioDestinationNode with its channelCount set to contextOptions.numberOfChannels.

Arguments for the OfflineAudioContext.OfflineAudioContext(contextOptions) method.
Parameter	Type	Nullable	Optional	Description
`contextOptions`				The initial parameters needed to construct this context.

OfflineAudioContext(numberOfChannels, length, sampleRate)

The OfflineAudioContext can constructed with the same arguments as AudioContext.createBuffer. A NotSupportedError exception MUST be thrown if any of the arguments is negative, zero, or outside its nominal range.

The OfflineAudioContext is constructed as if

new OfflineAudioContext({
    numberOfChannels: numberOfChannels,
    length: length,
    sampleRate: sampleRate
})

were called instead.

Arguments for the OfflineAudioContext.OfflineAudioContext(numberOfChannels, length, sampleRate) method.
Parameter	Type	Nullable	Optional	Description
`numberOfChannels`	unsigned long	✘	✘	Determines how many channels the buffer will have. See `createBuffer()` for the supported number of channels.
`length`	unsigned long	✘	✘	Determines the size of the buffer in sample-frames.
`sampleRate`	float	✘	✘	Describes the sample-rate of the linear PCM audio data in the buffer in sample-frames per second. See `createBuffer()` for valid sample rates.

1.3.2. Attributes

length, of type unsigned long, readonly: The size of the buffer in sample-frames. This is the same as the value of the length parameter for the constructor.
oncomplete, of type EventHandler: An EventHandler of type OfflineAudioCompletionEvent. It is the last event fired on an OfflineAudioContext.

1.3.3. Methods

startRendering()

Given the current connections and scheduled changes, starts rendering audio. The system shall ensure that the OfflineAudioContext is not garbage collected until either the promise is resolved and any callback function is called and completes, or until the suspend function is called.

Although the primary method of getting the rendered audio data is via its promise return value, the instance will also fire an event named complete for legacy reasons.

Let [[rendering started]] be an internal slot of this OfflineAudioContext. Initialize this slot to false.

When startRendering is called, the following steps MUST be performed on the control thread:

If the [[rendering started]] slot on the OfflineAudioContext is true, return a rejected promise with InvalidStateError, and abort these steps.
Set the [[rendering started]] slot of the OfflineAudioContext to true.
Let promise be a new promise.
Create a new AudioBuffer, with a number of channels, length and sample rate equal respectively to the numberOfChannels, length and sampleRate values passed to this instance’s constructor in the contextOptions parameter. Assign this buffer to an internal slot [[rendered buffer]] in the OfflineAudioContext.
If an exception was thrown during the preceding AudioBuffer constructor call, reject promise with this exception.
Otherwise, in the case that the buffer was successfully constructed, begin offline rendering.
Return promise.

To begin offline rendering, the following steps MUST happen on a rendering thread that is created for the occasion.

Given the current connections and scheduled changes, start rendering length sample-frames of audio into [[rendered buffer]]
For every render quantum, check and suspend the rendering if necessary.
If a suspended context is resumed, continue to render the buffer.
Once the rendering is complete, queue a task on the control thread’s event loop to perform the following steps:
1. Resolve the promise created by startRendering() with [[rendered buffer]].
2. Queue a task to fire an event named complete at this instance, using an instance of OfflineAudioCompletionEvent whose renderedBuffer property is set to [[rendered buffer]].

No parameters.

Return type: Promise<AudioBuffer>

suspend(suspendTime)

Schedules a suspension of the time progression in the audio context at the specified time and returns a promise. This is generally useful when manipulating the audio graph synchronously on OfflineAudioContext.

Note that the maximum precision of suspension is the size of the render quantum and the specified suspension time will be rounded down to the nearest render quantum boundary. For this reason, it is not allowed to schedule multiple suspends at the same quantized frame. Also, scheduling should be done while the context is not running to ensure precise suspension.

Arguments for the OfflineAudioContext.suspend() method.
Parameter	Type	Nullable	Optional	Description
`suspendTime`	double	✘	✘	Schedules a suspension of the rendering at the specified time, which is quantized and rounded down to the render quantum size. If the quantized frame number is negative or is less than or equal to the current time or is greater than or equal to the total render duration or is scheduled by another suspend for the same time, then the promise is rejected with `InvalidStateError`.

Return type: Promise<void>

1.3.4. `OfflineAudioContextOptions`

This specifies the options to use in constructing an OfflineAudioContext.

dictionary OfflineAudioContextOptions {
  unsigned long numberOfChannels = 1;
  required unsigned long length;
  required float sampleRate;
};

1.3.4.1. Dictionary `OfflineAudioContextOptions` Members

length, of type unsigned long: The length of the rendered AudioBuffer in sample-frames.
numberOfChannels, of type unsigned long, defaulting to 1: The number of channels for this OfflineAudioContext.
sampleRate, of type float: The sample rate for this OfflineAudioContext.

1.3.5. The `OfflineAudioCompletionEvent` Interface

This is an Event object which is dispatched to OfflineAudioContext for legacy reasons.

[Exposed=Window,
 Constructor (DOMString type, OfflineAudioCompletionEventInit eventInitDict)]
interface OfflineAudioCompletionEvent : Event {
  readonly attribute AudioBuffer renderedBuffer;
};

1.3.5.1. Attributes

renderedBuffer, of type AudioBuffer, readonly: An AudioBuffer containing the rendered audio data.

1.3.5.2. `OfflineAudioCompletionEventInit`

dictionary OfflineAudioCompletionEventInit : EventInit {
  required AudioBuffer renderedBuffer;
};

1.3.5.2.1. Dictionary `OfflineAudioCompletionEventInit` Members

renderedBuffer, of type AudioBuffer: Value to be assigned to the renderedBuffer attribute of the event.

1.4. The `AudioBuffer` Interface

This interface represents a memory-resident audio asset (for one-shot sounds and other short audio clips). Its format is non-interleaved 32-bit linear floating-point PCM values with a normal range of \([-1, 1]\), but values are not limited to this range. It can contain one or more channels. Typically, it would be expected that the length of the PCM data would be fairly short (usually somewhat less than a minute). For longer sounds, such as music soundtracks, streaming should be used with the audio element and MediaElementAudioSourceNode.

An AudioBuffer may be used by one or more AudioContexts, and can be shared between an OfflineAudioContext and an AudioContext.

AudioBuffer has four internal slots:

[[number of channels]]: The number of audio channels for this AudioBuffer, which is an unsigned long.
[[length]]: The length of each channel of this AudioBuffer, which is an unsigned long.
[[sample rate]]: The sample-rate, in Hz, of this AudioBuffer, a float
[[internal data]]: A data block holding the audio sample data.

[Exposed=Window,
 Constructor (AudioBufferOptions options)]
interface AudioBuffer {
  readonly attribute float sampleRate;
  readonly attribute unsigned long length;
  readonly attribute double duration;
  readonly attribute unsigned long numberOfChannels;
  Float32Array getChannelData (unsigned long channel);
  void copyFromChannel (Float32Array destination, unsigned long channelNumber, optional unsigned long startInChannel = 0);
  void copyToChannel (Float32Array source, unsigned long channelNumber, optional unsigned long startInChannel = 0);
};

1.4.1. Constructors

AudioBuffer(options)

If any of the values in options lie outside its nominal range, throw a NotSupportedError exception and abort the following steps.
Let b be a new AudioBuffer object.
Respectively assign the values of the attributes numberOfChannels, length, sampleRate of the AudioBufferOptions passed in the constructor to the internal slots [[number of channels]], [[length]], [[sample rate]].
Set the internal slot [[internal data]] of this AudioBuffer to the result of calling CreateByteDataBlock([[length]] * [[number of channels]]).

Note: This initializes the underlying storage to zero.
Return b.

Arguments for the AudioBuffer.AudioBuffer() method.
Parameter	Type	Nullable	Optional	Description
`options`	AudioBufferOptions	✘	✘

1.4.2. Attributes

duration, of type double, readonly

Duration of the PCM audio data in seconds.

This is computed from the [[sample rate]] and the [[length]] of the AudioBuffer by performing a division between the [[length]] and the [[sample rate]].

length, of type unsigned long, readonly

Length of the PCM audio data in sample-frames. This MUST return the value of [[length]].

numberOfChannels, of type unsigned long, readonly

The number of discrete audio channels. This MUST return the value of [[number of channels]].

sampleRate, of type float, readonly

The sample-rate for the PCM audio data in samples per second. This MUST return the value of [[sample rate]].

1.4.3. Methods

copyFromChannel(destination, channelNumber, startInChannel)

The copyFromChannel() method copies the samples from the specified channel of the AudioBuffer to the destination array.

Let buffer be the AudioBuffer with \(N_b\) frames, let \(N_f\) be the number of elements in the destination array, and \(k\) be the value of startInChannel. Then the number of frames copied from buffer to destination is \(\min(N_b - k, N_f)\). If this is less than \(N_f\), then the remaining elements of destination are not modified.

Arguments for the AudioBuffer.copyFromChannel() method.
Parameter	Type	Nullable	Optional	Description
`destination`	Float32Array	✘	✘	The array the channel data will be copied to.
`channelNumber`	unsigned long	✘	✘	The index of the channel to copy the data from. If `channelNumber` is greater or equal than the number of channel of the `AudioBuffer`, an `IndexSizeError` MUST be thrown.
`startInChannel`	unsigned long	✘	✔	An optional offset to copy the data from. If `startInChannel` is greater than the `length` of the `AudioBuffer`, an `IndexSizeError` MUST be thrown.

Return type: void

copyToChannel(source, channelNumber, startInChannel)

The copyToChannel() method copies the samples to the specified channel of the AudioBuffer from the source array.

A UnknownError may be thrown if source cannot be copied to the buffer.

Let buffer be the AudioBuffer with \(N_b\) frames, let \(N_f\) be the number of elements in the source array, and \(k\) be the value of startInChannel. Then the number of frames copied from source to the buffer is \(\min(N_b - k, N_f)\). If this is less than \(N_f\), then the remaining elements of buffer are not modified.

Arguments for the AudioBuffer.copyToChannel() method.
Parameter	Type	Nullable	Optional	Description
`source`	Float32Array	✘	✘	The array the channel data will be copied from.
`channelNumber`	unsigned long	✘	✘	The index of the channel to copy the data to. If `channelNumber` is greater or equal than the number of channel of the `AudioBuffer`, an `IndexSizeError` MUST be thrown.
`startInChannel`	unsigned long	✘	✔	An optional offset to copy the data to. If `startInChannel` is greater than the `length` of the `AudioBuffer`, an `IndexSizeError` MUST be thrown.

Return type: void

getChannelData(channel)

According to the rules described in acquire the content either get a reference to or get a copy of the bytes stored in [[internal data]] in a new Float32Array

A UnknownError may be thrown if the [[internal data]] or the new Float32Array cannot be created.

Arguments for the AudioBuffer.getChannelData() method.
Parameter	Type	Nullable	Optional	Description
`channel`	unsigned long	✘	✘	This parameter is an index representing the particular channel to get data for. An index value of 0 represents the first channel. This index value MUST be less than `numberOfChannels` or an `IndexSizeError` exception MUST be thrown.

Return type: Float32Array

Note: The methods copyToChannel() and copyFromChannel() can be used to fill part of an array by passing in a Float32Array that’s a view onto the larger array. When reading data from an AudioBuffer's channels, and the data can be processed in chunks, copyFromChannel() should be preferred to calling getChannelData() and accessing the resulting array, because it may avoid unnecessary memory allocation and copying.

An internal operation acquire the contents of an AudioBuffer is invoked when the contents of an AudioBuffer are needed by some API implementation. This operation returns immutable channel data to the invoker.

When an acquire the content operation occurs on an AudioBuffer, run the following steps:

If the operation IsDetachedBuffer on any of the AudioBuffer's ArrayBuffers return true, abort these steps, and return a zero-length channel data buffer to the invoker.
Detach all ArrayBuffers for arrays previously returned by getChannelData() on this AudioBuffer.
Retain the underlying [[internal data]] from those ArrayBuffers and return references to them to the invoker.
Attach ArrayBuffers containing copies of the data to the AudioBuffer, to be returned by the next call to getChannelData().

The acquire the contents of an AudioBuffer operation is invoked in the following cases:

When AudioBufferSourceNode.start is called, it acquires the contents of the node’s buffer. If the operation fails, nothing is played.
When the buffer of an AudioBufferSourceNode is set and AudioBufferSourceNode.start has been previously called, the setter acquires the content of the AudioBuffer. If the operation fails, nothing is played.
When a ConvolverNode's buffer is set to an AudioBuffer while the node is connected to an output node, or a ConvolverNode is connected to an output node while the ConvolverNode's buffer is set to an AudioBuffer, it acquires the content of the AudioBuffer.
When the dispatch of an AudioProcessingEvent completes, it acquires the contents of its outputBuffer.

Note: This means that copyToChannel() cannot be used to change the content of an AudioBuffer currently in use by an AudioNode that has acquired the content of an AudioBuffer since the AudioNode will continue to use the data previously acquired.

1.4.4. `AudioBufferOptions`

This specifies the options to use in constructing an AudioBuffer. The length and sampleRate members are required. A NotFoundError exception MUST be thrown if any of the required members are not specified.

dictionary AudioBufferOptions {
  unsigned long numberOfChannels = 1;
  required unsigned long length;
  required float sampleRate;
};

1.4.4.1. Dictionary `AudioBufferOptions` Members

The allowed values for the members of this dictionary are constrained. See createBuffer().

length, of type unsigned long: The length in sample frames of the buffer. See length for constraints.
numberOfChannels, of type unsigned long, defaulting to 1: The number of channels for the buffer. See numberOfChannels for constraints.
sampleRate, of type float: The sample rate in Hz for the buffer. See sampleRate for constraints.

1.5. The `AudioNode` Interface

AudioNodes are the building blocks of an AudioContext. This interface represents audio sources, the audio destination, and intermediate processing modules. These modules can be connected together to form processing graphs for rendering audio to the audio hardware. Each node can have inputs and/or outputs. A source node has no inputs and a single output. Most processing nodes such as filters will have one input and one output. Each type of AudioNode differs in the details of how it processes or synthesizes audio. But, in general, an AudioNode will process its inputs (if it has any), and generate audio for its outputs (if it has any).

Each output has one or more channels. The exact number of channels depends on the details of the specific AudioNode.

An output may connect to one or more AudioNode inputs, thus fan-out is supported. An input initially has no connections, but may be connected from one or more AudioNode outputs, thus fan-in is supported. When the connect() method is called to connect an output of an AudioNode to an input of an AudioNode, we call that a connection to the input.

Each AudioNode input has a specific number of channels at any given time. This number can change depending on the connection(s) made to the input. If the input has no connections then it has one channel which is silent.

For each input, an AudioNode performs a mixing (usually an up-mixing) of all connections to that input. Please see §3 Mixer Gain Structure for more informative details, and the §5 Channel Up-Mixing and Down-Mixing section for normative requirements.

The processing of inputs and the internal operations of an AudioNode take place continuously with respect to AudioContext time, regardless of whether the node has connected outputs, and regardless of whether these outputs ultimately reach an AudioContext's AudioDestinationNode.

For performance reasons, practical implementations will need to use block processing, with each AudioNode processing a fixed number of sample-frames of size block-size. In order to get uniform behavior across implementations, we will define this value explicitly. block-size is defined to be 128 sample-frames which corresponds to roughly 3ms at a sample-rate of 44.1 kHz.

[Exposed=Window]
interface AudioNode : EventTarget {
  AudioNode connect (AudioNode destinationNode,
                     optional unsigned long output = 0,
                     optional unsigned long input = 0);
  void connect (AudioParam destinationParam, optional unsigned long output = 0);
  void disconnect ();
  void disconnect (unsigned long output);
  void disconnect (AudioNode destinationNode);
  void disconnect (AudioNode destinationNode, unsigned long output);
  void disconnect (AudioNode destinationNode, unsigned long output, unsigned long input);
  void disconnect (AudioParam destinationParam);
  void disconnect (AudioParam destinationParam, unsigned long output);
  readonly attribute BaseAudioContext context;
  readonly attribute unsigned long numberOfInputs;
  readonly attribute unsigned long numberOfOutputs;
  attribute unsigned long channelCount;
  attribute ChannelCountMode channelCountMode;
  attribute ChannelInterpretation channelInterpretation;
};

1.5.1. AudioNode Creation

AudioNodes can be created in two ways: by using the constructor for this particular interface, or by using the factory method on the BaseAudioContext or AudioContext.

The BaseAudioContext passed as first argument of the constructor of an AudioNodes is called the associated BaseAudioContext of the AudioNode to be created. Similarly, when using the factory method, the associated BaseAudioContext of the AudioNode is the BaseAudioContext this factory method is called on.

To create a new AudioNode of a particular type n using its constructor, with a BaseAudioContext c as first argument, and an associated option object option as second argument, from the relevant global of c, execute these steps:

Let o be a new object of type n.
Initialize o, with c and option as arguments.
Return o

To create a new AudioNode of a particular type n using its factory method, called on a BaseAudioContext c, execute these steps:

Let o be a new object of type n.
Let option be a dictionary of the type associated to the interface associated to this factory method.
For each parameter passed to the factory method, set the dictionary member of the same name on option to the value of this parameter.
Initialize o with c and option as arguments.
Return o

Initializing an object o of interface n that inherits from AudioNode means executing the following steps, given the arguments context and dict passed to the constructor of this interface.

Set o’s associated BaseAudioContext to context.
Set its value for numberOfInputs, numberOfOutputs, channelCount, channelCountMode, channelInterpretation to the default value for this specific interface outlined in the section for each AudioNode.
If the AudioNode being constructed is a ConvolverNode, set its normalize attribute with the inverse of the value of the disableNormalization in dict, and then set its buffer attribute to the value of the buffer in dict member, in this order, and jump to the last step of this algorithm.

Note: This means that the buffer will be normalized according to the value of the normalize attribute.
For each member of dict passed in, execute these steps, with k the key of the member, and v its value:
1. If k is disableNormalization or buffer and n is ConvolverNode, jump to the beginning of this loop.
2. If k is the name of an AudioParam on this interface, set the value attribute of this AudioParam to v.
3. Else if k is the name of an attribute on this interface, set the object associated with this attribute to v.

The associated interface for a factory method is the interface of the objects that are returned from this method. The associated option object for an interface is the option object that can be passed to the constructor for this interface.

AudioNodes are EventTargets, as described in [DOM]. This means that it is possible to dispatch events to AudioNodes the same way that other EventTargets accept events.

enum ChannelCountMode {
  "max",
  "clamped-max",
  "explicit"
};

The ChannelCountMode, in conjuction with the node’s channelCount and channelInterpretation values, is used to determine the computedNumberOfChannels that controls how inputs to a node are to be mixed. The computedNumberOfChannels is determined as shown below. See §5 Channel Up-Mixing and Down-Mixing for more information on how mixing is to be done.

Enumeration description
"`max`"	computedNumberOfChannels is the maximum of the number of channels of all connections to an input. In this mode `channelCount` is ignored.
"`clamped-max`"	computedNumberOfChannels is determined as for "`max`" and then clamped to a maximum value of the given `channelCount`.
"`explicit`"	computedNumberOfChannels is the exact value as specified by the `channelCount`.

enum ChannelInterpretation {
  "speakers",
  "discrete"
};

Enumeration description
"`speakers`"	use up-mix equations or down-mix equations. In cases where the number of channels do not match any of these basic speaker layouts, revert to "`discrete`".
"`discrete`"	Up-mix by filling channels until they run out then zero out remaining channels. Down-mix by filling as many channels as possible, then dropping remaining channels.

1.5.2. Attributes

channelCount, of type unsigned long

channelCount is the number of channels used when up-mixing and down-mixing connections to any inputs to the node. The default value is 2 except for specific nodes where its value is specially determined. This attribute has no effect for nodes with no inputs. If this value is set to zero or to a value greater than the implementation’s maximum number of channels the implementation MUST throw a NotSupportedError exception.

In addition, some nodes have additional channelCount constraints on the possible values for the channel count:

AudioDestinationNode

The behavior depends on whether the destination node is the destination of an AudioContext or OfflineAudioContext:

AudioContext: The channel count MUST be between 1 and

o maxChannelCount. An IndexSizeError exception MUST be thrown for any attempt to set the count outside this range.

OfflineAudioContext: The channel count cannot be changed. An InvalidStateError exception MUST be thrown for any attempt to change the value.

ChannelMergerNode: The channel count cannot be changed, and an InvalidStateError exception MUST be thrown for any attempt to change the value.
ChannelSplitterNode: The channel count cannot be changed, and an InvalidStateError exception MUST be thrown for any attempt to change the value.
ConvolverNode: The channel count cannot changed from two, and a NotSupportedError exception MUST be thrown for any attempt to change the value.
DynamicsCompressorNode: The channel count cannot be greater than two, and a NotSupportedError exception MUST be thrown for any attempt to change the to a value greater than two.
PannerNode: The channel count cannot be greater than two, and a NotSupportedError exception MUST be thrown for any attempt to change the to a value greater than two.
ScriptProcessorNode: The channel count cannot be changed, and an InvalidStateError exception MUST be thrown for any attempt to change the value.
StereoPannerNode: The channel count cannot be greater than two, and a NotSupportedError exception MUST be thrown for any attempt to change the to a value greater than two.

See §5 Channel Up-Mixing and Down-Mixing for more information on this attribute.

channelCountMode, of type ChannelCountMode

channelCountMode determines how channels will be counted when up-mixing and down-mixing connections to any inputs to the node. The default value is "max". This attribute has no effect for nodes with no inputs.

In addition, some nodes have additional channelCountMode constraints on the possible values for the channel count mode:

AudioDestinationNode: If the AudioDestinationNode is the destination node of an OfflineAudioContext, then the channel count mode cannot be changed. An InvalidStateError exception MUST be thrown for any attempt to change the value.
ChannelMergerNode: The channel count mode cannot be changed from "explicit" and an InvalidStateError exception MUST be thrown for any attempt to change the value.
ChannelSplitterNode: The channel count mode cannot be changed from "explicit" and an InvalidStateError exception MUST be thrown for any attempt to change the value.
ConvolverNode: The channel count mode cannot be changed from "clamped-max", and a NotSupportedError exception MUST be thrown for any attempt to change the value.
DynamicsCompressorNode: The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".
PannerNode: The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".
ScriptProcessorNode: The channel count mode cannot be changed from "explicit" and an InvalidStateError exception MUST be thrown for any attempt to change the value.
StereoPannerNode: The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".

See the §5 Channel Up-Mixing and Down-Mixing section for more information on this attribute.

channelInterpretation, of type ChannelInterpretation

channelInterpretation determines how individual channels will be treated when up-mixing and down-mixing connections to any inputs to the node. The default value is "speakers". This attribute has no effect for nodes with no inputs.

In addition, some nodes have additional channelInterpretation constraints on the possible values for the channel interpretation:

ChannelSplitterNode: The channel intepretation can not be changed from "discrete" and a InvalidStateError exception MUST be thrown for any attempt to change the value.

See §5 Channel Up-Mixing and Down-Mixing for more information on this attribute.

context, of type BaseAudioContext, readonly

The BaseAudioContext which owns this AudioNode.

numberOfInputs, of type unsigned long, readonly

The number of inputs feeding into the AudioNode. For source nodes, this will be 0. This attribute is predetermined for many AudioNode types, but some AudioNodes, like the ChannelMergerNode and the AudioWorkletNode, have variable number of inputs.

numberOfOutputs, of type unsigned long, readonly

The number of outputs coming out of the AudioNode. This attribute is predetermined for some AudioNode types, but can be variable, like for the ChannelSplitterNode and the AudioWorkletNode.

1.5.3. Methods

connect(destinationNode, output, input)

There can only be one connection between a given output of one specific node and a given input of another specific node. Multiple connections with the same termini are ignored.

For example:

nodeA.connect(nodeB);
nodeA.connect(nodeB);

will have the same effect as

nodeA.connect(nodeB);

This method returns destination AudioNode object.

Arguments for the AudioNode.connect(destinationNode, output, input) method.
Parameter	Type	Nullable	Optional	Description
`destinationNode`				The `destination` parameter is the `AudioNode` to connect to. If the `destination` parameter is an `AudioNode` that has been created using another `AudioContext`, an `InvalidAccessError` MUST be thrown. That is, `AudioNode`s cannot be shared between `AudioContext`s.
`output`	unsigned long	✘	✔	The `output` parameter is an index describing which output of the `AudioNode` from which to connect. If this parameter is out-of-bound, an `IndexSizeError` exception MUST be thrown. It is possible to connect an `AudioNode` output to more than one input with multiple calls to connect(). Thus, "fan-out" is supported.
`input`				The `input` parameter is an index describing which input of the destination `AudioNode` to connect to. If this parameter is out-of-bounds, an `IndexSizeError` exception MUST be thrown. It is possible to connect an `AudioNode` to another `AudioNode` which creates a cycle: an `AudioNode` may connect to another `AudioNode`, which in turn connects back to the input or `AudioParam` of the first `AudioNode`. This is allowed only if there is at least one `DelayNode` in the cycle or a `NotSupportedError` exception MUST be thrown.

Return type: AudioNode

connect(destinationParam, output)

Connects the AudioNode to an AudioParam, controlling the parameter value with an audio-rate signal.

It is possible to connect an AudioNode output to more than one AudioParam with multiple calls to connect(). Thus, "fan-out" is supported.

It is possible to connect more than one AudioNode output to a single AudioParam with multiple calls to connect(). Thus, "fan-in" is supported.

An AudioParam will take the rendered audio data from any AudioNode output connected to it and convert it to mono by down-mixing if it is not already mono, then mix it together with other such outputs and finally will mix with the intrinsic parameter value (the value the AudioParam would normally have without any audio connections), including any timeline changes scheduled for the parameter.

The down-mixing to mono is equivalent to the down-mixing for an AudioNode with channelCount = 1, channelCountMode = "explicit", and channelInterpretation = "speakers".

There can only be one connection between a given output of one specific node and a specific AudioParam. Multiple connections with the same termini are ignored.

For example:

nodeA.connect(param);
nodeA.connect(param);

will have the same effect as

nodeA.connect(param);

Arguments for the AudioNode.connect(destinationParam, output) method.
Parameter	Type	Nullable	Optional	Description
`destinationParam`	AudioParam	✘	✘	The `destination` parameter is the `AudioParam` to connect to. This method does not return `destination` `AudioParam` object. If `destinationParam` belongs to an `AudioNode` that belongs to a `BaseAudioContext` that is different from the `BaseAudioContext` that has created the `AudioNode` on which this method was called, an `InvalidAccessError` MUST be thrown.
`output`	unsigned long	✘	✔	The `output` parameter is an index describing which output of the `AudioNode` from which to connect. If the `parameter` is out-of-bound, an `IndexSizeError` exception MUST be thrown.

Return type: void

disconnect()

Disconnects all outgoing connections from the AudioNode.

No parameters.

Return type: void

disconnect(output)

Disconnects a single output of the AudioNode from any other AudioNode or AudioParam objects to which it is connected.

Arguments for the AudioNode.disconnect(output) method.
Parameter	Type	Nullable	Optional	Description
`output`	unsigned long	✘	✘	This parameter is an index describing which output of the `AudioNode` to disconnect. It disconnects all outgoing connections from the given output. If this parameter is out-of-bounds, an `IndexSizeError` exception MUST be thrown.

Return type: void

disconnect(destinationNode)

Disconnects all outputs of the AudioNode that go to a specific destination AudioNode.

Arguments for the AudioNode.disconnect(destinationNode) method.
Parameter	Type	Nullable	Optional	Description
`destinationNode`				The `destinationNode` parameter is the `AudioNode` to disconnect. It disconnects all outgoing connections to the given `destinationNode`. If there is no connection to the `destinationNode`, an `InvalidAccessError` exception MUST be thrown.

Return type: void

disconnect(destinationNode, output)

Disconnects a specific output of the AudioNode from a specific input of some destination AudioNode.

Arguments for the AudioNode.disconnect(destinationNode, output) method.
Parameter	Type	Nullable	Optional	Description
`destinationNode`				The `destinationNode` parameter is the `AudioNode` to disconnect. If there is no connection to the `destinationNode` from the given output, an `InvalidAccessError` exception MUST be thrown.
`output`	unsigned long	✘	✘	The `output` parameter is an index describing which output of the `AudioNode` from which to disconnect. If this parameter is out-of-bound, an `IndexSizeError` exception MUST be thrown.

Return type: void

disconnect(destinationNode, output, input)

Disconnects a specific output of the AudioNode from a specific input of some destination AudioNode.

Arguments for the AudioNode.disconnect(destinationNode, output, input) method.
Parameter	Type	Nullable	Optional	Description
`destinationNode`				The `destinationNode` parameter is the `AudioNode` to disconnect. If there is no connection to the `destinationNode` from the given output, an `InvalidAccessError` exception MUST be thrown.
`output`	unsigned long	✘	✘	The `output` parameter is an index describing which output of the `AudioNode` from which to disconnect. If this parameter is out-of-bound, an `IndexSizeError` exception MUST be thrown.
`input`				The `input` parameter is an index describing which input of the destination `AudioNode` to disconnect. If this parameter is out-of-bounds, an `IndexSizeError` exception MUST be thrown.

Return type: void

disconnect(destinationParam)

Disconnects all outputs of the AudioNode that go to a specific destination AudioParam. The contribution of this AudioNode to the computed parameter value goes to 0 when this operation takes effect. The intrinsic parameter value is not affected by this operation.

Arguments for the AudioNode.disconnect(destinationParam) method.
Parameter	Type	Nullable	Optional	Description
`destinationParam`	AudioParam	✘	✘	The `destinationParam` parameter is the `AudioParam` to disconnect. If there is no connection to the `destinationParam`, an `InvalidAccessError` exception MUST be thrown.

Return type: void

disconnect(destinationParam, output)

Disconnects a specific output of the AudioNode from a specific destination AudioParam. The contribution of this AudioNode to the computed parameter value goes to 0 when this operation takes effect. The intrinsic parameter value is not affected by this operation.

Arguments for the AudioNode.disconnect(destinationParam, output) method.
Parameter	Type	Nullable	Optional	Description
`destinationParam`	AudioParam	✘	✘	The `destinationParam` parameter is the `AudioParam` to disconnect. If there is no connection to the `destinationParam`, an `InvalidAccessError` exception MUST be thrown.
`output`	unsigned long	✘	✘	The `output` parameter is an index describing which output of the `AudioNode` from which to disconnect. If the `parameter` is out-of-bound, an `IndexSizeError` exception MUST be thrown.

Return type: void

1.5.4. `AudioNodeOptions`

This specifies the options that can be used in constructing all AudioNodes. All members are optional. However, the specific values used for each node depends on the actual node.

dictionary AudioNodeOptions {
  unsigned long channelCount;
  ChannelCountMode channelCountMode;
  ChannelInterpretation channelInterpretation;
};

1.5.4.1. Dictionary `AudioNodeOptions` Members

channelCount, of type unsigned long: Desired number of channels for the channelCount attribute.
channelCountMode, of type ChannelCountMode: Desired mode for the channelCountMode attribute.
channelInterpretation, of type ChannelInterpretation: Desired mode for the channelInterpretation attribute.

1.5.5. Lifetime

The following behaviors provide a normative description of the conditions under which an AudioNode is alive, meaning that it MUST be retained in the graph by an implementation. Where these conditions do not apply, AudioNodes MAY be released by an implementation.

There are several types of references:

A normal reference obeying normal garbage collection rules.
A playing reference for AudioBufferSourceNodes, MediaElementAudioSourceNodes, MediaStreamAudioSourceNodes and OscillatorNodes. These nodes maintain a playing reference to themselves while they are currently playing.
A connection reference which occurs if another AudioNode is connected to one or more of its inputs. Connections to a node’s AudioParams do not imply a connection reference.
A tail-time reference which an AudioNode maintains on itself as long as it has any internal processing state which has not yet been emitted. For example, a ConvolverNode has a tail which continues to play even after receiving silent input (think about clapping your hands in a large concert hall and continuing to hear the sound reverberate throughout the hall). Some AudioNodes have this property. Please see details for specific nodes.
MediaStreams keep a MediaStreamAudioSourceNode alive as long as the underlying MediaStreamTrack that is playing through the MediaStreamAudioSourceNode has not ended (as per [mediacapture-streams]).
HTMLMediaElements keep their associated MediaElementAudioSourceNode alive as long as the HTMLMediaElement is in a state where audio could ever be played in the future.

Note: An HTMLMediaElement that has its src attribute set to "", and all its references dropped allows the MediaElementAudioSourceNode to be released as well (granted nothing keeps the MediaElementAudioSourceNode alive).

Any AudioNodes which are connected in a cycle and are directly or indirectly connected to a AudioDestinationNode or MediaStreamAudioDestinationNode within the AudioContext will stay alive as long as the AudioContext is alive.

Note: The uninterrupted operation of AudioNodes implies that as long as live references exist to a node, the node will continue processing its inputs and evolving its internal state even if it is disconnected from the audio graph. Since this processing will consume CPU and power, developers should carefully consider the resource usage of disconnected nodes. In particular, it is a good idea to minimize resource consumption by explicitly putting disconnected nodes into a stopped state when possible.

When an AudioNode has no references it will be deleted. Before it is deleted, it will disconnect itself from any other AudioNodes which it is connected to. In this way it releases all connection references (3) it has to other nodes.

Regardless of any of the above references, it can be assumed that the AudioNode will be deleted when its AudioContext is deleted.

1.6. The `AudioParam` Interface

AudioParam controls an individual aspect of an AudioNode's functioning, such as volume. The parameter can be set immediately to a particular value using the value attribute. Or, value changes can be scheduled to happen at very precise times (in the coordinate system of AudioContext's currentTime attribute), for envelopes, volume fades, LFOs, filter sweeps, grain windows, etc. In this way, arbitrary timeline-based automation curves can be set on any AudioParam. Additionally, audio signals from the outputs of AudioNodes can be connected to an AudioParam, summing with the intrinsic parameter value.

Some synthesis and processing AudioNodes have AudioParams as attributes whose values MUST be taken into account on a per-audio-sample basis. For other AudioParams, sample-accuracy is not important and the value changes can be sampled more coarsely. Each individual AudioParam will specify that it is either an a-rate parameter which means that its values MUST be taken into account on a per-audio-sample basis, or it is a k-rate parameter.

Implementations MUST use block processing, with each AudioNode processing one render quantum.

For each render quantum, the value of a k-rate parameter MUST be sampled at the time of the very first sample-frame, and that value MUST be used for the entire block. a-rate parameters MUST be sampled for each sample-frame of the block. Depending on the AudioParam, its rate can be controlled by setting the automationRate attribute to either "a-rate" or "k-rate". See the description of the individual AudioParams for further details.

Each AudioParam includes minValue and maxValue attributes that together form the simple nominal range for the parameter. In effect, value of the parameter is clamped to the range \([\mathrm{minValue}, \mathrm{maxValue}]\). See §1.6.3 Computation of Value for full details.

For many AudioParams the minValue and maxValue is intended to be set to the maximum possible range. In this case, maxValue should be set to the most-positive-single-float value, which is 3.4028235e38. (However, in JavaScript which only supports IEEE-754 double precision float values, this must be written as 3.4028234663852886e38.) Similarly, minValue should be set to the most-negative-single-float value, which is the negative of the most-positive-single-float: -3.4028235e38. (Similarly, this must be written in JavaScript as -3.4028234663852886e38.)

An AudioParam maintains a list of zero or more automation events. Each automation event specifies changes to the parameter’s value over a specific time range, in relation to its automation event time in the time coordinate system of the AudioContext's currentTime attribute. The list of automation events is maintained in ascending order of automation event time.

The behavior of a given automation event is a function of the AudioContext's current time, as well as the automation event times of this event and of adjacent events in the list. The following automation methods change the event list by adding a new event to the event list, of a type specific to the method:

setValueAtTime() - SetValue
linearRampToValueAtTime() - LinearRampToValue
exponentialRampToValueAtTime() - ExponentialRampToValue
setTargetAtTime() - SetTarget
setValueCurveAtTime() - SetValueCurve

The following rules will apply when calling these methods:

Automation event times are not quantized with respect to the prevailing sample rate. Formulas for determining curves and ramps are applied to the exact numerical times given when scheduling events.
If one of these events is added at a time where there is already one or more events, then it will be placed in the list after them, but before events whose times are after the event.
If setValueCurveAtTime() is called for time \(T\) and duration \(D\) and there are any events having a time strictly greater than \(T\), but strictly less than \(T + D\), then a NotSupportedError exception MUST be thrown. In other words, it’s not ok to schedule a value curve during a time period containing other events, but it’s ok to schedule a value curve exactly at the time of another event.
Similarly a NotSupportedError exception MUST be thrown if any automation method is called at a time which is contained in \([T, T+D)\), \(T\) being the time of the curve and \(D\) its duration.

Note: AudioParam attributes are read only, with the exception of the value attribute.

The automation rate of an AudioParam can be selected setting the automationRate attribute with one of the following values. However, some AudioParams have constraints on whether the automation rate can be changed.

enum AutomationRate {
  "a-rate",
  "k-rate"
};

Enumeration description
"`a-rate`"	This `AudioParam` is set for a-rate processing.
"`k-rate`"	This `AudioParam` is set for k-rate processing.

Each AudioParam has an internal slot [[current value]], initially set to the AudioParam's defaultValue.

[Exposed=Window]
interface AudioParam {
  attribute float value;
  attribute AutomationRate automationRate;
  readonly attribute float defaultValue;
  readonly attribute float minValue;
  readonly attribute float maxValue;
  AudioParam setValueAtTime (float value, double startTime);
  AudioParam linearRampToValueAtTime (float value, double endTime);
  AudioParam exponentialRampToValueAtTime (float value, double endTime);
  AudioParam setTargetAtTime (float target, double startTime, float timeConstant);
  AudioParam setValueCurveAtTime (sequence<float> values, double startTime, double duration);
  AudioParam cancelScheduledValues (double cancelTime);
  AudioParam cancelAndHoldAtTime (double cancelTime);
};

1.6.1. Attributes

automationRate, of type AutomationRate

The automation rate for the AudioParam. The default value depends on the actual AudioParam; see the description of the AudioParam for the default value.

Some nodes have additional automation rate constraints as follows:

AudioBufferSourceNode: The AudioParams playbackRate and detune MUST be "k-rate". An InvalidStateError must be thrown if the rate is changed to "a-rate".
DynamicsCompressorNode: The AudioParams threshold, knee, ratio, attack, and release MUST be "k-rate". An InvalidStateError must be thrown if the rate is changed to "a-rate".
PannerNode: If the panningModel is "HRTF", the setting of the automationRate for any AudioParam of the PannerNode is ignored. Likewise, the setting of the automationRate for any AudioParam of the AudioListener is ignored. In this case, the AudioParam behaves as if the automationRate were set to "k-rate".

defaultValue, of type float, readonly

Initial value for the value attribute.

maxValue, of type float, readonly

The nominal maximum value that the parameter can take. Together with minValue, this forms the nominal range for this parameter.

minValue, of type float, readonly

The nominal minimum value that the parameter can take. Together with maxValue, this forms the nominal range for this parameter.

value, of type float

The parameter’s floating-point value. This attribute is initialized to the defaultValue.

Getting this attribute returns the contents of the [[current value]] slot, which maintains the value of this parameter at the conclusion of the most recent render quantum on the audio rendering thread, or the most recently assigned value if no rendering has taken place.

Setting this attribute has the effect of assigning the requested value to the [[current value]] slot, and calling the setValueAtTime() method with the current AudioContext's currentTime and [[current value]]. Any exceptions that would be thrown by setValueAtTime() will also be thrown by setting this attribute.

1.6.2. Methods

cancelAndHoldAtTime(cancelTime)

This is similar to cancelScheduledValues() in that it cancels all scheduled parameter changes with times greater than or equal to cancelTime. However, in addition, the automation value that would have happened at cancelTime is then proprogated for all future time until other automation events are introduced.

The behavior of the timeline in the face of cancelAndHoldAtTime() when automations are running and can be introduced at any time after calling cancelAndHoldAtTime() and before cancelTime is reached is quite complicated. The behavior of cancelAndHoldAtTime() is therefore specified in the following algorithm.

Let \(t_c\) be the value of cancelTime. Then

Let \(E_1\) be the event (if any) at time \(t_1\) where \(t_1\) is the largest number satisfying \(t_1 \le t_c\).
Let \(E_2\) be the event (if any) at time \(t_2\) where \(t_2\) is the smallest number satisfying \(t_c \lt t_2\).
If \(E_2\) exists:
1. If \(E_2\) is a linear or exponential ramp,
  1. Effectively rewrite \(E_2\) to be the same kind of ramp ending at time \(t_c\) with an end value that would be the value of the original ramp at time \(t_c\).
  2. Go to step 5.
2. Otherwise, go to step 4.
If \(E_1\) exists:
1. If \(E_1\) is a setTarget event,
  1. Implicitly insert a setValueAtTime event at time \(t_c\) with the value that the setTarget would have at time \(t_c\).
  2. Go to step 5.
2. If \(E_1\) is a setValueCurve with a start time of \(t_3\) and a duration of \(d\)
  1. If \(t_c \gt t_3 + d\), go to step 5.
  2. Otherwise,
    1. Effectively replace this event with a setValueCurve event with a start time of \(t_3\) and a new duration of \(t_c-t_3\). However, this is not a true replacement; this automation MUST take care to produce the same output as the original, and not one computed using a different duration. (That would cause sampling of the value curve in a slightly different way, producing different results.)
    2. Go to step 5.
Remove all events with time greater than \(t_c\).

If no events are added, then the automation value after cancelAndHoldAtTime() is the constant value that the original timeline would have had at time \(t_c\).

Arguments for the AudioParam.cancelAndHoldAtTime() method.
Parameter	Type	Nullable	Optional	Description
`cancelTime`	double	✘	✘	The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the `AudioContext`'s `currentTime` attribute. A `RangeError` exception MUST be thrown if `cancelTime` is negative or is not a finite number. If `cancelTime` is less than `currentTime`, it is clamped to `currentTime`.

Return type: AudioParam

cancelScheduledValues(cancelTime)

Cancels all scheduled parameter changes with times greater than or equal to cancelTime. Cancelling a scheduled parameter change means removing the scheduled event from the event list. Any active automations whose automation event time is less than cancelTime are also cancelled, and such cancellations may cause discontinuities because the original value (from before such automation) is restored immediately. Any hold values scheduled by cancelAndHoldAtTime() are also removed if the hold time occurs after cancelTime.

Arguments for the AudioParam.cancelScheduledValues() method.
Parameter	Type	Nullable	Optional	Description
`cancelTime`	double	✘	✘	The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the `AudioContext`'s `currentTime` attribute. A `RangeError` exception MUST be thrown if `cancelTime` is negative or is not a finite number. If `cancelTime` is less than `currentTime`, it is clamped to `currentTime`.

Return type: AudioParam

exponentialRampToValueAtTime(value, endTime)

Schedules an exponential continuous change in parameter value from the previous scheduled parameter value to the given value. Parameters representing filter frequencies and playback rate are best changed exponentially because of the way humans perceive sound.

The value during the time interval \(T_0 \leq t < T_1\) (where \(T_0\) is the time of the previous event and \(T_1\) is the endTime parameter passed into this method) will be calculated as:

$$
  v(t) = V_0 \left(\frac{V_1}{V_0}\right)^\frac{t - T_0}{T_1 - T_0}
$$

where \(V_0\) is the value at the time \(T_0\) and \(V_1\) is the value parameter passed into this method. If \(V_0\) and \(V_1\) have opposite signs or if \(V_0\) is zero, then \(v(t) = V_0\) for \(T_0 \le t \lt T_1\).

This also implies an exponential ramp to 0 is not possible. A good approximation can be achieved using setTargetAtTime() with an appropriately chosen time constant.

If there are no more events after this ExponentialRampToValue event then for \(t \geq T_1\), \(v(t) = V_1\).

If there is no event preceding this event, the exponential ramp behaves as if setValueAtTime(value, currentTime) were called where value is the current value of the attribute and currentTime is the context currentTime at the time exponentialRampToValueAtTime is called.

If the preceding event is a SetTarget event, \(T_0\) and \(V_0\) are chosen from the current time and value of SetTarget automation. That is, if the SetTarget event has not started, \(T_0\) is the start time of the event, and \(V_0\) is the value just before the SetTarget event starts. In this case, the ExponentialRampToValue event effectively replaces the SetTarget event. If the SetTarget event has already started, \(T_0\) is the current context time, and \(V_0\) is the current SetTarget automation value at time \(T_0\). In both cases, the automation curve is continuous.

Arguments for the AudioParam.exponentialRampToValueAtTime() method.
Parameter	Type	Nullable	Optional	Description
`value`	float	✘	✘	The value the parameter will exponentially ramp to at the given time. A `RangeError` exception MUST be thrown if this value is equal to 0.
`endTime`	double	✘	✘	The time in the same time coordinate system as the `AudioContext`'s `currentTime` attribute where the exponential ramp ends. A `RangeError` exception MUST be thrown if `endTime` is negative or is not a finite number. If `endTime` is less than `currentTime`, it is clamped to `currentTime`.

Return type: AudioParam

linearRampToValueAtTime(value, endTime)

Schedules a linear continuous change in parameter value from the previous scheduled parameter value to the given value.

The value during the time interval \(T_0 \leq t < T_1\) (where \(T_0\) is the time of the previous event and \(T_1\) is the endTime parameter passed into this method) will be calculated as:

$$
  v(t) = V_0 + (V_1 - V_0) \frac{t - T_0}{T_1 - T_0}
$$

Where \(V_0\) is the value at the time \(T_0\) and \(V_1\) is the value parameter passed into this method.

If there are no more events after this LinearRampToValue event then for \(t \geq T_1\), \(v(t) = V_1\).

If there is no event preceding this event, the linear ramp behaves as if setValueAtTime(value, currentTime) were called where value is the current value of the attribute and currentTime is the context currentTime at the time linearRampToValueAtTime is called.

If the preceding event is a SetTarget event, \(T_0\) and \(V_0\) are chosen from the current time and value of SetTarget automation. That is, if the SetTarget event has not started, \(T_0\) is the start time of the event, and \(V_0\) is the value just before the SetTarget event starts. In this case, the LinearRampToValue event effectively replaces the SetTarget event. If the SetTarget event has already started, \(T_0\) is the current context time, and \(V_0\) is the current SetTarget automation value at time \(T_0\). In both cases, the automation curve is continuous.

Arguments for the AudioParam.linearRampToValueAtTime() method.
Parameter	Type	Nullable	Optional	Description
`value`	float	✘	✘	The value the parameter will linearly ramp to at the given time.
`endTime`	double	✘	✘	The time in the same time coordinate system as the `AudioContext`'s `currentTime` attribute at which the automation ends. A `RangeError` exception MUST be thrown if `endTime` is negative or is not a finite number. If `endTime` is less than `currentTime`, it is clamped to `currentTime`.

Return type: AudioParam

setTargetAtTime(target, startTime, timeConstant)

Start exponentially approaching the target value at the given time with a rate having the given time constant. Among other uses, this is useful for implementing the "decay" and "release" portions of an ADSR envelope. Please note that the parameter value does not immediately change to the target value at the given time, but instead gradually changes to the target value.

During the time interval: \(T_0 \leq t\), where \(T_0\) is the startTime parameter:

$$
  v(t) = V_1 + (V_0 - V_1)\, e^{-\left(\frac{t - T_0}{\tau}\right)}
$$

where \(V_0\) is the initial value (the .value attribute) at \(T_0\) (the startTime parameter), \(V_1\) is equal to the target parameter, and \(\tau\) is the timeConstant parameter.

If a LinearRampToValue or ExponentialRampToValue event follows this event, the behavior is described in linearRampToValueAtTime() or exponentialRampToValueAtTime(), respectively. For all other events, the SetTarget event ends at the time of the next event.

Arguments for the AudioParam.setTargetAtTime() method.
Parameter	Type	Nullable	Optional	Description
`target`	float	✘	✘	The value the parameter will start changing to at the given time.
`startTime`	double	✘	✘	The time at which the exponential approach will begin, in the same time coordinate system as the `AudioContext`'s `currentTime` attribute. A `RangeError` exception MUST be thrown if `start` is negative or is not a finite number. If `startTime` is less than `currentTime`, it is clamped to `currentTime`.
`timeConstant`	float	✘	✘	The time-constant value of first-order filter (exponential) approach to the target value. The larger this value is, the slower the transition will be. The value MUST be non-negative or a `RangeError` exception MUST be thrown. If `timeConstant` is zero, the output value jumps immediately to the final value. More precisely, timeConstant is the time it takes a first-order linear continuous time-invariant system to reach the value \(1 - 1/e\) (around 63.2%) given a step input response (transition from 0 to 1 value).

Return type: AudioParam

setValueAtTime(value, startTime)

Schedules a parameter value change at the given time.

If there are no more events after this SetValue event, then for \(t \geq T_0\), \(v(t) = V\), where \(T_0\) is the startTime parameter and \(V\) is the value parameter. In other words, the value will remain constant.

If the next event (having time \(T_1\)) after this SetValue event is not of type LinearRampToValue or ExponentialRampToValue, then, for \(T_0 \leq t < T_1\):

$$
  v(t) = V
$$

In other words, the value will remain constant during this time interval, allowing the creation of "step" functions.

If the next event after this SetValue event is of type LinearRampToValue or ExponentialRampToValue then please see linearRampToValueAtTime() or exponentialRampToValueAtTime(), respectively.

Arguments for the AudioParam.setValueAtTime() method.
Parameter	Type	Nullable	Optional	Description
`value`	float	✘	✘	The value the parameter will change to at the given time.
`startTime`	double	✘	✘	The time in the same time coordinate system as the `BaseAudioContext`'s `currentTime` attribute at which the parameter changes to the given value. A `RangeError` exception MUST be thrown if `startTime` is negative or is not a finite number. If `startTime` is less than `currentTime`, it is clamped to `currentTime`.

Return type: AudioParam

setValueCurveAtTime(values, startTime, duration)

Sets an array of arbitrary parameter values starting at the given time for the given duration. The number of values will be scaled to fit into the desired duration.

Let \(T_0\) be startTime, \(T_D\) be duration, \(V\) be the values array, and \(N\) be the length of the values array. Then, during the time interval: \(T_0 \le t < T_0 + T_D\), let

$$
  \begin{align*} k &= \left\lfloor \frac{N - 1}{T_D}(t-T_0) \right\rfloor \\
  \end{align*}
$$

Then \(v(t)\) is computed by linearly interpolating between \(V[k]\) and \(V[k+1]\),

After the end of the curve time interval (\(t \ge T_0 + T_D\)), the value will remain constant at the final curve value, until there is another automation event (if any).

An implicit call to setValueAtTime() is made at time \(T_0 + T_D\) with value \(V[N-1]\) so that following automations will start from the end of the setValueCurveAtTime() event.

Arguments for the AudioParam.setValueCurveAtTime() method.
Parameter	Type	Nullable	Optional	Description
`values`	sequence<float>	✘	✘	A sequence of float values representing a parameter value curve. These values will apply starting at the given time and lasting for the given duration. When this method is called, an internal copy of the curve is created for automation purposes. Subsequent modifications of the contents of the passed-in array therefore have no effect on the `AudioParam`. An `InvalidStateError` MUST be thrown if this attribute is a `sequence<float>` object that has a length less than 2.
`startTime`	double	✘	✘	The start time in the same time coordinate system as the `AudioContext`'s `currentTime` attribute at which the value curve will be applied. A `RangeError` exception MUST be thrown if `startTime` is negative or is not a finite number. If `startTime` is less than `currentTime`, it is clamped to `currentTime`.
`duration`	double	✘	✘	The amount of time in seconds (after the `startTime` parameter) where values will be calculated according to the `values` parameter. A `RangeError` exception MUST be thrown if `duration` is not strictly positive or is not a finite number.

Return type: AudioParam

1.6.3. Computation of Value

There are two different kind of AudioParams, simple parameters and compound parameters. Simple parameters (the default) are used on their own to compute the final audio output of an AudioNode. Compound parameters are AudioParams that are used with other AudioParams to compute a value that is then used as an input to compute the output of an AudioNode.

The computedValue is the final value controlling the audio DSP and is computed by the audio rendering thread during each rendering time quantum. It MUST be internally computed as follows:

An intrinsic parameter value will be calculated at each time, which is either the value set directly to the value attribute, or, if there are any automation events with times before or at this time, the value as calculated from these events. When read, the value attribute always returns the intrinsic value for the current time. If automation events are removed from a given time range, then the intrinsic value will remain unchanged and stay at its previous value until either the value attribute is directly set, or automation events are added for the time range.
The computedValue is the sum of the intrinsic value and the value of the input AudioParam buffer. When read, the value attribute always returns the computedValue for the current time.
If this AudioParam is a compound parameter, compute its final value with other AudioParams.

The nominal range for a computedValue are the lower and higher values this parameter can effectively have. For simple parameters, the computedValue is clamped to the simple nominal range for this parameter. Compound parameters have their final value clamped to their nominal range after having been computed from the different AudioParam values they are composed of.

When automation methods are used, clamping is still applied. However, the automation is run as if there were no clamping at all. Only when the automation values are to be applied to the output is the clamping done as specified above.

For example, consider a node \(N\) has an AudioParam \(p\) with a nominal range of \([0, 1]\), and following automation sequence

N.p.setValueAtTime(0, 0);
N.p.linearRampToValueAtTime(4, 1);
N.p.linearRampToValueAtTime(0, 2);

The initial slope of the curve is 4, until it reaches the maximum value of 1, at which time, the output is held constant. Finally, near time 2, the slope of the curve is -4. This is illustrated in the graph below where the dashed line indicates what would have happened without clipping, and the solid line indicates the actual expected behavior of the audioparam due to clipping to the nominal range.

AudioParam automation clipping to nominal — An example of clipping of an AudioParam automation from the nominal range.

1.6.4. `AudioParam` Automation Example

var curveLength = 44100;var curve = new Float32Array(curveLength);for (var i = 0; i < curveLength; ++i)  curve[i] = Math.sin(Math.PI * i / curveLength);var t0 = 0;var t1 = 0.1;var t2 = 0.2;var t3 = 0.3;var t4 = 0.325;var t5 = 0.5;var t6 = 0.6;var t7 = 0.7;var t8 = 1.0;var timeConstant = 0.1;param.setValueAtTime(0.2, t0);param.setValueAtTime(0.3, t1);param.setValueAtTime(0.4, t2);param.linearRampToValueAtTime(1, t3);param.linearRampToValueAtTime(0.8, t4);param.setTargetAtTime(.5, t4, timeConstant);// Compute where the setTargetAtTime will be at time t5 so we can make// the following exponential start at the right point so there’s no// jump discontinuity. From the spec, we have// v(t) = 0.5 + (0.8 - 0.5)*exp(-(t-t4)/timeConstant)// Thus v(t5) = 0.5 + (0.8 - 0.5)*exp(-(t5-t4)/timeConstant)param.setValueAtTime(0.5 + (0.8 - 0.5)*Math.exp(-(t5 - t4)/timeConstant), t5);param.exponentialRampToValueAtTime(0.75, t6);param.exponentialRampToValueAtTime(0.05, t7);param.setValueCurveAtTime(curve, t7, t8 - t7);

1.7. The `AudioScheduledSourceNode` Interface

The interface represents the common features of the source nodes such as AudioBufferSourceNode, ConstantSourceNode, and OscillatorNode.

Before a source is started (by calling start(), the source node MUST output silence (0). After a source has been stopped (by calling stop()), the source MUST then output silence (0).

AudioScheduledSourceNode cannot be instantiated directly, but is instead extended by the concrete interfaces for the source nodes.

[Exposed=Window]
interface AudioScheduledSourceNode : AudioNode {
  attribute EventHandler onended;
  void start(optional double when = 0);
  void stop(optional double when = 0);
};

1.7.1. Attributes

onended, of type EventHandler

A property used to set the EventHandler (described in HTML[HTML]) for the ended event that is dispatched to AudioScheduledSourceNode node types. When the source node has stopped playing (as determined by the concrete node), an event of type Event (described in HTML [HTML]) will be dispatched to the event handler.

For all AudioScheduledSourceNodes, the onended event is dispatched when the stop time determined by stop() is reached. For an AudioBufferSourceNode, the event is also dispatched because the duration has been reached or if the entire buffer has been played.

1.7.2. Methods

start(when)

Schedules a sound to playback at an exact time.

When this method is called, execute these steps:

If stop() has been called on this node, or if an earlier call to start() has already occurred, an InvalidStateError exception MUST be thrown.
Check for any errors that must be thrown due to parameter constraints described below.
Queue a control message to start the AudioScheduledSourceNode, including the parameter values in the messsage.

Arguments for the AudioScheduledSourceNode.start(when) method.
Parameter	Type	Nullable	Optional	Description
`when`	double	✘	✔	The `when` parameter describes at what time (in seconds) the sound should start playing. It is in the same time coordinate system as the `AudioContext`'s `currentTime` attribute. When the signal emitted by the `AudioScheduledSourceNode` depends on the sound’s start time, the exact value of `when` is always used without rounding to the nearest sample frame. If 0 is passed in for this value or if the value is less than currentTime, then the sound will start playing immediately. A `RangeError` exception MUST be thrown if `when` is negative.

Return type: void

stop(when)

Schedules a sound to stop playback at an exact time. If stop is called again after already having been called, the last invocation will be the only one applied; stop times set by previous calls will not be applied, unless the buffer has already stopped prior to any subsequent calls. If the buffer has already stopped, further calls to stop will have no effect. If a stop time is reached prior to the scheduled start time, the sound will not play.

When this method is called, execute these steps:

If an earlier call to start has not already occurred, an InvalidStateError exception MUST be thrown.
Check for any errors that must be thrown due to parameter constraints described below.
Queue a control message to stop the AudioScheduledSourceNode, including the parameter values in the messsage.

If the node is an AudioBufferSourceNode, running a control message to stop the AudioBufferSourceNode means invoking the handleStop() function in the playback algorithm.

Arguments for the AudioScheduledSourceNode.stop(when) method.
Parameter	Type	Nullable	Optional	Description
`when`	double	✘	✔	The `when` parameter describes at what time (in seconds) the source should stop playing. It is in the same time coordinate system as the `AudioContext`'s `currentTime` attribute. If 0 is passed in for this value or if the value is less than `currentTime`, then the sound will stop playing immediately. A `RangeError` exception MUST be thrown if `when` is negative.

Return type: void

1.8. The `AnalyserNode` Interface

This interface represents a node which is able to provide real-time frequency and time-domain analysis information. The audio stream will be passed un-processed from input to output.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1	This output may be left unconnected.
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	No

[Exposed=Window,
 Constructor (BaseAudioContext context, optional AnalyserOptions options)]
interface AnalyserNode : AudioNode {
  void getFloatFrequencyData (Float32Array array);
  void getByteFrequencyData (Uint8Array array);
  void getFloatTimeDomainData (Float32Array array);
  void getByteTimeDomainData (Uint8Array array);
  attribute unsigned long fftSize;
  readonly attribute unsigned long frequencyBinCount;
  attribute double minDecibels;
  attribute double maxDecibels;
  attribute double smoothingTimeConstant;
};

1.8.1. Constructors

AnalyserNode(context, options)

Arguments for the AnalyserNode.AnalyserNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `AnalyserNode` will be associated with.
`options`	AnalyserOptions	✘	✔	Optional initial parameter value for this `AnalyserNode`.

1.8.2. Attributes

fftSize, of type unsigned long

The size of the FFT used for frequency-domain analysis (in sample-frames). This MUST be a power of two in the range 32 to 32768, otherwise an IndexSizeError exception MUST be thrown. The default value is 2048. Note that large FFT sizes can be costly to compute.

If the fftSize is changed to a different value, then all state associated with smoothing of the frequency data (for getByteFrequencyData() and getFloatFrequencyData()) is reset. That is the previous block, \(\hat{X}_{-1}[k]\), used for smoothing over time is set to 0 for all \(k\).

Note that increasing fftSize does mean that the current time-domain data must be expanded to include past frames that it previously did not. This means that the AnalyserNode effectively MUST keep around the last 32768 sample- frames and the current time-domain data is the most recent fftSize sample-frames out of that.

frequencyBinCount, of type unsigned long, readonly

Half the FFT size.

maxDecibels, of type double

maxDecibels is the maximum power value in the scaling range for the FFT analysis data for conversion to unsigned byte values. The default value is -30. If the value of this attribute is set to a value less than or equal to minDecibels, an IndexSizeError exception MUST be thrown.

minDecibels, of type double

minDecibels is the minimum power value in the scaling range for the FFT analysis data for conversion to unsigned byte values. The default value is -100. If the value of this attribute is set to a value more than or equal to maxDecibels, an IndexSizeError exception MUST be thrown.

smoothingTimeConstant, of type double

A value from 0 -> 1 where 0 represents no time averaging with the last analysis frame. The default value is 0.8. If the value of this attribute is set to a value less than 0 or more than 1, an IndexSizeError exception MUST be thrown.

1.8.3. Methods

getByteFrequencyData(array)

Copies the current frequency data into the passed unsigned byte array. If the array has fewer elements than the frequencyBinCount, the excess elements will be dropped. If the array has more elements than the frequencyBinCount, the excess elements will be ignored. The most recent fftSize frames are used in computing the frequency data.

If another call to getByteFrequencyData() or getFloatFrequencyData() occurs within the same render quantum as a previous call, the current frequency data is not updated with the same data. Instead, the previously computed data is returned.

The values stored in the unsigned byte array are computed in the following way. Let \(Y[k]\) be the current frequency data as described in FFT windowing and smoothing. Then the byte value, \(b[k]\), is

$$
  b[k] = \left\lfloor
      \frac{255}{\mbox{dB}_{max} - \mbox{dB}_{min}}
      \left(Y[k] - \mbox{dB}_{min}\right)
    \right\rfloor
$$

where \(\mbox{dB}_{min}\) is minDecibels and \(\mbox{dB}_{max}\) is maxDecibels. If \(b[k]\) lies outside the range of 0 to 255, \(b[k]\) is clipped to lie in that range.

Arguments for the AnalyserNode.getByteFrequencyData() method.
Parameter	Type	Nullable	Optional	Description
`array`	Uint8Array	✘	✘	This parameter is where the frequency-domain analysis data will be copied.

Return type: void

getByteTimeDomainData(array)

Copies the current time-domain data (waveform data) into the passed unsigned byte array. If the array has fewer elements than the value of fftSize, the excess elements will be dropped. If the array has more elements than fftSize, the excess elements will be ignored. The most recent fftSize frames are used in computing the byte data.

The values stored in the unsigned byte array are computed in the following way. Let \(x[k]\) be the time-domain data. Then the byte value, \(b[k]\), is

$$
  b[k] = \left\lfloor 128(1 + x[k]) \right\rfloor.
$$

If \(b[k]\) lies outside the range 0 to 255, \(b[k]\) is clipped to lie in that range.

Arguments for the AnalyserNode.getByteTimeDomainData() method.
Parameter	Type	Nullable	Optional	Description
`array`	Uint8Array	✘	✘	This parameter is where the time-domain sample data will be copied.

Return type: void

getFloatFrequencyData(array)

Copies the current frequency data into the passed floating-point array. If the array has fewer elements than the frequencyBinCount, the excess elements will be dropped. If the array has more elements than the frequencyBinCount, the excess elements will be ignored. The most recent fftSize frames are used in computing the frequency data.

If another call to getFloatFrequencyData() or getByteFrequencyData() occurs within the same render quantum as a previous call, the current frequency data is not updated with the same data. Instead, the previously computed data is returned.

The frequency data are in dB units.

Arguments for the AnalyserNode.getFloatFrequencyData() method.
Parameter	Type	Nullable	Optional	Description
`array`	Float32Array	✘	✘	This parameter is where the frequency-domain analysis data will be copied.

Return type: void

getFloatTimeDomainData(array)

Copies the current time-domain data (waveform data) into the passed floating-point array. If the array has fewer elements than the value of fftSize, the excess elements will be dropped. If the array has more elements than fftSize, the excess elements will be ignored. The most recent fftSize frames are returned (after downmixing).

Arguments for the AnalyserNode.getFloatTimeDomainData() method.
Parameter	Type	Nullable	Optional	Description
`array`	Float32Array	✘	✘	This parameter is where the time-domain sample data will be copied.

Return type: void

1.8.4. `AnalyserOptions`

This specifies the options to be used when constructing an AnalyserNode. All members are optional; if not specified, the normal default values are used to construct the node.

dictionary AnalyserOptions : AudioNodeOptions {
  unsigned long fftSize = 2048;
  double maxDecibels = -30;
  double minDecibels = -100;
  double smoothingTimeConstant = 0.8;
};

1.8.4.1. Dictionary `AnalyserOptions` Members

fftSize, of type unsigned long, defaulting to 2048: The desired initial size of the FFT for frequency-domain analysis.
maxDecibels, of type double, defaulting to -30: The desired initial maximum power in dB for FFT analysis.
minDecibels, of type double, defaulting to -100: The desired initial minimum power in dB for FFT analysis.
smoothingTimeConstant, of type double, defaulting to 0.8: The desired initial smoothing constant for the FFT analysis.

1.8.5. Time-Domain Down-Mixing

When the current time-domain data are computed, the input signal must be down-mixed to mono as if channelCount is 1, channelCountMode is "max" and channelInterpretation is "speakers". This is independent of the settings for the AnalyserNode itself. The most recent fftSize frames are used for the down-mixing operation.

1.8.6. FFT Windowing and Smoothing over Time

When the current frequency data are computed, the following operations are to be performed:

Compute the current time-domain data.
Apply a Blackman window to the time domain input data.
Apply a Fourier transform to the windowed time domain input data to get imaginary and real frequency data.
Smooth over time the frequency domain data.
Conversion to dB.

In the following, let \(N\) be the value of the fftSize attribute of this AnalyserNode.

Applying a Blackman window consists in the following operation on the input time domain data. Let \(x[n]\) for \(n = 0, \ldots, N - 1\) be the time domain data. The Blackman window is defined by

$$
\begin{align*}
  \alpha &= \mbox{0.16} \\ a_0 &= \frac{1-\alpha}{2} \\
  a_1 &= \frac{1}{2} \\
  a_2 &= \frac{\alpha}{2} \\
  w[n] &= a_0 - a_1 \cos\frac{2\pi n}{N} + a_2 \cos\frac{4\pi n}{N}, \mbox{ for } n = 0, \ldots, N - 1
\end{align*}
$$

The windowed signal \(\hat{x}[n]\) is

$$
  \hat{x}[n] = x[n] w[n], \mbox{ for } n = 0, \ldots, N - 1
$$

Applying a Fourier transform consists of computing the Fourier transform in the following way. Let \(X[k]\) be the complex frequency domain data and \(\hat{x}[n]\) be the windowed time domain data computed above. Then

$$
  X[k] = \frac{1}{N} \sum_{n = 0}^{N - 1} \hat{x}[n]\, e^{\frac{-2\pi i k n}{N}}
$$

for \(k = 0, \dots, N/2-1\).

Smoothing over time frequency data consists in the following operation:

Let \(\hat{X}_{-1}[k]\) be the result of this operation on the previous block. The previous block is defined as being the buffer computed by the previous smoothing over time operation, or an array of \(N\) zeros if this is the first time we are smoothing over time.
Let \(\tau\) be the value of the smoothingTimeConstant attribute for this AnalyserNode.
Let \(X[k]\) be the result of applying a Fourier transform of the current block.

Then the smoothed value, \(\hat{X}[k]\), is computed by

$$
  \hat{X}[k] = \tau\, \hat{X}_{-1}[k] + (1 - \tau)\, |X[k]|
$$

for \(k = 0, \ldots, N - 1\).

Conversion to dB consists of the following operation, where \(\hat{X}[k]\) is computed in smoothing over time:

$$
  Y[k] = 20\log_{10}\hat{X}[k]
$$

for \(k = 0, \ldots, N-1\).

This array, \(Y[k]\), is copied to the output array for getFloatFrequencyData(). For getByteFrequencyData(), the \(Y[k]\) is clipped to lie between minDecibels and maxDecibels and then scaled to fit in an unsigned byte such that minDecibels is represented by the value 0 and maxDecibels is represented by the value 255.

1.9. The `AudioBufferSourceNode` Interface

This interface represents an audio source from an in-memory audio asset in an AudioBuffer. It is useful for playing audio assets which require a high degree of scheduling flexibility and accuracy. If sample-accurate playback of network- or disk-backed assets is required, an implementer should use AudioWorkletNode to implement playback.

The start() method is used to schedule when sound playback will happen. The start() method may not be issued multiple times. The playback will stop automatically when the buffer’s audio data has been completely played (if the loop attribute is false), or when the stop() method has been called and the specified time has been reached. Please see more details in the start() and stop() description.

Property	Value	Notes
`numberOfInputs`	0
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	No

The number of channels of the output always equals the number of channels of the AudioBuffer assigned to the buffer attribute, or is one channel of silence if buffer is null.

A playhead position for an AudioBufferSourceNode is defined as any quantity representing a time offset in seconds, relative to the time coordinate of the first sample frame in the buffer. Such values are to be considered independently from the node’s playbackRate and detune parameters. In general, playhead positions may be subsample-accurate and need not refer to exact sample frame positions. They may assume valid values between 0 and the duration of the buffer.

The playbackRate and detune attributes form a compound parameter. They are used together to determine a computedPlaybackRate value:

computedPlaybackRate(t) = playbackRate(t) * pow(2, detune(t) / 1200)

The nominal range for this compound parameter is \((-\infty, \infty)\).

AudioBufferSourceNodes are created with an internal boolean slot [[buffer set]], initially set to false.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional AudioBufferSourceOptions options)]
interface AudioBufferSourceNode : AudioScheduledSourceNode {
  attribute AudioBuffer? buffer;
  readonly attribute AudioParam playbackRate;
  readonly attribute AudioParam detune;
  attribute boolean loop;
  attribute double loopStart;
  attribute double loopEnd;
  void start (optional double when = 0,
              optional double offset,
              optional double duration);
};

1.9.1. Constructors

AudioBufferSourceNode(context, options)

Let node be a new AudioBufferSourceNode object. Initialize node, and return node.

Arguments for the AudioBufferSourceNode.AudioBufferSourceNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `AudioBufferSourceNode` will be associated with.
`options`	AudioBufferSourceOptions	✘	✔	Optional initial parameter value for this `AudioBufferSourceNode`.

1.9.2. Attributes

buffer, of type AudioBuffer, nullable

Represents the audio asset to be played.

To set the buffer attribute, execute these steps:

Let new buffer be the AudioBuffer or null value to be assigned to buffer.
If new buffer is not null and [[buffer set]] is true, throw an InvalidStateError and abort these steps.
If new buffer is not null, set [[buffer set]] to true.
Assign new buffer to the buffer attribute.
If start() has previously been called on this node, perform the operation acquire the content on buffer.

detune, of type AudioParam, readonly

An additional parameter, in cents, to modulate the speed at which is rendered the audio stream. This parameter is a compound parameter with playbackRate to form a computedPlaybackRate.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`k-rate`"	Has automation rate constraints

loop, of type boolean

Indicates if the region of audio data designated by loopStart and loopEnd should be played continuously in a loop. The default value is false.

loopEnd, of type double

An optional playhead position where looping should end if the loop attribute is true. Its value is exclusive of the content of the loop. Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer. If loopEnd is less than or equal to 0, or if loopEnd is greater than the duration of the buffer, looping will end at the end of the buffer.

loopStart, of type double

An optional playhead position where looping should begin if the loop attribute is true. Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer. If loopStart is less than 0, looping will begin at 0. If loopStart is greater than the duration of the buffer, looping will begin at the end of the buffer.

playbackRate, of type AudioParam, readonly

The speed at which to render the audio stream. This is a compound parameter with detune to form a computedPlaybackRate.

Parameter	Value	Notes
`defaultValue`	1
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`k-rate`"	Has automation rate constraints

1.9.3. Methods

start(when, offset, duration)

Schedules a sound to playback at an exact time.

When this method is called, execute these steps:

If stop has been called on this node, or if an earlier call to start has already occurred, an InvalidStateError exception MUST be thrown.
Check for any errors that must be thrown due to parameter constraints described below.
Queue a control message to start the AudioBufferSourceNode, including the parameter values in the messsage.

Running a control message to start the AudioBufferSourceNode means invoking the handleStart() function in the playback algorithm which follows.

Arguments for the AudioBufferSourceNode.start(when, offset, duration) method.
Parameter	Type	Nullable	Optional	Description
`when`	double	✘	✔	The `when` parameter describes at what time (in seconds) the sound should start playing. It is in the same time coordinate system as the `AudioContext`'s `currentTime` attribute. If 0 is passed in for this value or if the value is less than currentTime, then the sound will start playing immediately. A `RangeError` exception MUST be thrown if `when` is negative.
`offset`	double	✘	✔	The `offset` parameter supplies a playhead position where playback will begin. If 0 is passed in for this value, then playback will start from the beginning of the buffer. A `RangeError` exception MUST be thrown if `offset` is negative. If `offset` is greater than `loopEnd`, playback will begin at `loopEnd` (and immediately loop to `loopStart`). `offset` is silently clamped to [0, `duration`], when `startTime` is reached, where `duration` is the value of the `duration` attribute of the `AudioBuffer` set to the `buffer` attribute of this `AudioBufferSourceNode`.
`duration`	double	✘	✔	The `duration` parameter describes the duration of sound to be played, expressed as seconds of total buffer content to be output, including any whole or partial loop iterations. The units of `duration` are independent of the effects of `playbackRate`. For example, a `duration` of 5 seconds with a playback rate of 0.5 will output 5 seconds of buffer content at half speed, producing 10 seconds of audible output. A `RangeError` exception MUST be thrown if `duration` is negative.

Return type: void

1.9.4. `AudioBufferSourceOptions`

This specifies options for constructing a AudioBufferSourceNode. All members are optional; if not specified, the normal default is used in constructing the node.

dictionary AudioBufferSourceOptions {
  AudioBuffer? buffer;
  float detune = 0;
  boolean loop = false;
  double loopEnd = 0;
  double loopStart = 0;
  float playbackRate = 1;
};

1.9.4.1. Dictionary `AudioBufferSourceOptions` Members

buffer, of type AudioBuffer, nullable: The audio asset to be played. This is equivalent to assigning buffer to the buffer attribute of the AudioBufferSourceNode.
detune, of type float, defaulting to 0: The initial value for the detune AudioParam.
loop, of type boolean, defaulting to false: The initial value for the loop attribute.
loopEnd, of type double, defaulting to 0: The initial value for the loopEnd attribute.
loopStart, of type double, defaulting to 0: The initial value for the loopStart attribute.
playbackRate, of type float, defaulting to 1: The initial value for the playbackRate AudioParam.

1.9.5. Looping

This section is non-normative. Please see the playback algorithm for normative requirements.

Setting the loop attribute to true causes playback of the region of the buffer defined by the endpoints loopStart and loopEnd to continue indefinitely, once any part of the looped region has been played. While loop remains true, looped playback will continue until one of the following occurs:

stop() is called,
the scheduled stop time has been reached,
the duration has been exceeded, if start() was called with a duration value.

The body of the loop is considered to occupy a region from loopStart up to, but not including, loopEnd. The direction of playback of the looped region respects the sign of the node’s playback rate. For positive playback rates, looping occurs from loopStart to loopEnd; for negative rates, looping occurs from loopEnd to loopStart.

Looping does not affect the interpretation of the offset argument of start(). Playback always starts at the requested offset, and looping only begins once the body of the loop is encountered during playback.

The effective loop start and end points are required to lie within the range of zero and the buffer duration, as specified in the algorithm below. loopEnd is further constrained to be at or after loopStart. If any of these constraints are violated, the loop is considered to include the entire buffer contents.

Loop endpoints have subsample accuracy. When endpoints do not fall on exact sample frame offsets, or when the playback rate is not equal to 1, playback of the loop is interpolated to splice the beginning and end of the loop together just as if the looped audio occurred in sequential, non-looped regions of the buffer.

Loop-related properties may be varied during playback of the buffer, and in general take effect on the next rendering quantum. The exact results are defined by the normative playback algorithm which follows.

The default values of the loopStart and loopEnd attributes are both 0. Since a loopEnd value of zero is equivalent to the length of the buffer, the default endpoints cause the entire buffer to be included in the loop.

Note that the values of the loop endpoints are expressed as time offsets in terms of the sample rate of the buffer, meaning that these values are independent of the node’s playbackRate parameter which can vary dynamically during the course of playback.

1.9.6. Playback of AudioBuffer Contents

This normative section specifies the playback of the contents of the buffer, accounting for the fact that playback is influenced by the following factors working in combination, which can vary dynamically during playback:

A starting offset, which can expressed with sub-sample precision.
Loop points, which can expressed with sub-sample precision.
Playback rate and detuning parameters, which combine to yield a single computed playback rate that can assume non-infinite values which may be positive or negative.

The algorithm to be followed internally to generate output from an AudioBufferSourceNode conforms to the following principles:

Resampling of the buffer may be performed arbitrarily by the UA at any desired point to increase the efficiency or quality of the output.
Sub-sample start offsets or loop points may require additional interpolation between sample frames.
The playback of a looped buffer should behave identically to an unlooped buffer containing consecutive occurrences of the looped audio content, excluding any effects from interpolation.

The description of the algorithm is as follows:

let buffer; // AudioBuffer employed by this nodelet context; // AudioContext employed by this node// The following variables capture attribute and AudioParam values for the node.// They are updated on a k-rate basis, prior to each invocation of process().let loop;let detune;let loopStart;let loopEnd;let playbackRate;// Variables for the node’s playback parameterslet start = 0, offset = 0, duration = Infinity; // Set by start()let stop = Infinity; // Set by stop()// Variables for tracking node’s playback statelet bufferTime = 0, started = false, enteredLoop = false;let bufferTimeElapsed = 0;let dt = 1 / context.sampleRate;// Handle invocation of start method callfunction handleStart(when, pos, dur) {  if (arguments.length >= 1) {    start = when;  }  offset = pos;  if (arguments.length >= 3) {    duration = dur;  }}// Handle invocation of stop method callfunction handleStop(when) {  if (arguments.length >= 1) {    stop = when;  } else {    stop = context.currentTime;  }}// Interpolate a multi-channel signal value for some sample frame.// Returns an array of signal values.function playbackSignal(position) {  /*    This function provides the playback signal function for buffer, which is a    function that maps from a playhead position to a set of output signal    values, one for each output channel. If |position| corresponds to the    location of an exact sample frame in the buffer, this function returns    that frame. Otherwise, its return value is determined by a UA-supplied    algorithm that interpolates between sample frames in the neighborhood of    position.    If position is greater than or equal to loopEnd and there is no subsequent    sample frame in buffer, then interpolation should be based on the sequence    of subsequent frames beginning at loopStart.   */   ...}// Generate a single render quantum of audio to be placed// in the channel arrays defined by output. Returns an array// of |numberOfFrames| sample frames to be output.function process(numberOfFrames) {  let currentTime = context.currentTime; // context time of next rendered frame  let output = []; // accumulates rendered sample frames  // Combine the two k-rate parameters affecting playback rate  let computedPlaybackRate = playbackRate * Math.pow(2, detune / 1200);  // Determine loop endpoints as applicable  let actualLoopStart, actualLoopEnd;  if (loop && buffer != null) {    if (loopStart >= 0 && loopEnd > 0 && loopStart < loopEnd) {      actualLoopStart = loopStart;      actualLoopEnd = Math.min(loopEnd, buffer.duration);    } else {      actualLoopStart = 0;      actualLoopEnd = buffer.duration;    }  } else {    // If the loop flag is false, remove any record of the loop having been entered    enteredLoop = false;  }  // Handle null buffer case  if (buffer == null) {    stop = currentTime; // force zero output for all time  }  // Render each sample frame in the quantum  for (let index = 0; index < numberOfFrames; index++) {    // Check that currentTime and bufferTimeElapsed are    // within allowable range for playback    if (currentTime < start || currentTime >= stop || bufferTimeElapsed >= duration) {      output.push(0); // this sample frame is silent      currentTime += dt;      continue;    }    if (!started) {      // Take note that buffer has started playing and get initial      // playhead position.      bufferTime = offset;      started = true;    }    // Handle loop-related calculations    if (loop) {      // Determine if looped portion has been entered for the first time      if (!enteredLoop) {        if (offset < actualLoopEnd && bufferTime >= actualLoopStart) {          // playback began before or within loop, and playhead is          // now past loop start          enteredLoop = true;        }        if (offset >= actualLoopEnd && bufferTime < actualLoopEnd) {          // playback began after loop, and playhead is now prior          // to the loop end          enteredLoop = true;        }      }      // Wrap loop iterations as needed. Note that enteredLoop      // may become true inside the preceding conditional.      if (enteredLoop) {        while (bufferTime >= actualLoopEnd) {          bufferTime -= actualLoopEnd - actualLoopStart;        }        while (bufferTime < actualLoopStart) {          bufferTime += actualLoopEnd - actualLoopStart;        }      }    }    if (bufferTime >= 0 && bufferTime < buffer.duration) {      output.push(playbackSignal(bufferTime));    } else {      output.push(0); // past end of buffer, so output silent frame    }    bufferTime += dt * computedPlaybackRate;    bufferTimeElapsed += dt * computedPlaybackRate;    currentTime += dt;  } // End of render quantum loop  if (currentTime >= stop) {    // end playback state of this node.    // no further invocations of process() will occur.  }  return output;}

The following non-normative figures illustrate the behavior of the algorithm in assorted key scenarios. Dynamic resampling of the buffer is not considered, but as long as the times of loop positions are not changed this does not materially affect the resulting playback. In all figures, the following conventions apply:

context sample rate is 1000 Hz
AudioBuffer content is shown with the first sample frame at the x origin.
output signals are shown with the sample frame located at time start at the x origin.
linear interpolation is depicted throughout, although a UA could employ other interpolation techniques.
the duration values noted in the figures refer to the buffer, not arguments to start()

This figure illustrates basic playback of a buffer, with a simple loop that ends after the last sample frame in the buffer:

AudioBufferSourceNode basic playback — `AudioBufferSourceNode` basic playback

This figure illustrates playbackRate interpolation, showing half-speed playback of buffer contents in which every other output sample frame is interpolated. Of particular note is the last sample frame in the looped output, which is interpolated using the loop start point:

AudioBufferSourceNode playbackRate interpolation — `AudioBufferSourceNode` playbackRate interpolation

This figure illustrates sample rate interpolation, showing playback of a buffer whose sample rate is 50% of the context sample rate, resulting in a computed playback rate of 0.5 that corrects for the difference in sample rate between the buffer and the context. The resulting output is the same as the preceding example, but for different reasons.

AudioBufferSourceNode sample rate interpolation — `AudioBufferSourceNode` sample rate interpolation.

This figure illustrates subsample offset playback, in which the offset within the buffer begins at exactly half a sample frame. Consequently, every output frame is interpolated:

AudioBufferSourceNode subsample offset playback — `AudioBufferSourceNode` subsample offset playback

This figure illustrates subsample loop playback, showing how fractional frame offsets in the loop endpoints map to interpolated data points in the buffer that respect these offsets as if they were references to exact sample frames:

AudioBufferSourceNode subsample loop playback — `AudioBufferSourceNode` subsample loop playback

1.10. The `AudioDestinationNode` Interface

This is an AudioNode representing the final audio destination and is what the user will ultimately hear. It can often be considered as an audio output device which is connected to speakers. All rendered audio to be heard will be routed to this node, a "terminal" node in the AudioContext's routing graph. There is only a single AudioDestinationNode per AudioContext, provided through the destination attribute of AudioContext.

The output of a AudioDestinationNode is produced by summing its input, allowing to capture the output of an AudioContext into, for example, a MediaStreamAudioDestinationNode, or a MediaRecorder (described in [mediastream-recording]).

The AudioDestinationNode can be either the destination of an AudioContext or OfflineAudioContext, and the channel properties depend on what the context is.

For an AudioContext, the defaults are

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`explicit`"
`channelInterpretation`	"`speakers`"
tail-time reference	No

The channelCount can be set to any value less than or equal to maxChannelCount. An IndexSizeError exception MUST be thrown if this value is not within the valid range. Giving a concrete example, if the audio hardware supports 8-channel output, then we may set channelCount to 8, and render 8 channels of output.

For an OfflineAudioContext, the defaults are

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	numberOfChannels
`channelCountMode`	"`explicit`"
`channelInterpretation`	"`speakers`"
tail-time reference	No

where numberOfChannels is the number of channels specified when constructing the OfflineAudioContext. This value may not be changed; a NotSupportedError exception MUST be thrown if channelCount is changed to a different value.

[Exposed=Window]
interface AudioDestinationNode : AudioNode {
  readonly attribute unsigned long maxChannelCount;
};

1.10.1. Attributes

maxChannelCount, of type unsigned long, readonly: The maximum number of channels that the channelCount attribute can be set to. An AudioDestinationNode representing the audio hardware end-point (the normal case) can potentially output more than 2 channels of audio if the audio hardware is multi-channel. maxChannelCount is the maximum number of channels that this hardware is capable of supporting.

1.11. The `AudioListener` Interface

This interface represents the position and orientation of the person listening to the audio scene. All PannerNode objects spatialize in relation to the BaseAudioContext's listener. See §7 Spatialization/Panning for more details about spatialization.

The positionX, positionY, positionZ parameters represent the location of the listener in 3D Cartesian coordinate space. PannerNode objects use this position relative to individual audio sources for spatialization.

The forwardX, forwardY, forwardZ parameters represent a direction vector in 3D space. Both a forward vector and an up vector are used to determine the orientation of the listener. In simple human terms, the forward vector represents which direction the person’s nose is pointing. The up vector represents the direction the top of a person’s head is pointing. These two vectors are expected to be linearly independent. For normative requirements of how these values are to be interpreted, see the §7 Spatialization/Panning section.

[Exposed=Window]
interface AudioListener {
  readonly attribute AudioParam positionX;
  readonly attribute AudioParam positionY;
  readonly attribute AudioParam positionZ;
  readonly attribute AudioParam forwardX;
  readonly attribute AudioParam forwardY;
  readonly attribute AudioParam forwardZ;
  readonly attribute AudioParam upX;
  readonly attribute AudioParam upY;
  readonly attribute AudioParam upZ;
  void setPosition (float x, float y, float z);
  void setOrientation (float x, float y, float z, float xUp, float yUp, float zUp);
};

1.11.1. Attributes

forwardX, of type AudioParam, readonly

Sets the x coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

forwardY, of type AudioParam, readonly

Sets the y coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

forwardZ, of type AudioParam, readonly

Sets the z coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	-1
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

positionX, of type AudioParam, readonly

Sets the x coordinate position of the audio listener in a 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

positionY, of type AudioParam, readonly

Sets the y coordinate position of the audio listener in a 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

positionZ, of type AudioParam, readonly

Sets the z coordinate position of the audio listener in a 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

upX, of type AudioParam, readonly

Sets the x coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

upY, of type AudioParam, readonly

Sets the y coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	1
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

upZ, of type AudioParam, readonly

Sets the z coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

1.11.2. Methods

setOrientation(x, y, z, xUp, yUp, zUp)

This method is DEPRECATED. It is equivalent to setting forwardX.value, forwardY.value, forwardZ.value, upX.value, upY.value, and upZ.value directly with the given x, y, z, xUp, yUp, and zUp values, respectively.

Consequently, if any of the forwardX, forwardY, forwardZ, upX, upY and upZ AudioParams have an automation curve set using setValueCurveAtTime() at the time this method is called, a NotSupportedError MUST be thrown.

Describes which direction the listener is pointing in the 3D cartesian coordinate space. Both a front vector and an up vector are provided. In simple human terms, the front vector represents which direction the person’s nose is pointing. The up vector represents the direction the top of a person’s head is pointing. These two vectors are expected to be linearly independent. For normative requirements of how these values are to be interpreted, see the §7 Spatialization/Panning.

The x, y, z parameters represent a front direction vector in 3D space, with the default value being (0,0,-1).

The xUp, yUp, zUp parameters represent an up direction vector in 3D space, with the default value being (0,1,0).

Arguments for the AudioListener.setOrientation() method.
Parameter	Type	Nullable	Optional
`x`	float	✘	✘
`y`	float	✘	✘
`z`	float	✘	✘
`xUp`	float	✘	✘
`yUp`	float	✘	✘
`zUp`	float	✘	✘

Return type: void

setPosition(x, y, z)

This method is DEPRECATED. It is equivalent to setting positionX.value, positionY.value, and positionZ.value directly with the given x, y, and z values, respectively.

Consequently, any of the positionX, positionY, and positionZ AudioParams for this AudioListener have an automation curve set using setValueCurveAtTime() at the time this method is called, a NotSupportedError MUST be thrown.

Sets the position of the listener in a 3D cartesian coordinate space. PannerNode objects use this position relative to individual audio sources for spatialization.

The x, y, z parameters represent the coordinates in 3D space.

The default value is (0,0,0).

Arguments for the AudioListener.setPosition() method.
Parameter	Type	Nullable	Optional
`x`	float	✘	✘
`y`	float	✘	✘
`z`	float	✘	✘

1.12. The `AudioProcessingEvent` Interface - DEPRECATED

This is an Event object which is dispatched to ScriptProcessorNode nodes. It will be removed when the ScriptProcessorNode is removed, as the replacement AudioWorkletNode uses a different approach.

The event handler processes audio from the input (if any) by accessing the audio data from the inputBuffer attribute. The audio data which is the result of the processing (or the synthesized data if there are no inputs) is then placed into the outputBuffer.

[Exposed=Window,
 Constructor (DOMString type, AudioProcessingEventInit eventInitDict)]
interface AudioProcessingEvent : Event {
  readonly attribute double playbackTime;
  readonly attribute AudioBuffer inputBuffer;
  readonly attribute AudioBuffer outputBuffer;
};

1.12.1. Attributes

inputBuffer, of type AudioBuffer, readonly: An AudioBuffer containing the input audio data. It will have a number of channels equal to the numberOfInputChannels parameter of the createScriptProcessor() method. This AudioBuffer is only valid while in the scope of the onaudioprocess function. Its values will be meaningless outside of this scope.
outputBuffer, of type AudioBuffer, readonly: An AudioBuffer where the output audio data MUST be written. It will have a number of channels equal to the numberOfOutputChannels parameter of the createScriptProcessor() method. Script code within the scope of the onaudioprocess function is expected to modify the Float32Array arrays representing channel data in this AudioBuffer. Any script modifications to this AudioBuffer outside of this scope will not produce any audible effects.
playbackTime, of type double, readonly: The time when the audio will be played in the same time coordinate system as the AudioContext's currentTime.

1.12.2. `AudioProcessingEventInit`

dictionary AudioProcessingEventInit : EventInit {
  required double playbackTime;
  required AudioBuffer inputBuffer;
  required AudioBuffer outputBuffer;
};

1.12.2.1. Dictionary `AudioProcessingEventInit` Members

inputBuffer, of type AudioBuffer: Value to be assigned to the inputBuffer attribute of the event.
outputBuffer, of type AudioBuffer: Value to be assigned to the outputBuffer attribute of the event.
playbackTime, of type double: Value to be assigned to the playbackTime attribute of the event.

1.13. The `BiquadFilterNode` Interface

BiquadFilterNode is an AudioNode processor implementing very common low-order filters.

Low-order filters are the building blocks of basic tone controls (bass, mid, treble), graphic equalizers, and more advanced filters. Multiple BiquadFilterNode filters can be combined to form more complex filters. The filter parameters such as frequency can be changed over time for filter sweeps, etc. Each BiquadFilterNode can be configured as one of a number of common filter types as shown in the IDL below. The default filter type is "lowpass".

Both frequency and detune form a compound parameter and are both a-rate. They are used together to determine a computedFrequency value:

computedFrequency(t) = frequency(t) * pow(2, detune(t) / 1200)

The nominal range for this compound parameter is [0, Nyquist frequency].

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	Yes	Continues to output non-silent audio with zero input. Since this is an IIR filter, the filter produces non-zero input forever, but in practice, this can be limited after some finite time where the output is sufficiently close to zero. The actual time depends on the filter coefficients.

The number of channels of the output always equals the number of channels of the input.

enum BiquadFilterType {
  "lowpass",
  "highpass",
  "bandpass",
  "lowshelf",
  "highshelf",
  "peaking",
  "notch",
  "allpass"
};

Enumeration description
"`lowpass`"	A lowpass filter allows frequencies below the cutoff frequency to pass through and attenuates frequencies above the cutoff. It implements a standard second-order resonant lowpass filter with 12dB/octave rolloff. frequency The cutoff frequency Q Controls how peaked the response will be at the cutoff frequency. A large value makes the response more peaked. Please note that for this filter type, this value is not a traditional Q, but is a resonance value in decibels. gain Not used in this filter type
"`highpass`"	A highpass filter is the opposite of a lowpass filter. Frequencies above the cutoff frequency are passed through, but frequencies below the cutoff are attenuated. It implements a standard second-order resonant highpass filter with 12dB/octave rolloff. frequency The cutoff frequency below which the frequencies are attenuated Q Controls how peaked the response will be at the cutoff frequency. A large value makes the response more peaked. Please note that for this filter type, this value is not a traditional Q, but is a resonance value in decibels. gain Not used in this filter type
"`bandpass`"	A bandpass filter allows a range of frequencies to pass through and attenuates the frequencies below and above this frequency range. It implements a second-order bandpass filter. frequency The center of the frequency band Q Controls the width of the band. The width becomes narrower as the Q value increases. gain Not used in this filter type
"`lowshelf`"	The lowshelf filter allows all frequencies through, but adds a boost (or attenuation) to the lower frequencies. It implements a second-order lowshelf filter. frequency The upper limit of the frequences where the boost (or attenuation) is applied. Q Not used in this filter type. gain The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.
"`highshelf`"	The highshelf filter is the opposite of the lowshelf filter and allows all frequencies through, but adds a boost to the higher frequencies. It implements a second-order highshelf filter frequency The lower limit of the frequences where the boost (or attenuation) is applied. Q Not used in this filter type. gain The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.
"`peaking`"	The peaking filter allows all frequencies through, but adds a boost (or attenuation) to a range of frequencies. frequency The center frequency of where the boost is applied. Q Controls the width of the band of frequencies that are boosted. A large value implies a narrow width. gain The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.
"`notch`"	The notch filter (also known as a band-stop or band-rejection filter) is the opposite of a bandpass filter. It allows all frequencies through, except for a set of frequencies. frequency The center frequency of where the notch is applied. Q Controls the width of the band of frequencies that are attenuated. A large value implies a narrow width. gain Not used in this filter type.
"`allpass`"	An allpass filter allows all frequencies through, but changes the phase relationship between the various frequencies. It implements a second-order allpass filter frequency The frequency where the center of the phase transition occurs. Viewed another way, this is the frequency with maximal group delay. Q Controls how sharp the phase transition is at the center frequency. A larger value implies a sharper transition and a larger group delay. gain Not used in this filter type.

All attributes of the BiquadFilterNode are a-rate AudioParams.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional BiquadFilterOptions options)]
interface BiquadFilterNode : AudioNode {
  attribute BiquadFilterType type;
  readonly attribute AudioParam frequency;
  readonly attribute AudioParam detune;
  readonly attribute AudioParam Q;
  readonly attribute AudioParam gain;
  void getFrequencyResponse (Float32Array frequencyHz, Float32Array magResponse, Float32Array phaseResponse);
};

1.13.1. Constructors

BiquadFilterNode(context, options)

Let node be a new BiquadFilterNode object. Initialize node, and return node.

Arguments for the BiquadFilterNode.BiquadFilterNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `BiquadFilterNode` will be associated with.
`options`	BiquadFilterOptions	✘	✔	Optional initial parameter value for this `BiquadFilterNode`.

1.13.2. Attributes

Q, of type AudioParam, readonly

The Q factor of the filter. This is not used for lowshelf or highshelf filters.

Parameter	Value	Notes
`defaultValue`	1
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

detune, of type AudioParam, readonly

A detune value, in cents, for the frequency. It forms a compound parameter with frequency.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

frequency, of type AudioParam, readonly

The frequency at which the BiquadFilterNode will operate, in Hz. It forms a compound parameter with detune.

Parameter	Value	Notes
`defaultValue`	350
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

gain, of type AudioParam, readonly

The gain of the filter. Its value is in dB units. The gain is only used for lowshelf, highshelf, and peaking filters.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

type, of type BiquadFilterType

The type of this BiquadFilterNode. Its default value is "lowpass". The exact meaning of the other parameters depend on the value of the type attribute.

1.13.3. Methods

getFrequencyResponse(frequencyHz, magResponse, phaseResponse)

Given the current filter parameter settings, synchronously calculates the frequency response for the specified frequencies. The three parameters MUST be Float32Arrays of the same length, or an InvalidAccessError MUST be thrown.

The frequency response returned MUST be computed with the AudioParam sampled for the current processing block.

Arguments for the BiquadFilterNode.getFrequencyResponse() method.
Parameter	Type	Nullable	Optional	Description
`frequencyHz`	Float32Array	✘	✘	This parameter specifies an array of frequencies at which the response values will be calculated.
`magResponse`	Float32Array	✘	✘	This parameter specifies an output array receiving the linear magnitude response values. If a value in the `frequencyHz` parameter is not within [0; sampleRate/2], where `sampleRate` is the value of the `sampleRate` property of the `AudioContext`, the corresponding value at the same index of the `magResponse` array MUST be `NaN`.
`phaseResponse`	Float32Array	✘	✘	This parameter specifies an output array receiving the phase response values in radians. If a value in the `frequencyHz` parameter is not within [0; sampleRate/2], where `sampleRate` is the value of the `sampleRate` property of the `AudioContext`, the corresponding value at the same index of the `phaseResponse` array MUST be `NaN`.

Return type: void

1.13.4. `BiquadFilterOptions`

This specifies the options to be used when constructing a BiquadFilterNode. All members are optional; if not specified, the normal default values are used to construct the node.

dictionary BiquadFilterOptions : AudioNodeOptions {
  BiquadFilterType type = "lowpass";
  float Q = 1;
  float detune = 0;
  float frequency = 350;
  float gain = 0;
};

1.13.4.1. Dictionary `BiquadFilterOptions` Members

Q, of type float, defaulting to 1: The desired initial value for Q.
detune, of type float, defaulting to 0: The desired initial value for detune.
frequency, of type float, defaulting to 350: The desired initial value for frequency.
gain, of type float, defaulting to 0: The desired initial value for gain.
type, of type BiquadFilterType, defaulting to "lowpass": The desired initial type of the filter.

1.13.5. Filters Characteristics

There are multiple ways of implementing the type of filters available through the BiquadFilterNode each having very different characteristics. The formulas in this section describe the filters that a conforming implementation MUST implement, as they determine the characteristics of the different filter types. They are inspired by formulas found in the Audio EQ Cookbook.

The transfer function for the filters implemented by the BiquadFilterNode is:

$$
H(z) = \frac{\frac{b_0}{a_0} + \frac{b_1}{a_0}z^{-1} + \frac{b_2}{a_0}z^{-2}}
            {1+\frac{a_1}{a_0}z^{-1}+\frac{a_2}{a_0}z^{-2}}
$$

The initial filter state is 0. The coefficients in the transfer function above are different for each node type. The following intermediate variable are necessary for their computation, based on the computedValue of the AudioParams of the BiquadFilterNode.

Let \(F_s\) be the value of the sampleRate attribute for this AudioContext.
Let \(f_0\) be the value of the computedFrequency.
Let \(G\) be the value of the gain AudioParam.
Let \(Q\) be the value of the Q AudioParam.

Finally let

$$
\begin{align*}
  A &= 10^{\frac{G}{40}} \\
  \omega_0 &= 2\pi\frac{f_0}{F_s} \\
  \alpha_Q &= \frac{\sin\omega_0}{2Q} \\
  \alpha_{Q_{dB}} &= \frac{\sin\omega_0}{2 \cdot 10^{Q/20}} \\
  S &= 1 \\
  \alpha_S &= \frac{\sin\omega_0}{2}\sqrt{\left(A+\frac{1}{A}\right)\left(\frac{1}{S}-1\right)+2}
\end{align*}
$$

The six coefficients (\(b_0, b_1, b_2, a_0, a_1, a_2\)) for each filter type, are:

"lowpass"

$$
  \begin{align*}
    b_0 &= \frac{1 - \cos\omega_0}{2} \\
    b_1 &= 1 - \cos\omega_0 \\
    b_2 &= \frac{1 - \cos\omega_0}{2} \\
    a_0 &= 1 + \alpha_{Q_{dB}} \\
    a_1 &= -2 \cos\omega_0 \\
    a_2 &= 1 - \alpha_{Q_{dB}}
  \end{align*}
$$

"highpass"

$$
  \begin{align*}
    b_0 &= \frac{1 + \cos\omega_0}{2} \\
    b_1 &= -(1 + \cos\omega_0) \\
    b_2 &= \frac{1 + \cos\omega_0}{2} \\
    a_0 &= 1 + \alpha_{Q_{dB}} \\
    a_1 &= -2 \cos\omega_0 \\
    a_2 &= 1 - \alpha_{Q_{dB}}
  \end{align*}
$$

"bandpass"

$$
  \begin{align*}
    b_0 &= \alpha_Q \\
    b_1 &= 0 \\
    b_2 &= -\alpha_Q \\
    a_0 &= 1 + \alpha_Q \\
    a_1 &= -2 \cos\omega_0 \\
    a_2 &= 1 - \alpha_Q
  \end{align*}
$$

"notch"

$$
  \begin{align*}
    b_0 &= 1 \\
    b_1 &= -2\cos\omega_0 \\
    b_2 &= 1 \\
    a_0 &= 1 + \alpha_Q \\
    a_1 &= -2 \cos\omega_0 \\
    a_2 &= 1 - \alpha_Q
  \end{align*}
$$

"allpass"

$$
  \begin{align*}
    b_0 &= 1 - \alpha_Q \\
    b_1 &= -2\cos\omega_0 \\
    b_2 &= 1 + \alpha_Q \\
    a_0 &= 1 + \alpha_Q \\
    a_1 &= -2 \cos\omega_0 \\
    a_2 &= 1 - \alpha_Q
  \end{align*}
$$

"peaking"

$$
  \begin{align*}
    b_0 &= 1 + \alpha_Q\, A \\
    b_1 &= -2\cos\omega_0 \\
    b_2 &= 1 - \alpha_Q\,A \\
    a_0 &= 1 + \frac{\alpha_Q}{A} \\
    a_1 &= -2 \cos\omega_0 \\
    a_2 &= 1 - \frac{\alpha_Q}{A}
  \end{align*}
$$

"lowshelf"

$$
  \begin{align*}
    b_0 &= A \left[ (A+1) - (A-1) \cos\omega_0 + 2 \alpha_S \sqrt{A})\right] \\
    b_1 &= 2 A \left[ (A-1) - (A+1) \cos\omega_0 )\right] \\
    b_2 &= A \left[ (A+1) - (A-1) \cos\omega_0 - 2 \alpha_S \sqrt{A}) \right] \\
    a_0 &= (A+1) + (A-1) \cos\omega_0 + 2 \alpha_S \sqrt{A} \\
    a_1 &= -2 \left[ (A-1) + (A+1) \cos\omega_0\right] \\
    a_2 &= (A+1) + (A-1) \cos\omega_0 - 2 \alpha_S \sqrt{A})
  \end{align*}
$$

"highshelf"

$$
  \begin{align*}
    b_0 &= A\left[ (A+1) + (A-1)\cos\omega_0 + 2\alpha_S\sqrt{A} )\right] \\
    b_1 &= -2A\left[ (A-1) + (A+1)\cos\omega_0 )\right] \\
    b_2 &= A\left[ (A+1) + (A-1)\cos\omega_0 - 2\alpha_S\sqrt{A} )\right] \\
    a_0 &= (A+1) - (A-1)\cos\omega_0 + 2\alpha_S\sqrt{A} \\
    a_1 &= 2\left[ (A-1) - (A+1)\cos\omega_0\right] \\
    a_2 &= (A+1) - (A-1)\cos\omega_0 - 2\alpha_S\sqrt{A}
  \end{align*}
$$

1.14. The `ChannelMergerNode` Interface

The ChannelMergerNode is for use in more advanced applications and would often be used in conjunction with ChannelSplitterNode.

Property	Value	Notes
`numberOfInputs`	see notes	Defaults to 6, but is determined by `ChannelMergerOptions`,`numberOfInputs` or the value specified by `createChannelMerger`.
`numberOfOutputs`	1
`channelCount`	1	Has channelCount constraints
`channelCountMode`	"`explicit`"	Has channelCountMode constraints
`channelInterpretation`	"`speakers`"
tail-time reference	No

This interface represents an AudioNode for combining channels from multiple audio streams into a single audio stream. It has a variable number of inputs (defaulting to 6), but not all of them need be connected. There is a single output whose audio stream has a number of channels equal to the number of inputs.

To merge multiple inputs into one stream, each input gets downmixed into one channel (mono) based on the specified mixing rule. An unconnected input still counts as one silent channel in the output. Changing input streams does not affect the order of output channels.

For example, if a default ChannelMergerNode has two connected stereo inputs, the first and second input will be downmixed to mono respectively before merging. The output will be a 6-channel stream whose first two channels are be filled with the first two (downmixed) inputs and the rest of channels will be silent.

Also the ChannelMergerNode can be used to arrange multiple audio streams in a certain order for the multi-channel speaker array such as 5.1 surround set up. The merger does not interpret the channel identities (such as left, right, etc.), but simply combines channels in the order that they are input.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional ChannelMergerOptions options)]
interface ChannelMergerNode : AudioNode {
};

1.14.1. Constructors

ChannelMergerNode(context, options)

Let node be a new ChannelMergerNode object. Initialize node, and return node.

Arguments for the ChannelMergerNode.ChannelMergerNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `ChannelMergerNode` will be associated with.
`options`	ChannelMergerOptions	✘	✔	Optional initial parameter value for this `ChannelMergerNode`.

1.14.2. `ChannelMergerOptions`

dictionary ChannelMergerOptions : AudioNodeOptions {
  unsigned long numberOfInputs = 6;
};

1.14.2.1. Dictionary `ChannelMergerOptions` Members

numberOfInputs, of type unsigned long, defaulting to 6: The number inputs for the ChannelMergerNode. See createChannelMerger() for constraints on this value.

1.15. The `ChannelSplitterNode` Interface

The ChannelSplitterNode is for use in more advanced applications and would often be used in conjunction with ChannelMergerNode.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	see notes	This defaults to 6, but is otherwise determined from `ChannelSplitterOptions.numberOfOutputs` or the value specified by `createChannelSplitter` or the `numberOfOutputs` member of the `ChannelSplitterOptions` dictionary for the `constructor`.
`channelCount`	`numberOfOutputs`	Has channelCount constraints
`channelCountMode`	"`explicit`"	Has channelCountMode constraints
`channelInterpretation`	"`discrete`"	Has channelInterpretation constraints
tail-time reference	No

This interface represents an AudioNode for accessing the individual channels of an audio stream in the routing graph. It has a single input, and a number of "active" outputs which equals the number of channels in the input audio stream. For example, if a stereo input is connected to an ChannelSplitterNode then the number of active outputs will be two (one from the left channel and one from the right). There are always a total number of N outputs (determined by the numberOfOutputs parameter to the AudioContext method createChannelSplitter()), The default number is 6 if this value is not provided. Any outputs which are not "active" will output silence and would typically not be connected to anything.

Please note that in this example, the splitter does not interpret the channel identities (such as left, right, etc.), but simply splits out channels in the order that they are input.

One application for ChannelSplitterNode is for doing "matrix mixing" where individual gain control of each channel is desired.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional ChannelSplitterOptions options)]
interface ChannelSplitterNode : AudioNode {
};

1.15.1. Constructors

ChannelSplitterNode(context, options)

Let node be a new ChannelSplitterNode object. Initialize node, and return node.

Arguments for the ChannelSplitterNode.ChannelSplitterNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `ChannelSplitterNode` will be associated with.
`options`	ChannelSplitterOptions	✘	✔	Optional initial parameter value for this `ChannelSplitterNode`.

1.15.2. `ChannelSplitterOptions`

dictionary ChannelSplitterOptions : AudioNodeOptions {
  unsigned long numberOfOutputs = 6;
};

1.15.2.1. Dictionary `ChannelSplitterOptions` Members

numberOfOutputs, of type unsigned long, defaulting to 6: The number outputs for the ChannelSplitterNode. See createChannelSplitter() for constraints on this value.

1.16. The `ConstantSourceNode` Interface

This interface represents a constant audio source whose output is nominally a constant value. It is useful as a constant source node in general and can be used as if it were a constructible AudioParam by automating its offset or connecting another node to it.

The single output of this node consists of one channel (mono).

Property	Value	Notes
`numberOfInputs`	0
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	No

[Exposed=Window,
 Constructor (BaseAudioContext context, optional ConstantSourceOptions options)]
interface ConstantSourceNode : AudioScheduledSourceNode {
  readonly attribute AudioParam offset;
};

1.16.1. Constructors

ConstantSourceNode(context, options)

Let node be a new ConstantSourceNode object. Initialize node, and return node.

Arguments for the ConstantSourceNode.ConstantSourceNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `ConstantSourceNode` will be associated with.
`options`	ConstantSourceOptions	✘	✔	Optional initial parameter value for this `ConstantSourceNode`.

1.16.2. Attributes

offset, of type AudioParam, readonly

The constant value of the source.

Parameter	Value	Notes
`defaultValue`	1
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

1.16.3. `ConstantSourceOptions`

This specifies options for constructing a ConstantSourceNode. All members are optional; if not specified, the normal defaults are used for constructing the node.

dictionary ConstantSourceOptions {
  float offset = 1;
};

1.16.3.1. Dictionary `ConstantSourceOptions` Members

offset, of type float, defaulting to 1: The initial value for the offset AudioParam of this node.

1.17. The `ConvolverNode` Interface

This interface represents a processing node which applies a linear convolution effect given an impulse response.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2	Has channelCount constraints
`channelCountMode`	"`clamped-max`"	Has channelCountMode constraints
`channelInterpretation`	"`speakers`"
tail-time reference	Yes	Continues to output non-silent audio with zero input for the length of the `buffer`.

The input of this node is either mono (1 channel) or stereo (2 channels) and cannot be increased. Connections from nodes with more channels will be down-mixed appropriately.

There are channelCount constraints and channelCountMode constraints for this node. These constraints ensure that the input to the node is either mono or stereo.

ConvolverNodes are created with an internal flag buffer set, initially set to false.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional ConvolverOptions options)]
interface ConvolverNode : AudioNode {
  attribute AudioBuffer? buffer;
  attribute boolean normalize;
};

1.17.1. Constructors

ConvolverNode(context, options)

Let node be a new ConvolverNode object. Initialize node. Set an internal boolean slot [[buffer set]], and initialize it to false. Return node.

Arguments for the ConvolverNode.ConvolverNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `ConvolverNode` will be associated with.
`options`	ConvolverOptions	✘	✔	Optional initial parameter value for this `ConvolverNode`.

1.17.2. Attributes

buffer, of type AudioBuffer, nullable

A mono, stereo, or 4-channel AudioBuffer containing the (possibly multi-channel) impulse response used by the ConvolverNode. The AudioBuffer MUST have 1, 2, or 4 channels or a NotSupportedError exception MUST be thrown. This AudioBuffer MUST be of the same sample-rate as the AudioContext or a NotSupportedError exception MUST be thrown. At the time when this attribute is set, the buffer and the state of the normalize attribute will be used to configure the ConvolverNode with this impulse response having the given normalization. The initial value of this attribute is null.

To set the buffer attribute, execute these steps:

Let new buffer be an AudioBuffer to be assigned to buffer or null .
If new buffer is not null and [[buffer set]] is true, throw an InvalidStateError and abort these steps.
If new buffer is not null, set [[buffer set]] to true.
Assign new buffer to the buffer attribute.

Note: The ConvolverNode produces a mono output only in the single case where there is a single input channel and a single-channel buffer. In all other cases, the output is stereo. In particular, when the buffer has four channels and there are two input channels, the ConvolverNode performs matrix "true" stereo convolution. For normative information please see the channel configuration diagrams

normalize, of type boolean

Controls whether the impulse response from the buffer will be scaled by an equal-power normalization when the buffer atttribute is set. Its default value is true in order to achieve a more uniform output level from the convolver when loaded with diverse impulse responses. If normalize is set to false, then the convolution will be rendered with no pre-processing/scaling of the impulse response. Changes to this value do not take effect until the next time the buffer attribute is set.

If the normalize attribute is false when the buffer attribute is set then the ConvolverNode will perform a linear convolution given the exact impulse response contained within the buffer.

Otherwise, if the normalize attribute is true when the buffer attribute is set then the ConvolverNode will first perform a scaled RMS-power analysis of the audio data contained within buffer to calculate a normalizationScale given this algorithm:

function calculateNormalizationScale(buffer) {  var GainCalibration = 0.00125;  var GainCalibrationSampleRate = 44100;  var MinPower = 0.000125;  // Normalize by RMS power.  var numberOfChannels = buffer.numberOfChannels;  var length = buffer.length;  var power = 0;  for (var i = 0; i < numberOfChannels; i++) {    var channelPower = 0;    var channelData = buffer.getChannelData(i);    for (var j = 0; j < length; j++) {      var sample = channelData[j];      channelPower += sample * sample;    }    power += channelPower;  }  power = Math.sqrt(power / (numberOfChannels * length));  // Protect against accidental overload.  if (!isFinite(power) || isNaN(power) || power < MinPower)    power = MinPower;  var scale = 1 / power;  // Calibrate to make perceived volume same as unprocessed.  scale *= GainCalibration;  // Scale depends on sample-rate.  if (buffer.sampleRate)    scale *= GainCalibrationSampleRate / buffer.sampleRate;  // True-stereo compensation.  if (numberOfChannels == 4)    scale *= 0.5;  return scale;}

During processing, the ConvolverNode will then take this calculated normalizationScale value and multiply it by the result of the linear convolution resulting from processing the input with the impulse response (represented by the buffer) to produce the final output. Or any mathematically equivalent operation may be used, such as pre-multiplying the input by normalizationScale, or pre-multiplying a version of the impulse-response by normalizationScale.

1.17.3. `ConvolverOptions`

The specifies options for constructing a ConvolverNode. All members are optional; if not specified, the node is contructing using the normal defaults.

dictionary ConvolverOptions : AudioNodeOptions {
  AudioBuffer? buffer;
  boolean disableNormalization = false;
};

1.17.3.1. Dictionary `ConvolverOptions` Members

buffer, of type AudioBuffer, nullable: The desired buffer for the ConvolverNode. This buffer will be normalized according to the value of disableNormalization.
disableNormalization, of type boolean, defaulting to false: The opposite of the desired initial value for the normalize attribute of the ConvolverNode.

1.17.4. Channel Configurations for Input, Impulse Response and Output

Implementations MUST support the following allowable configurations of impulse response channels in a ConvolverNode to achieve various reverb effects with 1 or 2 channels of input.

As shown in the diagram below, single channel convolution operates on a mono audio input, using a mono impulse response, and generating a mono output. The remaining images in the diagram illustrate the supported cases for mono and stereo playback where the number of channels of the input is 1 or 2, and the number of channels in the buffer is 1, 2, or 4. Developers desiring more complex and arbitrary matrixing can use a ChannelSplitterNode, multiple single-channel ConvolverNodes and a ChannelMergerNode.

reverb matrixing — A graphical representation of supported input and output channel count possibilities when using a `ConvolverNode`.

1.18. The `DelayNode` Interface

A delay-line is a fundamental building block in audio applications. This interface is an AudioNode with a single input and single output.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	Yes	Continues to output non-silent audio with zero input up to the `maxDelayTime` of the node.

The number of channels of the output always equals the number of channels of the input.

It delays the incoming audio signal by a certain amount. Specifically, at each time t, input signal input(t), delay time delayTime(t) and output signal output(t), the output will be output(t) = input(t - delayTime(t)). The default delayTime is 0 seconds (no delay).

When the number of channels in a DelayNode's input changes (thus changing the output channel count also), there may be delayed audio samples which have not yet been output by the node and are part of its internal state. If these samples were received earlier with a different channel count, they MUST be upmixed or downmixed before being combined with newly received input so that all internal delay-line mixing takes place using the single prevailing channel layout.

Note: By definition, a DelayNode introduces an audio processing latency equal to the amount of the delay.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional DelayOptions options)]
interface DelayNode : AudioNode {
  readonly attribute AudioParam delayTime;
};

1.18.1. Constructors

DelayNode(context, options)

Let node be a new DelayNode object. Initialize node, and return node.

Arguments for the DelayNode.DelayNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `DelayNode` will be associated with.
`options`	DelayOptions	✘	✔	Optional initial parameter value for this `DelayNode`.

1.18.2. Attributes

delayTime, of type AudioParam, readonly

An AudioParam object representing the amount of delay (in seconds) to apply. Its default value is 0 (no delay). The minimum value is 0 and the maximum value is determined by the maxDelayTime argument to the AudioContext method createDelay() or the maxDelayTime member of the DelayOptions dictionary for the constructor.

If DelayNode is part of a cycle, then the value of the delayTime attribute is clamped to a minimum of one render quantum.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	0
`maxValue`	`maxDelayTime`
`automationRate`	"`a-rate`"

1.18.3. `DelayOptions`

This specifies options for constructing a DelayNode. All members are optional; if not given, the node is constructed using the normal defaults.

dictionary DelayOptions : AudioNodeOptions {
  double maxDelayTime = 1;
  double delayTime = 0;
};

1.18.3.1. Dictionary `DelayOptions` Members

delayTime, of type double, defaulting to 0: The initial delay time for the node.
maxDelayTime, of type double, defaulting to 1: The maximum delay time for the node. See createDelay(maxDelayTime) for constraints.

1.19. The `DynamicsCompressorNode` Interface

DynamicsCompressorNode is an AudioNode processor implementing a dynamics compression effect.

Dynamics compression is very commonly used in musical production and game audio. It lowers the volume of the loudest parts of the signal and raises the volume of the softest parts. Overall, a louder, richer, and fuller sound can be achieved. It is especially important in games and musical applications where large numbers of individual sounds are played simultaneous to control the overall signal level and help avoid clipping (distorting) the audio output to the speakers.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2	Has channelCount constraints
`channelCountMode`	"`clamped-max`"	Has channelCountMode constraints
`channelInterpretation`	"`speakers`"
tail-time reference	Yes	This node has a tail-time reference such that this node continues to output non-silent audio with zero input due to the look-ahead delay.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional DynamicsCompressorOptions options)]
interface DynamicsCompressorNode : AudioNode {
  readonly attribute AudioParam threshold;
  readonly attribute AudioParam knee;
  readonly attribute AudioParam ratio;
  readonly attribute float reduction;
  readonly attribute AudioParam attack;
  readonly attribute AudioParam release;
};

1.19.1. Constructors

DynamicsCompressorNode(context, options)

Let node be a new DynamicsCompressorNode object. Initialize node. Let [[internal reduction]] be a private slot on this DynamicsCompressorNode, that holds a floating point number, in decibels. Set [[internal reduction]] to 0.0. Return node.

Arguments for the DynamicsCompressorNode.DynamicsCompressorNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `DynamicsCompressorNode` will be associated with.
`options`	DynamicsCompressorOptions	✘	✔	Optional initial parameter value for this `DynamicsCompressorNode`.

1.19.2. Attributes

attack, of type AudioParam, readonly

The amount of time (in seconds) to reduce the gain by 10dB.

Parameter	Value	Notes
`defaultValue`	.003
`minValue`	0
`maxValue`	1
`automationRate`	"`k-rate`"	Has automation rate constraints

knee, of type AudioParam, readonly

A decibel value representing the range above the threshold where the curve smoothly transitions to the "ratio" portion.

Parameter	Value	Notes
`defaultValue`	30
`minValue`	0
`maxValue`	40
`automationRate`	"`k-rate`"	Has automation rate constraints

ratio, of type AudioParam, readonly

The amount of dB change in input for a 1 dB change in output.

Parameter	Value	Notes
`defaultValue`	12
`minValue`	1
`maxValue`	20
`automationRate`	"`k-rate`"	Has automation rate constraints

reduction, of type float, readonly

A read-only decibel value for metering purposes, representing the current amount of gain reduction that the compressor is applying to the signal. If fed no signal the value will be 0 (no gain reduction). When this attribute is read, return the value of the private slot [[internal reduction]].

release, of type AudioParam, readonly

The amount of time (in seconds) to increase the gain by 10dB.

Parameter	Value	Notes
`defaultValue`	.25
`minValue`	0
`maxValue`	1
`automationRate`	"`k-rate`"	Has automation rate constraints

threshold, of type AudioParam, readonly

The decibel value above which the compression will start taking effect.

Parameter	Value	Notes
`defaultValue`	-24
`minValue`	-100
`maxValue`	0
`automationRate`	"`k-rate`"	Has automation rate constraints

1.19.3. `DynamicsCompressorOptions`

This specifies the options to use in constructing a DynamicsCompressorNode. All members are optional; if not specified the normal defaults are used in constructing the node.

dictionary DynamicsCompressorOptions : AudioNodeOptions {
  float attack = 0.003;
  float knee = 30;
  float ratio = 12;
  float release = 0.25;
  float threshold = -24;
};

1.19.3.1. Dictionary `DynamicsCompressorOptions` Members

attack, of type float, defaulting to 0.003: The initial value for the attack AudioParam.
knee, of type float, defaulting to 30: The initial value for the knee AudioParam.
ratio, of type float, defaulting to 12: The initial value for the ratio AudioParam.
release, of type float, defaulting to 0.25: The initial value for the release AudioParam.
threshold, of type float, defaulting to -24: The initial value for the threshold AudioParam.

1.19.4. Processing

Dynamics compression can be implemented in a variety of ways. The DynamicsCompressorNode implements a dynamics processor that has the following characteristics:

Fixed look-ahead (this means that an DynamicsCompressorNode adds a fixed latency to the signal chain).
Configurable attack speed, release speed, threshold, knee hardness and ratio.
Side-chaining is not supported.
The gain reduction is reported via the reduction property on the DynamicsCompressorNode.
The compression curve has three parts:
- The first part is the identity: \(f(x) = x\).
- The second part is the soft-knee portion, which MUST be a monotonically increasing function.
- The third part is a linear function: \(f(x) = \frac{1}{ratio} \cdot x \).
This curve MUST be continuous and piece-wise differentiable, and corresponds to a target output level, based on the input level.

Graphically, such a curve would look something like this:

Graphical representation of a compression curve — A typical compression curve, showing the knee portion (soft or hard) as well as the threshold.

Internally, the DynamicsCompressorNode is described with a combination of other AudioNodes, as well as a special algorithm, to compute the gain reduction value.

The following AudioNode graph is used internally, input and output respectively being the input and output AudioNode, context the BaseAudioContext for this DynamicsCompressorNode, and new class, EnvelopeFollower, that instantiate a special object that behaves like an AudioNode, described below:

var delay = new DelayNode(context, {delayTime: 0.006});
var gain = new GainNode(context);
var compression = new EnvelopeFollower();

input.connect(delay).connect(gain).connect(output);
input.connect(compression).connect(gain.gain);

Schema of
the internal graph used by the DynamicCompressorNode — The graph of internal `AudioNode`s used as part of the `DynamicsCompressorNode` processing algorithm.

Note: This implements the pre-delay and the application of the reduction gain.

The following algorithm describes the processing performed by an EnvelopeFollower object, to be applied to the input signal to produce the gain reduction value. An EnvelopeFollower has two slots holding floating point values. Those values persist accros invocation of this algorithm.

Let [[detector average]] be a floating point number, initialized to 0.0.
Let [[compressor gain]] be a floating point number, initialized to 1.0.

The following algorithm allow determining a value for reduction gain, for each sample of input, for a render quantum of audio.

Let attack and release have the values of attack and release, respectively, sampled at the time of processing (those are k-rate parameters), mutiplied by the sample-rate of the BaseAudioContext this DynamicsCompressorNode is associated with.
Let detector average be the value of the slot [[detector average]].
Let compressor gain be the value of the slot [[compressor gain]].
For each sample input of the render quantum to be processed, execute the following steps:
1. If the absolute value of input is less than 0.0001, let attenuation be 1.0. Else, let shaped input be the value of applying the compression curve to the absolute value of input. Let attenuation be shaped input divided by the absolute value of input.
2. Let releasing be true if attenuation is greater than compressor gain, false otherwise.
3. Let detector rate be the result of applying the detector curve to attenuation.
4. Substract detector average to attenuation, and multiply the result by detector rate. Add this new result to detector average.
5. Clamp detector average to a maximum of 1.0.
6. Let envelope rate be the result of computing the envelope rate based on values of attack and release.
7. If releasing is true, set compressor gain to the multiplication of compressor gain by envelope rate, clamped to a maximum of 1.0.
8. Else, if releasing is false, let gain increment to be detector average minus compressor gain. Multiply gain increment by envelope rate, and add the result to compressor gain.
9. Compute reduction gain to be compressor gain multiplied by the return value of computing the makeup gain.
10. Compute metering gain to be reduction gain, converted to decibel.
Set [[compressor gain]] to compressor gain.
Set [[detector average]] to detector average.
Atomically set the internal slot [[internal reduction]] to the value of metering gain.

Note: This step makes the metering gain update once per block, at the end of the block processing.

The makeup gain is a fixed gain stage that only depends on ratio, knee and threshold parameter of the compressor, and not on the input signal. The intent here is to increase the output level of the compressor so it is comparable to the input level.

Computing the makeup gain means executing the following steps:

Let full range gain be the value returned by applying the compression curve to the value 1.0.
Let full range makeup gain the inverse of full range gain.
Return the result of taking 0.6 power of full range makeup gain.

Computing the envelope rate is done by applying a function to the ratio of the compressor gain and the detector average. User-agents are allowed to choose the shape of the envelope function. However, this function MUST respect the following constraints:

The envelope rate MUST be the calculated from the ratio of the compressor gain and the detector average.

Note: When attacking, this number less than or equal to 1, when releasing, this number is strictly greater than 1.
The attack curve MUST be a continuous, monotonically increasing function in the range \([0, 1]\). The shape of this curve MAY be controlled by attack.
The release curve MUST be a continuous, monotonically decreasing function that is always greater than 1. The shape of this curve MAY be controlled by release.

This operation returns the value computed by applying this function to the ratio of compressor gain and detector average.

Applying the detector curve to the change rate when attacking or releasing allow implementing adaptive release. It is a function that MUST respect the following constraints:

The output of the function MUST be in \([0,1]\).
The function MUST be monotonically increasing, continuous.

Note: It is allowed, for example, to have a compressor that performs an adaptive release, that is, releasing faster the harder the compression, or to have curves for attack and release that are not of the same shape.

Applying a compression curve to a value means computing the value of this sample when passed to a function, and returning the computed value. This function MUST respect the following characteristics:

Let threshold and knee have the values of threshold and knee, respectively, converted to linear units sampled at the time of processing of this block (as k-rate parameters).
Let ratio have the value of the ratio, sampled at the time of processing of this block (as a k-rate parameter).
This function is the identity up to the value of the linear threshold (i.e., \(f(x) = x\)).
From the threshold up to the threshold + knee, User-Agents can choose the curve shape. The whole function MUST be monotonically increasing and continuous.

Note: If the knee is 0, the DynamicsCompressorNode is called a hard-knee compressor.
This function is linear, based on the ratio, after the threshold and the soft knee (i.e., \(f(x) = \frac{1}{ratio} \cdot x \)).

Converting a value \(v\) in linear gain unit to decibel means executing the following steps:

If \(v\) is equal to zero, return -1000.
Else, return \( 20 \, \log_{10}{v} \)

Converting a value \(v\) in decibels to linear gain unit means returning \(10^\frac{v}{20}\)

1.20. The `GainNode` Interface

Changing the gain of an audio signal is a fundamental operation in audio applications. The GainNode is one of the building blocks for creating mixers. This interface is an AudioNode with a single input and single output:

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	No

Each sample of each channel of the input data of the GainNode MUST be multiplied by the computedValue of the gain AudioParam.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional GainOptions options)]
interface GainNode : AudioNode {
  readonly attribute AudioParam gain;
};

1.20.1. Constructors

GainNode(context, options)

Let gain be a new GainNode object. Initialize gain, and return gain.

Arguments for the GainNode.GainNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `GainNode` will be associated with.
`options`	GainOptions	✘	✔	Optional initial parameter values for this `GainNode`.

1.20.2. Attributes

gain, of type AudioParam, readonly

Represents the amount of gain to apply.

Parameter	Value	Notes
`defaultValue`	1
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"

1.20.3. `GainOptions`

This specifies options to use in constructing a GainNode. All members are optional; if not specified, the normal defaults are used in constructing the node.

dictionary GainOptions : AudioNodeOptions {
  float gain = 1.0;
};

1.20.3.1. Dictionary `GainOptions` Members

gain, of type float, defaulting to 1.0: The initial gain value for the gain AudioParam.

1.21. The `IIRFilterNode` Interface

IIRFilterNode is an AudioNode processor implementing a general IIR Filter. In general, it is best to use BiquadFilterNode's to implement higher-order filters for the following reasons:

Generally less sensitive to numeric issues
Filter parameters can be automated
Can be used to create all even-ordered IIR filters

However, odd-ordered filters cannot be created, so if such filters are needed or automation is not needed, then IIR filters may be appropriate.

Once created, the coefficients of the IIR filter cannot be changed.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	Yes	Continues to output non-silent audio with zero input. Since this is an IIR filter, the filter produces non-zero input forever, but in practice, this can be limited after some finite time where the output is sufficiently close to zero. The actual time depends on the filter coefficients.

The number of channels of the output always equals the number of channels of the input.

[Exposed=Window,
 Constructor (BaseAudioContext context, IIRFilterOptions options)]
interface IIRFilterNode : AudioNode {
  void getFrequencyResponse (Float32Array frequencyHz, Float32Array magResponse, Float32Array phaseResponse);
};

1.21.1. Constructors

IIRFilterNode(context, options)

Arguments for the IIRFilterNode.IIRFilterNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `IIRFilterNode` will be associated with.
`options`	IIRFilterOptions	✘	✘	Optional initial parameter value for this `IIRFilterNode`.

1.21.2. Methods

getFrequencyResponse(frequencyHz, magResponse, phaseResponse)

Arguments for the IIRFilterNode.getFrequencyResponse() method.
Parameter	Type	Nullable	Optional	Description
`frequencyHz`	Float32Array	✘	✘	This parameter specifies an array of frequencies at which the response values will be calculated.
`magResponse`	Float32Array	✘	✘	This parameter specifies an output array receiving the linear magnitude response values. If a value in the `frequencyHz` parameter is not within [0; sampleRate/2], where `sampleRate` is the value of the `sampleRate` property of the `AudioContext`, the corresponding value at the same index of the `magResponse` array MUST be `NaN`.
`phaseResponse`	Float32Array	✘	✘	This parameter specifies an output array receiving the phase response values in radians. If a value in the `frequencyHz` parameter is not within [0; sampleRate/2], where `sampleRate` is the value of the `sampleRate` property of the `AudioContext`, the corresponding value at the same index of the `phaseResponse` array MUST be `NaN`.

Return type: void

1.21.3. IIRFilterOptions

The IIRFilterOptions dictionary is used to specify the filter coefficients of the IIRFilterNode.

dictionary IIRFilterOptions : AudioNodeOptions {
  required sequence<double> feedforward;
  required sequence<double> feedback;
};

1.21.3.1. Dictionary `IIRFilterOptions` Members

feedforward, of type sequence<double>: The feedforward coefficients for the IIRFilterNode. This member is required. If not specifed, a NotFoundError MUST be thrown. See feedforward argument of createIIRFilter() for other constraints.
feedback, of type sequence<double>: The feedback coefficients for the IIRFilterNode. This member is required. If not specifed, a NotFoundError MUST be thrown. See feedback argument of createIIRFilter() for other constraints.

1.21.4. Filter Definition

Let \(b_m\) be the feedforward coefficients and \(a_n\) be the feedback coefficients specified by createIIRFilter() or the IIRFilterOptions dictionary for the constructor. Then the transfer function of the general IIR filter is given by

$$
  H(z) = \frac{\sum_{m=0}^{M} b_m z^{-m}}{\sum_{n=0}^{N} a_n z^{-n}}
$$

where \(M + 1\) is the length of the \(b\) array and \(N + 1\) is the length of the \(a\) array. The coefficient \(a_0\) MUST not be 0 (see feedback parameter for createIIRFilter()). At least one of \(b_m\) MUST be non-zero (see feedforward parameter for createIIRFilter()).

Equivalently, the time-domain equation is:

$$
  \sum_{k=0}^{N} a_k y(n-k) = \sum_{k=0}^{M} b_k x(n-k)
$$

The initial filter state is the all-zeroes state.

1.22. The `MediaElementAudioSourceNode` Interface

This interface represents an audio source from an audio or video element.

Property	Value	Notes
`numberOfInputs`	0
`numberOfOutputs`	1
tail-time reference	No

The number of channels of the output corresponds to the number of channels of the media referenced by the HTMLMediaElement. Thus, changes to the media element’s src attribute can change the number of channels output by this node.

[Exposed=Window,
 Constructor (AudioContext context, MediaElementAudioSourceOptions options)]
interface MediaElementAudioSourceNode : AudioNode {
  [SameObject] readonly attribute HTMLMediaElement mediaElement;
};

1.22.1. Constructors

MediaElementAudioSourceNode(context, options)

Let node be a new MediaElementAudioSourceNode object. Initialize node, and return node.

Arguments for the MediaElementAudioSourceNode.MediaElementAudioSourceNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	AudioContext	✘	✘	The `BaseAudioContext` this new `MediaElementAudioSourceNode` will be associated with.
`options`	MediaElementAudioSourceOptions	✘	✘	Parameter value for this `MediaElementAudioSourceNode`.

1.22.2. Attributes

mediaElement, of type HTMLMediaElement, readonly: The HTMLMediaElement used when constructing this MediaElementAudioSourceNode.

A MediaElementAudioSourceNode is created given an HTMLMediaElement using the AudioContext createMediaElementSource() method or the mediaElement member of the MediaElementAudioSourceOptions dictionary for the constructor.

The number of channels of the single output equals the number of channels of the audio referenced by the HTMLMediaElement passed in as the argument to createMediaElementSource(), or is 1 if the HTMLMediaElement has no audio.

The HTMLMediaElement MUST behave in an identical fashion after the MediaElementAudioSourceNode has been created, except that the rendered audio will no longer be heard directly, but instead will be heard as a consequence of the MediaElementAudioSourceNode being connected through the routing graph. Thus pausing, seeking, volume, src attribute changes, and other aspects of the HTMLMediaElement MUST behave as they normally would if not used with a MediaElementAudioSourceNode.

var mediaElement = document.getElementById('mediaElementID');
var sourceNode = context.createMediaElementSource(mediaElement);
sourceNode.connect(filterNode);

1.22.3. `MediaElementAudioSourceOptions`

This specifies the options to use in constructing a MediaElementAudioSourceNode.

dictionary MediaElementAudioSourceOptions {
  required HTMLMediaElement mediaElement;
};

1.22.3.1. Dictionary `MediaElementAudioSourceOptions` Members

mediaElement, of type HTMLMediaElement: The media element that will be re-routed. This MUST be specified.

1.22.4. Security with MediaElementAudioSourceNode and Cross-Origin Resources

HTMLMediaElement allows the playback of cross-origin resources. Because Web Audio allows inspection of the content of the resource (e.g. using a MediaElementAudioSourceNode, and an AudioWorkletNode or ScriptProcessorNode to read the samples), information leakage can occur if scripts from one origin inspect the content of a resource from another origin.

To prevent this, a MediaElementAudioSourceNode MUST output silence instead of the normal output of the HTMLMediaElement if it has been created using an HTMLMediaElement for which the execution of the fetch algorithm [FETCH] labeled the resource as CORS-cross-origin.

1.23. The `MediaStreamAudioDestinationNode` Interface

This interface is an audio destination representing a MediaStream with a single MediaStreamTrack whose kind is "audio". This MediaStream is created when the node is created and is accessible via the stream attribute. This stream can be used in a similar way as a MediaStream obtained via getUserMedia(), and can, for example, be sent to a remote peer using the RTCPeerConnection (described in [webrtc]) addStream() method.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	0
`channelCount`	2
`channelCountMode`	"`explicit`"
`channelInterpretation`	"`speakers`"
tail-time reference	No

The number of channels of the input is by default 2 (stereo).

[Exposed=Window,
 Constructor (AudioContext context, optional AudioNodeOptions options)]
interface MediaStreamAudioDestinationNode : AudioNode {
  readonly attribute MediaStream stream;
};

1.23.1. Constructors

MediaStreamAudioDestinationNode(context, options)

Let node be a new MediaStreamAudioDestinationNode object. Initialize node, and return node.

Arguments for the MediaStreamAudioDestinationNode.MediaStreamAudioDestinationNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	AudioContext	✘	✘	The `BaseAudioContext` this new `MediaStreamAudioDestinationNode` will be associated with.
`options`	AudioNodeOptions	✘	✔	Optional initial parameter value for this `MediaStreamAudioDestinationNode`.

1.23.2. Attributes

stream, of type MediaStream, readonly: A MediaStream containing a single MediaStreamTrack with the same number of channels as the node itself, and whose kind attribute has the value "audio".

1.24. The `MediaStreamAudioSourceNode` Interface

This interface represents an audio source from a MediaStream. The track that will be used as the source of audio and will be output from this node is the first MediaStreamTrack whose kind attribute has the value "audio", when alphabetically sorting the tracks of this MediaStream by their id attribute. Those interfaces are described in [mediacapture-streams].

Note: The behaviour for picking the track to output is weird for legacy reasons. MediaStreamTrackAudioSourceNode should be used instead.

Property	Value	Notes
`numberOfInputs`	0
`numberOfOutputs`	1
tail-time reference	No

The number of channels of the output corresponds to the number of channels of the MediaStreamTrack. If there is no valid audio track, then the number of channels output will be one silent channel.

[Exposed=Window,
 Constructor (AudioContext context, MediaStreamAudioSourceOptions options)]
interface MediaStreamAudioSourceNode : AudioNode {
  [SameObject] readonly attribute MediaStream mediaStream;
};

1.24.1. Constructors

MediaStreamAudioSourceNode(context, options)

Let node be a new MediaStreamAudioSourceNode object. Initialize node, and return node.

Arguments for the MediaStreamAudioSourceNode.MediaStreamAudioSourceNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	AudioContext	✘	✘	The `BaseAudioContext` this new `MediaStreamAudioSourceNode` will be associated with.
`options`	MediaStreamAudioSourceOptions	✘	✘	Initial parameter values for this `MediaStreamAudioSourceNode`. If the `mediaStream` parameter does not reference a MediaStream whose `kind` attribute has the value `"audio"`, an `InvalidStateError` MUST be thrown.

1.24.2. Attributes

mediaStream, of type MediaStream, readonly: The MediaStream used when constructing this MediaStreamAudioSourceNode.

1.24.3. `MediaStreamAudioSourceOptions`

This specifies the options for constructing a MediaStreamAudioSourceNode.

dictionary MediaStreamAudioSourceOptions {
  required MediaStream mediaStream;
};

1.24.3.1. Dictionary `MediaStreamAudioSourceOptions` Members

mediaStream, of type MediaStream: The media stream that will act as a source. This MUST be specified.

1.25. The `MediaStreamTrackAudioSourceNode` Interface

This interface represents an audio source from a MediaStreamTrack.

Property	Value	Notes
`numberOfInputs`	0
`numberOfOutputs`	1
tail-time reference	No

The number of channels of the output corresponds to the number of channels of the MediaStreamTrack.

[Exposed=Window,
 Constructor (AudioContext context, MediaStreamTrackAudioSourceOptions options)]
interface MediaStreamTrackAudioSourceNode : AudioNode {
};

1.25.1. Constructors

MediaStreamTrackAudioSourceNode(context, options)

Let node be a new MediaStreamTrackAudioSourceNode object. Initialize node, and return node.

Arguments for the MediaStreamTrackAudioSourceNode.MediaStreamTrackAudioSourceNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	AudioContext	✘	✘	The `BaseAudioContext` this new `MediaStreamTrackAudioSourceNode` will be associated with.
`options`	MediaStreamTrackAudioSourceOptions	✘	✘	Initial parameter value for this `MediaStreamTrackAudioSourceNode`.

1.25.2. `MediaStreamTrackAudioSourceOptions`

This specifies the options for constructing a MediaStreamTrackAudioSourceNode. This is required.

dictionary MediaStreamTrackAudioSourceOptions {
  required MediaStreamTrack mediaStreamTrack;
};

1.25.2.1. Dictionary `MediaStreamTrackAudioSourceOptions` Members

mediaStreamTrack, of type MediaStreamTrack: The media stream track that will act as a source. If this MediaStreamTrack kind attribute is not "audio", an InvalidStateError MUST be thrown.

1.26. The `OscillatorNode` Interface

OscillatorNode represents an audio source generating a periodic waveform. It can be set to a few commonly used waveforms. Additionally, it can be set to an arbitrary periodic waveform through the use of a PeriodicWave object.

Oscillators are common foundational building blocks in audio synthesis. An OscillatorNode will start emitting sound at the time specified by the start() method.

Mathematically speaking, a continuous-time periodic waveform can have very high (or infinitely high) frequency information when considered in the frequency domain. When this waveform is sampled as a discrete-time digital audio signal at a particular sample-rate, then care MUST be taken to discard (filter out) the high-frequency information higher than the Nyquist frequency before converting the waveform to a digital form. If this is not done, then aliasing of higher frequencies (than the Nyquist frequency) will fold back as mirror images into frequencies lower than the Nyquist frequency. In many cases this will cause audibly objectionable artifacts. This is a basic and well understood principle of audio DSP.

There are several practical approaches that an implementation may take to avoid this aliasing. Regardless of approach, the idealized discrete-time digital audio signal is well defined mathematically. The trade-off for the implementation is a matter of implementation cost (in terms of CPU usage) versus fidelity to achieving this ideal.

It is expected that an implementation will take some care in achieving this ideal, but it is reasonable to consider lower-quality, less-costly approaches on lower-end hardware.

Both frequency and detune are a-rate parameters, and form a compound parameter. They are used together to determine a computedOscFrequency value:

computedOscFrequency(t) = frequency(t) * pow(2, detune(t) / 1200)

The OscillatorNode’s instantaneous phase at each time is the definite time integral of computedOscFrequency, assuming a phase angle of zero at the node’s exact start time. Its nominal range is [-Nyquist frequency, Nyquist frequency].

Property	Value	Notes
`numberOfInputs`	0
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	No

enum OscillatorType {
  "sine",
  "square",
  "sawtooth",
  "triangle",
  "custom"
};

Enumeration description
"`sine`"	A sine wave
"`square`"	A square wave of duty period 0.5
"`sawtooth`"	A sawtooth wave
"`triangle`"	A triangle wave
"`custom`"	A custom periodic wave

[Exposed=Window,
 Constructor (BaseAudioContext context, optional OscillatorOptions options)]
interface OscillatorNode : AudioScheduledSourceNode {
  attribute OscillatorType type;
  readonly attribute AudioParam frequency;
  readonly attribute AudioParam detune;
  void setPeriodicWave (PeriodicWave periodicWave);
};

1.27. Constructors

OscillatorNode(context, options)

Let node be a new OscillatorNode object. Initialize node, and return node.

Arguments for the OscillatorNode.OscillatorNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `OscillatorNode` will be associated with.
`options`	OscillatorOptions	✘	✔	Optional initial parameter value for this `OscillatorNode`.

1.27.1. Attributes

detune, of type AudioParam, readonly

A detuning value (in cents) which will offset the frequency by the given amount. Its default value is 0. This parameter is a-rate. It forms a compound parameter with frequency to form the computedOscFrequency. The nominal range listed below allows this parameter to detune the frequency over the entire possible range of frequencies.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	\(\approx -153600\)
`maxValue`	\(\approx 153600\)	This value is approximately \(1200 \log_2 \mathrm{FLT\_MAX}\) where FLT_MAX is the largest `float` value.
`automationRate`	"`a-rate`"

frequency, of type AudioParam, readonly

The frequency (in Hertz) of the periodic waveform. Its default value is 440. This parameter is a-rate. It forms a compound parameter with detune to form the computedOscFrequency. Its nominal range is [-Nyquist frequency, Nyquist frequency].

Parameter	Value	Notes
`defaultValue`	440
`minValue`	-Nyquist frequency
`maxValue`	Nyquist frequency
`automationRate`	"`a-rate`"

type, of type OscillatorType

The shape of the periodic waveform. It may directly be set to any of the type constant values except for "custom". Doing so MUST throw an InvalidStateError exception. The setPeriodicWave() method can be used to set a custom waveform, which results in this attribute being set to "custom". The default value is "sine". When this attribute is set, the phase of the oscillator MUST be conserved.

1.27.2. Methods

setPeriodicWave(periodicWave)

Sets an arbitrary custom periodic waveform given a PeriodicWave.

Arguments for the OscillatorNode.setPeriodicWave() method.
Parameter	Type	Nullable	Optional	Description
`periodicWave`	PeriodicWave	✘	✘

Return type: void

1.27.3. `OscillatorOptions`

This specifies the options to be used when constructing an OscillatorNode. All of the members are optional; if not specified, the normal default values are used for constructing the oscillator.

dictionary OscillatorOptions : AudioNodeOptions {
  OscillatorType type = "sine";
  float frequency = 440;
  float detune = 0;
  PeriodicWave periodicWave;
};

1.27.3.1. Dictionary `OscillatorOptions` Members

detune, of type float, defaulting to 0: The initial detune value for the OscillatorNode.
frequency, of type float, defaulting to 440: The initial frequency for the OscillatorNode.
periodicWave, of type PeriodicWave: The PeriodicWave for the OscillatorNode. If this is specified, then any valid value for type is ignored; it is treated as if "custom" were specified.
type, of type OscillatorType, defaulting to "sine": The type of oscillator to be constructed. If this is set to "custom" without also specifying a periodicWave, then an InvalidStateError exception MUST be thrown. If periodicWave is specified, then any valid value for type is ignored; it is treated as if it were set to "custom".

1.27.4. Basic Waveform Phase

The idealized mathematical waveforms for the various oscillator types are defined here. In summary, all waveforms are defined mathematically to be an odd function with a positive slope at time 0. The actual waveforms produced by the oscillator may differ to prevent aliasing affects.

The oscillator MUST produce the same result as if a PeriodicWave with the appropriate Fourier series and with normalization enabled were used to create these basic waveforms.

"sine"

The waveform for sine oscillator is:

$$
  x(t) = \sin t
$$

"square"

The waveform for the square wave oscillator is:

$$
  x(t) = \begin{cases}
         1 & \mbox{for } 0≤ t < \pi \\
         -1 & \mbox{for } -\pi < t < 0.
         \end{cases}
$$

This is extended to all \(t\) by using the fact that the waveform is an odd function with period \(2\pi\).

"sawtooth"

The waveform for the sawtooth oscillator is the ramp:

$$
  x(t) = \frac{t}{\pi} \mbox{ for } -\pi < t ≤ \pi;
$$

This is extended to all \(t\) by using the fact that the waveform is an odd function with period \(2\pi\).

"triangle"

The waveform for the triangle oscillator is:

$$
  x(t) = \begin{cases}
           \frac{2}{\pi} t & \mbox{for } 0 ≤ t ≤ \frac{\pi}{2} \\
           1-\frac{2}{\pi} (t-\frac{\pi}{2}) & \mbox{for }
           \frac{\pi}{2} < t ≤ \pi.
         \end{cases}
$$

This is extended to all \(t\) by using the fact that the waveform is an odd function with period \(2\pi\).

1.28. The `PannerNode` Interface

This interface represents a processing node which positions / spatializes an incoming audio stream in three-dimensional space. The spatialization is in relation to the AudioContext's AudioListener (listener attribute).

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2	Has channelCount constraints
`channelCountMode`	"`clamped-max`"	Has channelCountMode constraints
`channelInterpretation`	"`speakers`"
tail-time reference	Maybe	If the `panningModel` is set to "`HRTF`", the node will produce non-silent output for silent input due to the inherent processing for head responses. Otherwise there are no tail-time references.

The input of this node is either mono (1 channel) or stereo (2 channels) and cannot be increased. Connections from nodes with fewer or more channels will be up-mixed or down-mixed appropriately.

The output of this node is hard-coded to stereo (2 channels) and cannot be configured.

The PanningModelType enum determines which spatialization algorithm will be used to position the audio in 3D space. The default is "equalpower".

enum PanningModelType {
    "equalpower",
    "HRTF"
};

Enumeration description
"`equalpower`"	A simple and efficient spatialization algorithm using equal-power panning. Note: When this panning model is used, all the `AudioParam`s used to compute the output of this node are a-rate.
"`HRTF`"	A higher quality spatialization algorithm using a convolution with measured impulse responses from human subjects. This panning method renders stereo output. Note:When this panning model is used, all the `AudioParam`s used to compute the output of this node are k-rate.

The effective automation rate for an AudioParam of a PannerNode is determined by the panningModel and automationRate of the AudioParam. If the panningModel is "HRTF", the effective automation rate is "k-rate", independent of the setting of the automationRate. Otherwise the effective automation rate is the value of automationRate.

The DistanceModelType enum determines which algorithm will be used to reduce the volume of an audio source as it moves away from the listener. The default is "inverse".

In the description of each distance model below, let \(d\) be the distance between the listener and the panner; \(d_{ref}\) be the value of the refDistance attribute; \(d_{max}\) be the value of the maxDistance attribute; and \(f\) be the value of the rolloffFactor attribute.

enum DistanceModelType {
  "linear",
  "inverse",
  "exponential"
};

Enumeration description

"linear"

A linear distance model which calculates distanceGain according to:

$$
  1 - f\frac{\max(\min(d, d’_{max}), d’_{ref}) - d’_{ref}}{d’_{max} - d’_{ref}}
$$

where \(d’_{ref} = \min(d_{ref}, d_{max})\) and \(d’_{max} = \max(d_{ref}, d_{max})\). In the case where \(d’_{ref} = d’_{max}\), the value of the linear model is taken to be \(1-f\).

Note that \(d\) is clamped to the interval \([d’_{ref},\, d’_{max}]\).

"inverse"

An inverse distance model which calculates distanceGain according to:

$$
  \frac{d_{ref}}{d_{ref} + f (\max(d, d_{ref}) - d_{ref})}
$$

That is, \(d\) is clamped to the interval \([d_{ref},\, \infty)\). If \(d_{ref} = 0\), the value of the inverse model is taken to be 0, independent of the value of \(d\) and \(f\).

"exponential"

An exponential distance model which calculates distanceGain according to:

$$
  \left(\frac{\max(d, d_{ref})}{d_{ref}}\right)^{-f}
$$

That is, \(d\) is clamped to the interval \([d_{ref},\, \infty)\). If \(d_{ref} = 0\), the value of the exponential model is taken to be 0, independent of \(d\) and \(f\).

[Exposed=Window,
 Constructor (BaseAudioContext context, optional PannerOptions options)]
interface PannerNode : AudioNode {
  attribute PanningModelType panningModel;
  readonly attribute AudioParam positionX;
  readonly attribute AudioParam positionY;
  readonly attribute AudioParam positionZ;
  readonly attribute AudioParam orientationX;
  readonly attribute AudioParam orientationY;
  readonly attribute AudioParam orientationZ;
  attribute DistanceModelType distanceModel;
  attribute double refDistance;
  attribute double maxDistance;
  attribute double rolloffFactor;
  attribute double coneInnerAngle;
  attribute double coneOuterAngle;
  attribute double coneOuterGain;
  void setPosition (float x, float y, float z);
  void setOrientation (float x, float y, float z);
};

1.28.1. Constructors

PannerNode(context, options)

If options is given, check for valid values. If values are not valid, an error is thrown. Validity is determined as if the attribute of the same name as the options member were set to the given value. If an error would be thrown for when setting this value, the same error MUST be thrown when constructing this node.

Otherwise, let node be a new PannerNode object. Initialize node, and return node.

Arguments for the PannerNode.PannerNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `PannerNode` will be associated with.
`options`	PannerOptions	✘	✔	Optional initial parameter value for this `PannerNode`.

1.28.2. Attributes

coneInnerAngle, of type double

A parameter for directional audio sources, this is an angle, in degrees, inside of which there will be no volume reduction. The default value is 360. The behavior is undefined if the angle is outside the interval [0, 360].

coneOuterAngle, of type double

A parameter for directional audio sources, this is an angle, in degrees, outside of which the volume will be reduced to a constant value of coneOuterGain. The default value is 360. The behavior is undefined if the angle is outside the interval [0, 360].

coneOuterGain, of type double

A parameter for directional audio sources, this is the gain outside of the coneOuterAngle. The default value is 0. It is a linear value (not dB) in the range [0, 1]. An InvalidStateError MUST be thrown if the parameter is outside this range.

distanceModel, of type DistanceModelType

Specifies the distance model used by this PannerNode. Defaults to "inverse".

maxDistance, of type double

The maximum distance between source and listener, after which the volume will not be reduced any further. The default value is 10000. A RangeError exception MUST be thrown if this is set to a non-positive value.

orientationX, of type AudioParam, readonly

Describes the x component of the vector of the direction the audio source is pointing in 3D Cartesian coordinate space. Depending on how directional the sound is (controlled by the cone attributes), a sound pointing away from the listener can be very quiet or completely silent.

Parameter	Value	Notes
`defaultValue`	1
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"	Has automation rate constraints

orientationY, of type AudioParam, readonly

Describes the y component of the vector of the direction the audio source is pointing in 3D cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"	Has automation rate constraints

orientationZ, of type AudioParam, readonly

Describes the Z component of the vector of the direction the audio source is pointing in 3D cartesian coordinate space.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"	Has automation rate constraints

panningModel, of type PanningModelType

Specifies the panning model used by this PannerNode. Defaults to "equalpower".

positionX, of type AudioParam, readonly

Sets the x coordinate position of the audio source in a 3D Cartesian system.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"	Has automation rate constraints

positionY, of type AudioParam, readonly

Sets the y coordinate position of the audio source in a 3D Cartesian system.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"	Has automation rate constraints

positionZ, of type AudioParam, readonly

Sets the z coordinate position of the audio source in a 3D Cartesian system.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	most-negative-single-float	Approximately -3.4028235e38
`maxValue`	most-positive-single-float	Approximately 3.4028235e38
`automationRate`	"`a-rate`"	Has automation rate constraints

refDistance, of type double

A reference distance for reducing volume as source moves further from the listener. The default value is 1. A RangeError exception MUST be thrown if this is set to a negative value.

rolloffFactor, of type double

Describes how quickly the volume is reduced as source moves away from listener. The default value is 1. A RangeError exception MUST be thrown if this is set to a negative value.

The nominal range for the rolloffFactor specifies the minimum and maximum values the rolloffFactor can have. Values outside the range are clamped to lie within this range. The nominal range depends on the distanceModel as follows:

"linear": The nominal range is \([0, 1]\).
"inverse": The nominal range is \([0, \infty)\).
"exponential": The nominal range is \([0, \infty)\).

1.28.3. Methods

setOrientation(x, y, z)

This method is DEPRECATED. It is equivalent to setting orientationX.value, orientationY.value, and orientationZ.value attribute directly, with the x, y and z parameters, respectively.

Consequently, if any of the orientationX, orientationY, and orientationZ AudioParams have an automation curve set using setValueCurveAtTime() at the time this method is called, a NotSupportedError MUST be thrown.

Describes which direction the audio source is pointing in the 3D cartesian coordinate space. Depending on how directional the sound is (controlled by the cone attributes), a sound pointing away from the listener can be very quiet or completely silent.

The x, y, z parameters represent a direction vector in 3D space.

The default value is (1,0,0).

Arguments for the PannerNode.setOrientation() method.
Parameter	Type	Nullable	Optional
`x`	float	✘	✘
`y`	float	✘	✘
`z`	float	✘	✘

Return type: void

setPosition(x, y, z)

This method is DEPRECATED. It is equivalent to setting positionX.value, positionY.value, and positionZ.value attribute directly with the x, y and z parameters, respectively.

Consequently, if any of the positionX, positionY, and positionZ AudioParams have an automation curve set using setValueCurveAtTime() at the time this method is called, a NotSupportedError MUST be thrown.

Sets the position of the audio source relative to the listener attribute. A 3D cartesian coordinate system is used.

The x, y, z parameters represent the coordinates in 3D space.

The default value is (0,0,0).

Arguments for the PannerNode.setPosition() method.
Parameter	Type	Nullable	Optional
`x`	float	✘	✘
`y`	float	✘	✘
`z`	float	✘	✘

Return type: void

1.28.4. `PannerOptions`

This specifies options for constructing a PannerNode. All members are optional; if not specified, the normal default is used in constructing the node.

dictionary PannerOptions : AudioNodeOptions {
  PanningModelType panningModel = "equalpower";
  DistanceModelType distanceModel = "inverse";
  float positionX = 0;
  float positionY = 0;
  float positionZ = 0;
  float orientationX = 1;
  float orientationY = 0;
  float orientationZ = 0;
  double refDistance = 1;
  double maxDistance = 10000;
  double rolloffFactor = 1;
  double coneInnerAngle = 360;
  double coneOuterAngle = 360;
  double coneOuterGain = 0;
};

1.28.4.1. Dictionary `PannerOptions` Members

coneInnerAngle, of type double, defaulting to 360: The initial value for the coneInnerAngle attribute of the node.
coneOuterAngle, of type double, defaulting to 360: The initial value for the coneOuterAngle attribute of the node.
coneOuterGain, of type double, defaulting to 0: The initial value for the coneOuterGain attribute of the node.
distanceModel, of type DistanceModelType, defaulting to "inverse": The distance model to use for the node.
maxDistance, of type double, defaulting to 10000: The initial value for the maxDistance attribute of the node.
orientationX, of type float, defaulting to 1: The initial X value for the orientationX AudioParam.
orientationY, of type float, defaulting to 0: The initial Y value for the orientationY AudioParam.
orientationZ, of type float, defaulting to 0: The initial Z value for the orientationZ AudioParam.
panningModel, of type PanningModelType, defaulting to "equalpower": The panning model to use for the node.
positionX, of type float, defaulting to 0: The initial X value for the positionX AudioParam.
positionY, of type float, defaulting to 0: The initial Y value for the positionY AudioParam.
positionZ, of type float, defaulting to 0: The initial Z value for the positionZ AudioParam.
refDistance, of type double, defaulting to 1: The initial value for the refDistance attribute of the node.
rolloffFactor, of type double, defaulting to 1: The initial value for the rolloffFactor attribute of the node.

1.28.5. Channel Limitations

The set of channel limitations for StereoPannerNode also apply to PannerNode.

1.29. The `PeriodicWave` Interface

PeriodicWave represents an arbitrary periodic waveform to be used with an OscillatorNode.

A conforming implementation MUST support PeriodicWave up to at least 8192 elements.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional PeriodicWaveOptions options)]
interface PeriodicWave {
};

1.29.1. Constructors

PeriodicWave(context, options)

Let p be a new PeriodicWave object. Let [[associated context]] be a reference to the BaseAudioContext passed as first argument of this constructor.
If the real and imag parameters of the PeriodicWaveOptions are not of the same length, an IndexSizeError exception MUST be thrown.
If options has not been passed in, let [[real]] and [[imag]] be two internal slots of type Float32Array and length 2. Set the second element of the [[imag]] array to 1.

Note: When setting this PeriodicWave on an OscillatorNode, this is equivalent to using the built-in type "sine".
Otherwise, let [[real]] and [[imag]] be two internal slots of type Float32Array, both of length equal to the maximum length of the real and imag attributes of the PeriodicWaveOptions passed in. Make a copy of those arrays into their respective internal slots.
Let [[normalize]] be an internal slot of the PeriodicWave that is initialized to the inverse of the disableNormalization attribute of the PeriodicWaveConstraints on the PeriodicWaveOptions.
Return p.

Arguments for the PeriodicWave.PeriodicWave() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `PeriodicWave` will be associated with. Unlike `AudioBuffer`, `PeriodicWave`s can’t be shared accross `AudioContext`s or `OfflineAudioContext`s. It is associated with a particular `BaseAudioContext`.
`options`	PeriodicWaveOptions	✘	✔	Optional initial parameter value for this `PeriodicWave`.

1.29.2. `PeriodicWaveConstraints`

The PeriodicWaveConstraints dictionary is used to specify how the waveform is normalized.

dictionary PeriodicWaveConstraints {
  boolean disableNormalization = false;
};

1.29.2.1. Dictionary `PeriodicWaveConstraints` Members

disableNormalization, of type boolean, defaulting to false: Controls whether the periodic wave is normalized or not. If true, the waveform is not normalized; otherwise, the waveform is normalized.

1.29.3. `PeriodicWaveOptions`

The PeriodicWaveOptions dictionary is used to specify how the waveform is constructed. If only one of real or imag is specified. the other is treated as if it were an array of all zeroes of the same length, as specified below in description of the dictionary members. If neither is given, a PeriodicWave is created that MUST be equivalent to an OscillatorNode with type "sine". If both are given, the sequences must have the same length; otherwise an error of type NotSupportedError MUST be thrown.

dictionary PeriodicWaveOptions : PeriodicWaveConstraints {
  sequence<float> real;
  sequence<float> imag;
};

1.29.3.1. Dictionary `PeriodicWaveOptions` Members

imag, of type sequence<float>

The imag parameter represents an array of sine terms. The first element (index 0) does not exist in the Fourier series. Implementations MUST set it to zero when computing the waveform. The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.

This defaults to a sequence of all zeroes of the same length as real if real is given.

real, of type sequence<float>

The real parameter represents an array of cosine terms. The first element (index 0) is the DC-offset of the periodic waveform. Implementations MUST set it to zero when computing the waveform. The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.

This defaults to a sequence of all zeroes of the same length as imag if imag is given.

1.29.4. Waveform Generation

The createPeriodicWave() method takes two arrays to specify the Fourier coefficients of the PeriodicWave. Let \(a\) and \(b\) represent the real and imaginary arrays of length \(L\). Then the basic time-domain waveform, \(x(t)\), can be computed using:

$$
  x(t) = \sum_{k=1}^{L-1} \left(a[k]\cos2\pi k t + b[k]\sin2\pi k t\right)
$$

This is the basic (unnormalized) waveform.

1.29.5. Waveform Normalization

If the internal slot [[normalize]] of this PeriodicWave is true (the default), the waveform defined in the previous section is normalized so that the maximum value is 1. The normalization is done as follows.

Let

$$
  \tilde{x}(n) = \sum_{k=1}^{L-1} \left(a[k]\cos\frac{2\pi k n}{N} + b[k]\sin\frac{2\pi k n}{N}\right)
$$

where \(N\) is a power of two. (Note: \(\tilde{x}(n)\) can conveniently be computed using an inverse FFT.) The fixed normalization factor \(f\) is computed as follows.

$$
  f = \max_{n = 0, \ldots, N - 1} |\tilde{x}(n)|
$$

Thus, the actual normalized waveform \(\hat{x}(n)\) is:

$$
  \hat{x}(n) = \frac{\tilde{x}(n)}{f}
$$

This fixed normalization factor MUST be applied to all generated waveforms.

1.29.6. Oscillator Coefficients

The builtin oscillator types are created using PeriodicWave objects. For completeness the coefficients for the PeriodicWave for each of the builtin oscillator types is given here. This is useful if a builtin type is desired but without the default normalization.

In the following descriptions, let \(a\) be the array of real coefficients and \(b\) be the array of imaginary coefficients for createPeriodicWave(). In all cases \(a[n] = 0\) for all \(n\) because the waveforms are odd functions. Also, \(b[0] = 0\) in all cases. Hence, only \(b[n]\) for \(n \ge 1\) is specified below.

"sine"

$$
  b[n] = \begin{cases}
           1 & \mbox{for } n = 1 \\
           0 & \mbox{otherwise}
         \end{cases}
$$

"square"

$$
  b[n] = \frac{2}{n\pi}\left[1 - (-1)^n\right]
$$

"sawtooth"

$$
  b[n] = (-1)^{n+1} \dfrac{2}{n\pi}
$$

"triangle"

$$
  b[n] = \frac{8\sin\dfrac{n\pi}{2}}{(\pi n)^2}
$$

1.30. The `ScriptProcessorNode` Interface - DEPRECATED

This interface is an AudioNode which can generate, process, or analyse audio directly using a script. This node type is deprecated, to be replaced by the AudioWorkletNode; this text is only here for informative purposes until implementations remove this node type.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	`numberOfInputChannels`	This is the number of channels specified when constructing this node. There are channelCount constraints.
`channelCountMode`	"`explicit`"	Has channelCountMode constraints
`channelInterpretation`	"`speakers`"
tail-time reference	No

The ScriptProcessorNode is constructed with a bufferSize which MUST be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. This value controls how frequently the onaudioprocess event is dispatched and how many sample-frames need to be processed each call. onaudioprocess events are only dispatched if the ScriptProcessorNode has at least one input or one output connected. Lower numbers for bufferSize will result in a lower (better) latency. Higher numbers will be necessary to avoid audio breakup and glitches. This value will be picked by the implementation if the bufferSize argument to createScriptProcessor() is not passed in, or is set to 0.

numberOfInputChannels and numberOfOutputChannels determine the number of input and output channels. It is invalid for both numberOfInputChannels and numberOfOutputChannels to be zero.

[Exposed=Window]
interface ScriptProcessorNode : AudioNode {
  attribute EventHandler onaudioprocess;
  readonly attribute long bufferSize;
};

1.30.1. Attributes

bufferSize, of type long, readonly: The size of the buffer (in sample-frames) which needs to be processed each time onaudioprocess is called. Legal values are (256, 512, 1024, 2048, 4096, 8192, 16384).
onaudioprocess, of type EventHandler: A property used to set the EventHandler (described in HTML[HTML]) for the onaudioprocess event that is dispatched to ScriptProcessorNode node types. An event of type AudioProcessingEvent will be dispatched to the event handler.

1.31. The `StereoPannerNode` Interface

This interface represents a processing node which positions an incoming audio stream in a stereo image using a low-cost equal-power panning algorithm. This panning effect is common in positioning audio components in a stereo stream.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2	Has channelCount constraints
`channelCountMode`	"`clamped-max`"	Has channelCountMode constraints
`channelInterpretation`	"`speakers`"
tail-time reference	No

The input of this node is stereo (2 channels) and cannot be increased. Connections from nodes with fewer or more channels will be up-mixed or down-mixed appropriately.

The output of this node is hard-coded to stereo (2 channels) and cannot be configured.

[Exposed=Window,
 Constructor (BaseAudioContext context, optional StereoPannerOptions options)]
interface StereoPannerNode : AudioNode {
  readonly attribute AudioParam pan;
};

1.31.1. Constructors

StereoPannerNode(context, options)

Let node be a new StereoPannerNode object. Initialize node, and return node.

Arguments for the StereoPannerNode.StereoPannerNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `StereoPannerNode` will be associated with.
`options`	StereoPannerOptions	✘	✔	Optional initial parameter value for this `StereoPannerNode`.

1.31.2. Attributes

pan, of type AudioParam, readonly

The position of the input in the output’s stereo image. -1 represents full left, +1 represents full right.

Parameter	Value	Notes
`defaultValue`	0
`minValue`	-1
`maxValue`	1
`automationRate`	"`a-rate`"

1.31.3. `StereoPannerOptions`

This specifies the options to use in constructing a StereoPannerNode. All members are optional; if not specified, the normal default is used in constructing the node.

dictionary StereoPannerOptions : AudioNodeOptions {
  float pan = 0;
};

1.31.3.1. Dictionary `StereoPannerOptions` Members

pan, of type float, defaulting to 0: The initial value for the pan AudioParam.

1.31.4. Channel Limitations

Because its processing is constrained by the above definitions, StereoPannerNode is limited to mixing no more than 2 channels of audio, and producing exactly 2 channels. It is possible to use a ChannelSplitterNode, intermediate processing by a subgraph of GainNodes and/or other nodes, and recombination via a ChannelMergerNode to realize arbitrary approaches to panning and mixing.

1.32. The `WaveShaperNode` Interface

WaveShaperNode is an AudioNode processor implementing non-linear distortion effects.

Non-linear waveshaping distortion is commonly used for both subtle non-linear warming, or more obvious distortion effects. Arbitrary non-linear shaping curves may be specified.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	Maybe	There is a tail-time reference only if the `oversample` attribute is set to "`2x`" or "`4x`". The actual duration of this tail-time depends on the implementation.

The number of channels of the output always equals the number of channels of the input.

WaveShaperNodes are created with an internal flag curve set, initially set to false.

enum OverSampleType {
  "none",
  "2x",
  "4x"
};

Enumeration description
"`none`"	Don’t oversample
"`2x`"	Oversample two times
"`4x`"	Oversample four times

[Exposed=Window,
 Constructor (BaseAudioContext context, optional WaveShaperOptions options)]
interface WaveShaperNode : AudioNode {
  attribute Float32Array? curve;
  attribute OverSampleType oversample;
};

1.32.1. Constructors

WaveShaperNode(context, options)

Let node be a new WaveShaperNode object. Initialize node. Set an internal boolean slot [[curve set]], and initialize it to false. Return node.

Arguments for the WaveShaperNode.WaveShaperNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `WaveShaperNode` will be associated with.
`options`	WaveShaperOptions	✘	✔	Optional initial parameter value for this `WaveShaperNode`.

1.32.2. Attributes

curve, of type Float32Array, nullable

The shaping curve used for the waveshaping effect. The input signal is nominally within the range [-1; 1]. Each input sample within this range will index into the shaping curve, with a signal level of zero corresponding to the center value of the curve array if there are an odd number of entries, or interpolated between the two centermost values if there are an even number of entries in the array. Any sample value less than -1 will correspond to the first value in the curve array. Any sample value greater than +1 will correspond to the last value in the curve array.

The implementation MUST perform linear interpolation between adjacent points in the curve. Initially the curve attribute is null, which means that the WaveShaperNode will pass its input to its output without modification.

Values of the curve are spread with equal spacing in the [-1; 1] range. This means that a curve with a even number of value will not have a value for a signal at zero, and a curve with an odd number of value will have a value for a signal at zero.

A InvalidStateError MUST be thrown if this attribute is set with a Float32Array that has a length less than 2.

When this attribute is set, an internal copy of the curve is created by the WaveShaperNode. Subsequent modifications of the contents of the array used to set the attribute therefore have no effect: the attribute MUST be set again in order to change the curve.

To set the curve attribute, execute these steps:

Let new curve be a Float32Array to be assigned to curve or null. .
If new curve is not null and [[curve set]] is true, throw an InvalidStateError and abort these steps.
If new curve is not null, set [[curve set]] to true.
Assign new curve to the curve attribute.

oversample, of type OverSampleType

Specifies what type of oversampling (if any) should be used when applying the shaping curve. The default value is "none", meaning the curve will be applied directly to the input samples. A value of "2x" or "4x" can improve the quality of the processing by avoiding some aliasing, with the "4x" value yielding the highest quality. For some applications, it’s better to use no oversampling in order to get a very precise shaping curve.

A value of "2x" or "4x" means that the following steps MUST be performed:

Up-sample the input samples to 2x or 4x the sample-rate of the AudioContext. Thus for each render quantum, generate 256 (for 2x) or 512 (for 4x) samples.
Apply the shaping curve.
Down-sample the result back to the sample-rate of the AudioContext. Thus taking the 256 (or 512) processed samples, generating 128 as the final result.

The exact up-sampling and down-sampling filters are not specified, and can be tuned for sound quality (low aliasing, etc.), low latency, and performance.

Note: Use of oversampling introduces some degree of audio processing latency due to the up-sampling and down-sampling filters. The amount of this latency can vary from one implementation to another.

1.32.3. `WaveShaperOptions`

This specifies the options for constructing a WaveShaperNode. All members are optional; if not specified, the normal default is used in constructing the node.

dictionary WaveShaperOptions : AudioNodeOptions {
  sequence<float> curve;
  OverSampleType oversample = "none";
};

1.32.3.1. Dictionary `WaveShaperOptions` Members

curve, of type sequence<float>: The shaping curve for the waveshaping effect.
oversample, of type OverSampleType, defaulting to "none": The type of oversampling to use for the shaping curve.

1.33. The `AudioWorklet` Interface

[Exposed=Window, SecureContext]
interface AudioWorklet : Worklet {
};

1.33.1. Concepts

The AudioWorklet object allows developers to supply scripts (such as JavaScript or WebAssembly code) to process audio on the rendering thread, supporting custom AudioNodes. This processing mechanism ensures the synchronous execution of the script code with other built-in AudioNodes in the audio graph.

An associated pair of objects MUST be defined in order to realize this mechanism: AudioWorkletNode and AudioWorkletProcessor. The former represents the interface for the main global scope similar to other AudioNode objects, and the latter implements the internal audio processing within a special scope named AudioWorkletGlobalScope.

AudioWorklet concept — `AudioWorkletNode` and `AudioWorkletProcessor`

Each BaseAudioContext possesses exactly one AudioWorklet. Scripts imported using this AudioWorklet are run in one or more AudioWorkletGlobalScope global scopes, which are created to process audio associated with that BaseAudioContext.

Importing a script via the addModule(moduleUrl) method registers class definitions of AudioWorkletProcessor under the AudioWorkletGlobalScope. There are two internal storage areas for the imported class definitions and the active instances created from the definition.

node name to processor definition map: Belongs to AudioWorkletGlobalScope. This map associates a string key to the corresponding AudioWorkletProcessor definition. Initially this map is empty and becomes populated when registerProcessor() method is called.
node name to parameter descriptor map: Belongs to BaseAudioContext. This map contains an identical set of string keys from node name to processor definition map that are associated with the matching parameterDescriptors values. This internal storage is populated as a consequence of calling the registerProcessor() method in the rendering thread. The population is guaranteed to complete prior to the resolution of the promise returned by addModule() on a context’s audioWorklet.

// bypass-processor.js script file, runs on AudioWorkletGlobalScope
class BypassProcessor extends AudioWorkletProcessor {
  process (inputs, outputs) {
    // Single input, single channel.
    let input = inputs[0];
    let output = outputs[0];
    output[0].set(input[0]);

    // Process only while there are inputs.
    return false;
  }
});

registerProcessor('bypass-processor', BypassProcessor);

// The main global scope
let context = new AudioContext();
context.audioWorklet.addModule('bypass-processor.js').then(() => {
  let bypassNode = new AudioWorkletNode(context, 'bypass-processor');
});

At the instantiation of AudioWorkletNode in the main global scope, the counterpart AudioWorkletProcessor will also be created in AudioWorkletGlobalScope. These two objects communicate via the asynchronous message passing described in §2 Processing model.

1.33.2. The `AudioWorkletGlobalScope` Interface

This special execution context is designed to enable the generation, processing, and analysis of audio data directly using a script in the audio rendering thread. The user-supplied script code is evaluated in this scope to define one or more AudioWorkletProcessor subclasses, which in turn are used to instantiate AudioWorkletProcessors, in a 1:1 association with AudioWorkletNodes in the main scope.

At least one AudioWorkletGlobalScope exists for each AudioContext that contains one or more AudioWorkletNodes. The running of imported scripts is performed by the UA as defined in [worklets-1], in such a way that all scripts are applied consistently to every global scope, and all scopes thus exhibit identical behavior. Beyond these guarantees, the creation of global scopes is transparent to the author and cannot be observed from the main window scope.

AudioWorkletGlobalScope has a node name to processor definition map. This map stores definitions of AudioWorkletProcessor with the associated string key. Initially it is empty and populated when registerProcessor() method is called, but this storage is internal and is not directly exposed to the user.

Note: The AudioWorkletGlobalScope may also contain any other data and code to be shared by these instances. As an example, multiple processors might share an ArrayBuffer defining a wavetable or an impulse response.

Note: Every AudioWorkletGlobalScope is associated with a single BaseAudioContext, and with a single audio rendering thread for that context. This prevents data races from occurring in global scope code running within concurrent threads.

[Global=(Worklet, AudioWorklet), Exposed=AudioWorklet]
interface AudioWorkletGlobalScope : WorkletGlobalScope {
  void registerProcessor (DOMString name, VoidFunction processorCtor);
  readonly attribute unsigned long long currentFrame;
  readonly attribute double currentTime;
  readonly attribute float sampleRate;
};

1.33.2.1. Attributes

currentFrame, of type unsigned long long, readonly: The current frame of the block of audio being processed. This must be equal to the value of the [[current frame]] internal slot of the BaseAudioContext.
currentTime, of type double, readonly: The context time of the block of audio being processed. By definition this will be equal to the value of BaseAudioContext's currentTime attribute that was most recently observable in the control thread.
sampleRate, of type float, readonly: The sample rate of the associated BaseAudioContext.

1.33.2.2. Methods

registerProcessor(name, processorCtor)

Registers a class definition derived from AudioWorkletProcessor.

When the registerProcessor(name, processorCtor) method is called, the user agent MUST run the following steps:

If the name is the empty string, throw a NotSupportedError exception and abort these steps because the empty string is not a valid key.
If the name exists as a key in the node name to processor definition map, throw a NotSupportedError exception and abort these steps because registering a definition with a duplicated key is not allowed.
If the result of IsConstructor(argument=processorCtor) is false, throw a TypeError and abort these steps.
Let prototype be the result of Get(O=processorCtor, P="prototype").
If the result of Type(argument=prototype) is not Object, throw a TypeError and abort all these steps.
If the result of IsCallable(argument=Get(O=prototype, P="process")) is false, throw a TypeError and abort these steps.
Let descriptors be the result of Get(O=processorCtor, P="parameterDescriptors").
If descriptors is neither an array nor undefined, throw a TypeError and abort these steps.
Let definition be a new AudioWorkletProcessor definition with:
- node name being name
- processor class constructor being processorCtor
Add the key-value pair (name - definition) to the node name to processor definition map of the associated AudioWorkletGlobalScope.
Queue a task to the control thread to add the key-value pair (name - descriptors) to the node name to parameter descriptor map of the associated BaseAudioContext.

Note: The class constructor should only be looked up once, thus it does not have the opportunity to dynamically change its definition.

Arguments for the AudioWorkletGlobalScope.registerProcessor(name, processorCtor) method.
Parameter	Type	Nullable	Optional	Description
`name`	DOMString	✘	✘	A string key that represents a class definition to be registered. This key is used to look up the constructor of `AudioWorkletProcessor` during construction of an `AudioWorkletNode`.
`processorCtor`	VoidFunction	✘	✘	A class definition extended from `AudioWorkletProcessor`.

Return type: void

1.33.3. The `AudioWorkletNode` Interface

This interface represents a user-defined AudioNode which lives on the control thread. The user can create an AudioWorkletNode from an BaseAudioContext, and such a node can be connected with other built-in AudioNodes to form an audio graph.

Property	Value	Notes
`numberOfInputs`	1
`numberOfOutputs`	1
`channelCount`	2
`channelCountMode`	"`max`"
`channelInterpretation`	"`speakers`"
tail-time reference	See notes	Any tail-time is handled by the node itself

Every AudioWorkletNode has an associated processor reference, initially null, which refers to the AudioWorkletProcessor handling the processing for this node.

Every AudioWorkletProcessor has an associated active source flag, initially true. This flag causes the node to be retained in memory and perform audio processing in the absence of any connected inputs.

[Exposed=Window]
interface AudioParamMap {
  readonly maplike<DOMString, AudioParam>;
};

This interface has "entries", "forEach", "get", "has", "keys", "values", @@iterator methods and a "size" getter brought by readonly maplike.

[Exposed=Window,
 SecureContext,
 Constructor (BaseAudioContext context, DOMString name, optional AudioWorkletNodeOptions options)]
interface AudioWorkletNode : AudioNode {
  readonly attribute AudioParamMap parameters;
  readonly attribute MessagePort port;
  attribute EventHandler onprocessorerror;
};

1.33.3.1. Constructors

AudioWorkletNode(context, name, options)

Let node be a new AudioWorkletNode object. Initialize node. Perform the construction procedure of an AudioWorkletNode and the corresponding AudioWorkletProcessor object. Return node.

Arguments for the AudioWorkletNode.AudioWorkletNode() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `AudioWorkletNode` will be associated with.
`name`	DOMString	✘	✘	A string that can be used as a key in the `BaseAudioContext`’s node name to parameter descriptor map.
`options`	AudioWorkletNodeOptions	✘	✔	Optional initial parameters value for this `AudioWorkletNode`.

1.33.3.2. Attributes

onprocessorerror, of type EventHandler: When an exception is thrown from the processor’s constructor, process method, or any user-defined class method, the processor will queue a task on the control thread to fire onprocessorerror event to the node. Note that once an exception is thrown, the processor will output silence throughout its lifetime.
parameters, of type AudioParamMap, readonly: The parameters attribute is a collection of AudioParam objects with associated names. This maplike object is populated from a list of AudioParamDescriptors in the AudioWorkletProcessor class definition at the instantiation.
port, of type MessagePort, readonly: Every AudioWorkletNode has an associated port which is a MessagePort. It is connected to the port on the corresponding AudioWorkletProcessor object allowing bidirectional communication between a pair of AudioWorkletNode and AudioWorkletProcessor.

1.33.3.3. `AudioWorkletNodeOptions`

The AudioWorkletNodeOptions dictionary can be used for the custom initialization of AudioNode attributes in the AudioWorkletNode constructor.

dictionary AudioWorkletNodeOptions : AudioNodeOptions {
  unsigned long numberOfInputs = 1;
  unsigned long numberOfOutputs = 1;
  sequence<unsigned long> outputChannelCount;
  record<DOMString, double> parameterData;
  object? processorOptions = null;
};

1.33.3.3.1. Dictionary `AudioWorkletNodeOptions` Members

numberOfInputs, of type unsigned long, defaulting to 1: This is used to initialize the value of AudioNode numberOfInputs attribute.
numberOfOutputs, of type unsigned long, defaulting to 1: This is used to initialize the value of AudioNode numberOfOutputs attribute.
outputChannelCount, of type sequence<unsigned long>: This array is used to configure the number of channels in each output. For example, outputChannelCount: [n, m] specifies the number of channels in the first output to n and the second output to m respectively. IndexSizeError MUST be thrown if the length of sequence does not match numberOfOutputs. A NotSupportedError exception MUST be thrown if a channel count is not in the valid range of AudioNode’s channelCount.
parameterData, of type record<DOMString, double>: This is a list of user-defined key-value pairs that are used to initialize AudioParam values in AudioWorkletNode. If the string key of an entry in the list does not match any name of AudioParam objects in the node, it is ignored.
processorOptions, of type object, nullable, defaulting to null: This holds any additional user-defined data that may be used to initialize the corresponding AudioWorkletProcessor for this AudioWorkletNode.

1.33.3.3.2. Configuring Channels with `AudioWorkletNodeOptions`

With a combination of numberOfInputs, numberOfOutputs, and outputChannelCount, various channel configurations can be achieved.

numberOfInputs = 0, numberOfOutputs = 0: NotSupportedError MUST be thrown by the constructor.
numberOfInputs = 1, numberOfOutputs = 1: If outputChannelCount is unspecified, the output channel count will match computedNumberOfChannels from the input. If outputChannelCount is specified with one value, the output channel count will match the given value.
All other cases: If outputChannelCount is unspecified, it will be mono for all outputs. If specified, the values will be used as given.

1.33.4. The `AudioWorkletProcessor` Interface

This interface represents an audio processing code that runs on the audio rendering thread. It lives in an AudioWorkletGlobalScope and the definition of the class manifests the actual audio processing mechanism of a custom audio node. AudioWorkletProcessor can only be instantiated by the construction of an AudioWorkletNode instance. Every AudioWorkletProcessor has an associated node reference, initially null.

[Exposed=AudioWorklet,
Constructor (optional AudioWorkletNodeOptions options)]
interface AudioWorkletProcessor {
  readonly attribute MessagePort port;
};

1.33.4.1. Constructors

AudioWorkletProcessor(options)

When the constructor for AudioWorkletProcessor is invoked, the following steps are performed:

Compare the value of NewTarget with the active function object; if the two are equal, throw a TypeError.

Note: This check prevents the invocation of the constructor directly from JavaScript. The interface object may only be called internally by the UA.
Initialize the AudioWorkletProcessor using the contents of the options dictionary that was passed to the constructor for AudioWorkletNode. The contents of this dictionary will have been serialized and deserialized according to the algorithm for instantiating an AudioWorkletProcessor.

Note: The base implementation of AudioWorkletProcessor does not examine the values in the options dictionary. However, processor classes that extend AudioWorkletProcessor will use this dictionary to initialize themselves based on the options relevant to their specific node type.

Arguments for the AudioWorkletProcessor.AudioWorkletProcessor() method.
Parameter	Type	Nullable	Optional	Description
`options`	AudioWorkletNodeOptions	✘	✔	User-specified dictionary for custom initializations.

1.33.4.2. Attributes

port, of type MessagePort, readonly: Every AudioWorkletProcessor has an associated port which is a MessagePort. It is connected to the port on the corresponding AudioWorkletProcessor object allowing bidirectional communication between a pair of AudioWorkletNode and AudioWorkletProcessor.

1.33.4.3. Methods

User can define a custom audio processor by extending AudioWorkletProcessor. The subclass MUST define a method named process() that implements the audio processing algorithm and may have a valid static property named parameterDescriptors which is an iterable of AudioParamDescriptor that is looked up by the AudioWorkletProcessor constructor to create instances of AudioParam in the parameters maplike storage in the node. The prototype type check step and the process method is callable step of registerProcessor() ensure the validity of a given AudioWorkletProcessor subclass.

process(inputs, outputs, parameters)

Implements the audio processing algorithm for the AudioWorkletProcessor.

The process() method is called synchronously by the audio rendering thread at every render quantum, if ANY of the following active processing conditions are true:

The associated AudioWorkletProcessor's active source flag is equal to true.
There are one or more connected inputs to the AudioWorkletNode.

The return value of this method controls the lifetime of the AudioWorkletProcessor's associated AudioWorkletNode. At the conclusion of each call to the process() method, ToBoolean (described in [ECMASCRIPT]) is applied to the return value and the result is assigned to the associated AudioWorkletProcessor’s active source flag. This in turn affects whether subsequent invocations of process() occur, and has an impact on the lifetime of the node. To propagate the flag change queue a task on the control thread to update the corresponding AudioWorkletNode's state property accordingly.

This lifetime policy can support a variety of approaches found in built-in nodes, including the following:

Nodes that transform their inputs, and are active only while connected inputs and/or script references exist. Such nodes SHOULD return false from process() which allows the presence or absence of connected inputs to determine whether active processing occurs.
Nodes that transform their inputs, but which remain active for a tail-time after their inputs are disconnected. In this case, process() SHOULD return true for some period of time after inputs is found to contain zero channels. The current time may be obtained from the global scope’s currentTime to measure the start and end of this tail-time interval, or the interval could be calculated dynamically depending on the processor’s internal state.
Nodes that act as sources of output, typically with a lifetime. Such nodes SHOULD return true from process() until the point at which they are no longer producing an output.

Note that the preceding definition implies that when no return value is provided from an implementation of process(), the effect is identical to returning false (since the effective return value is the falsy value undefined). This is a reasonable behavior for any AudioWorkletProcessor that is active only when it has active inputs.

If process() is not called during some rendering quantum due to the lack of any applicable active processing conditions, the result is as if the processor emitted silence for this period.

Arguments for the AudioWorkletProcessor.process(inputs, outputs, parameters)) method.
Parameter	Type	Nullable	Optional	Description
`inputs`				The input audio buffer from the incoming connections provided by the user agent. It has type `sequence<sequence<Float32Array>>`. `inputs[n][m]` is a `Float32Array` of audio samples for the \(m\)th channel of the \(n\)th input. While the number of inputs is fixed at construction, the number of channels can be changed dynamically based on computedNumberOfChannels. If no connections exist to the \(n\)th input of the node during the current render quantum, then the content of `inputs[n]` is an empty array, indicating that zero channels of input are available. This is the only circumstance under which the number of elements of `inputs[n]` can be zero.
`outputs`				The output audio buffer that is to be consumed by the user agent. It has type `<sequence<sequence<Float32Array>>`. `outputs[n][m]` is a `Float32Array` object containing the audio samples for \(m\)th channel of \(n\)th output. Each of the `Float32Array`s are zero-filled. The number of channels in the output will match computedNumberOfChannels only when the node has single output.
`parameters`				A map of string keys and associated `Float32Array`s. `parameters["name"]` corresponds to the automation values of the AudioParam named `"name"`. For each array, the array contains the computedValue or the parameter for all frames in the render quantum. However, if no automation is scheduled during this render quantum, the array MAY have length 1 with the array element being the constant value of the `AudioParam` for the render quantum.

Return type: boolean

The example below shows how AudioParam can be defined and used in an AudioWorkletProcessor.

class MyProcessor extends AudioWorkletProcessor {
  static get parameterDescriptors() {
    return [{
      name: 'myParam',
      defaultValue: 0.5,
      minValue: 0,
      maxValue: 1,
      automationRate: "k-rate"
    }];
  }

  process(inputs, outputs, parameters) {
    // Get the first input and output.
    let input = inputs[0];
    let output = outputs[0];
    let myParam = parameters.myParam;

    // A simple amplifier for single input and output. Note that the
    // automationRate is "k-rate", so it will have a single value at index [0]
    // for each render quantum.
    for (let channel = 0; channel < output.length; ++channel) {
      for (let i = 0; i < output[channel].length; ++i) {
        output[channel][i] = input[channel][i] * myParam[0];
      }
    }
  }
}

1.33.4.4. `AudioParamDescriptor`

The AudioParamDescriptor dictionary is used to specify properties for an AudioParam object that is used in an AudioWorkletNode.

dictionary AudioParamDescriptor {
  required DOMString name;
  float defaultValue = 0;
  float minValue = -3.4028235e38;
  float maxValue = 3.4028235e38;
  AutomationRate automationRate = "a-rate";
};

1.33.4.4.1. Dictionary `AudioParamDescriptor` Members

automationRate, of type AutomationRate, defaulting to "a-rate": Represents the default automation rate.
defaultValue, of type float, defaulting to 0: Represents the default value of the parameter. If this value is out of the range of float data type or the range defined by minValue and maxValue, an NotSupportedError exception MUST be thrown.
maxValue, of type float, defaulting to 3.4028235e38: Represents the maximum value. An NotSupportedError exception MUST be thrown if this value is out of range of float data type or it is smaller than minValue. This value is the most positive finite single precision floating-point number.
minValue, of type float, defaulting to -3.4028235e38: Represents the minimum value. An NotSupportedError exception MUST be thrown if this value is out of range of float data type or it is greater than maxValue. This value is the most negative finite single precision floating-point number.
name, of type DOMString: Represents the name of a parameter. An NotSupportedError exception MUST be thrown when a duplicated name is found when registering the class definition.

1.33.5. The instantiation of `AudioWorkletNode` and `AudioWorkletProcessor`

When the constructor of AudioWorkletNode is invoked in the main global scope, the corresponding AudioWorkletProcessor instance is automatically created in AudioWorkletGlobalScope. After the construction, they maintain the internal reference to each other until the AudioWorkletNode instance is destroyed.

Note that the instantiation of these two objects spans the control thread and the rendering thread.

When AudioWorkletNode(context, nodeName, options) constructor is invoked, the user agent MUST perform the following steps on the control thread, where the constructor was called.

Let node be the instance being created by constructor of AudioWorkletNode or its subclass.
If nodeName does not exists as a key in the BaseAudioContext’s node name to parameter descriptor map, throw a NotSupportedError exception and abort these steps.
Let messageChannel be a new MessageChannel.
Let nodePort be the value of messageChannel’s port1 attribute.
Let processorPortOnThisSide be the value of messageChannel’s port2 attribute.
Let processorPortSerialization be StructuredSerializeWithTransfer(processorPortOnThisSide, « processorPortOnThisSide »).
Let optionsSerialization be StructuredSerialize(options).
Set node’s port to nodePort.
Let parameterDescriptors be the result of retrieval of nodeName from node name to parameter descriptor map:
1. Let audioParamMap be a new AudioParamMap object.
2. For each descriptor of parameterDescriptors:
  1. Let paramName be the value of descriptor’s name.
  2. Let audioParam be a new AudioParam instance.
  3. Append (paramName, audioParam) to audioParamMap’s entries.
3. For each paramNameInOption → value of options:
  1. If there exists an entry with name member equal to paramNameInOption inside audioParamMap, set that AudioParam's value to value.
  2. paramNameInOption will be ignored when:
    - audioParamMap does not have any entry with the same name member.
    - value is not a type of Number or out of the range specified in AudioParamDescriptor.
4. Set node’s parameters to audioParamMap.
Queue a control message to create an AudioWorkletProcessor, given nodeName, processorPortSerialization, optionsSerialization, and node.
Return node.

In order to process a control message for the construction of an AudioWorkletProcessor, given a string nodeName, a serialization record processorPortSerialization, and an AudioWorkletNode node, perform the following steps on the rendering thread. If any of these steps throws an exception (either explicitly or implicitly), abort the rest of steps and queue a task on the control thread to fire onprocessorerror event to node.

Let processorPort be StructuredDeserializeWithTransfer(processorPortSerialization, the current Realm).
Let options be StructuredDeserialize(optionsSerialization, the current Realm).
Let processorConstructor be the result of looking up nodeName on the AudioWorkletGlobalScope's node name to processor definition map.
If processorConstructor is undefined, throw a NotSupportedError DOMException.
Let processor be the result of Construct(processorConstructor, « options »).
If processor does not implement the AudioWorkletProcessor interface, throw an InvalidStateError DOMException.
Set processor’s port to processorPort.
Set processor’s node reference to node.
Set node’s processor reference to processor.
Queue a task the control thread to set the associated AudioWorkletNode's state to running, then fire a statechange event.

1.33.6. AudioWorklet Sequence of Events

The following figure illustrates an idealized sequence of events occurring relative to an AudioWorklet:

The steps depicted in the diagram are one possible sequence of events involving the creation of an AudioContext and an associated AudioWorkletGlobalScope, followed by the creation of an AudioWorkletNode and its associated AudioWorkletProcessor.

An AudioContext is created.
In the main scope, context.audioWorklet is requested to add a script module.
Since none exists yet, a new AudioWorkletGlobalScope is created in association with the context. This is the global scope in which AudioWorkletProcessor class definitions will be evaluated. (On subsequent calls, this previously created scope will be used.)
The imported script is run in the newly created global scope.
As part of running the imported script, an AudioWorkletProcessor is registered under a key ("custom" in the above diagram) within the AudioWorkletGlobalScope. This populates maps both in the global scope and in the AudioContext.
The promise for the addModule() call is resolved.
In the main scope, an AudioWorkletNode is created using the user-specified key along with a dictionary of options.
As part of the node’s creation, this key is used to look up the correct AudioWorkletProcessor subclass for instantiation.
An instance of the AudioWorkletProcessor subclass is instantiated with a structured clone of the same options dictionary. This instance is paired with the previously created AudioWorkletNode.

1.33.7. AudioWorklet Examples

1.33.7.1. The BitCrusher Node

Bitcrushing is a mechanism by which the quality of an audio stream is reduced both by quantizing the sample value (simulating a lower bit-depth), and by quantizing in time resolution (simulating a lower sample rate). This example shows how to use AudioParams (in this case, treated as a-rate) inside an AudioWorkletProcessor.

let context = new AudioContext();context.audioWorklet.addModule('bitcrusher.js').then(() => {  let context = new AudioContext();  let osc = new OscillatorNode(context);  let amp = new GainNode(context);  // Create a worklet node. 'BitCrusher' identifies the  // AudioWorkletProcessor previously registered when  // bitcrusher.js was imported. The options automatically  // initialize the correspondingly named AudioParams.  let bitcrusher = new AudioWorkletNode(context, 'BitCrusher', {    bitDepth: 8,    frequencyReduction: 0.5  });  osc.connect(bitcrusher).connect(amp).connect(context.destination);  osc.start();});

registerProcessor('BitCrusher', class extends AudioWorkletProcessor {  static get parameterDescriptors () {    return [{      name: 'bitDepth',      defaultValue: 12,      minValue: 1,      maxValue: 16    }, {      name: 'frequencyReduction',      defaultValue: 0.5,      minValue: 0,      maxValue: 1    }];  }  constructor (options) {    // We don’t need to look at options: only AudioParams are initialized,    // which were taken care of by the node.    super(options);    this._phase = 0;    this._lastSampleValue = 0;  }  process (inputs, outputs, parameters) {    let input = inputs[0];    let output = outputs[0];    let bitDepthLength = parameters.bitDepth.length;    let reductionLength = parameters.frequencyReduction.length;    if (bitDepthLength > 1) {      // General case      for (let channel = 0; channel < output.length; ++channel) {        for (let i = 0; i < output[channel].length; ++i) {          let step = Math.pow(0.5, bitDepth[i]);          // Use modulo for indexing to handle the case where          // the length of the frequencyReduction array is 1.          this._phase += frequencyReduction[i % reductionLength];          if (this._phase >= 1.0) {            this._phase -= 1.0;            this._lastSampleValue =              step * Math.floor(input[channel][i] / step + 0.5);          }          output[channel][i] = this._lastSampleValue;        }      }    } else {      // Because we know bitDepth is constant for this call,      // we can lift the computation of step outside the loop,      // saving many operations.      let step = Math.pow(0.5, bitDepth[0]);      for (let channel = 0; channel < output.length; ++channel) {        for (let i = 0; i < output[channel].length; ++i) {          this._phase += frequencyReduction[i % reductionLength];          if (this._phase >= 1.0) {            this._phase -= 1.0;            this._lastSampleValue =              step * Math.floor(input[channel][i] / step + 0.5);          }          output[channel][i] = this._lastSampleValue;        }      }    }    // No need to return a value; this node’s lifetime is dependent only on its    // input connections.  }});

Note: In the definition of AudioWorkletProcessor class, an InvalidStateError will be thrown if the author-supplied constructor uses JavaScript’s return-override feature, or does not properly call super().

1.33.7.2. VU Meter Node

This example of a simple sound level meter further illustrates how to create an AudioWorkletNode subclass that acts like a native AudioNode, accepting constructor options and encapsulating the inter-thread communication (asynchronous) between AudioWorkletNode and AudioWorkletProcessor. This node does not use any output.

class VUMeterNode extends AudioWorkletNode {  constructor (context, options) {    // Setting default values for the input, the output and the channel count.    options.numberOfInputs = 1;    options.numberOfOutputs = 0;    options.channelCount = 1;    options.updatingInterval = options.hasOwnProperty('updatingInterval')      ? options.updatingInterval      : 100;    super(context, 'VUMeter', options);    // States in AudioWorkletNode    this._updatingInterval = options.updatingInterval;    this._volume = 0;    // Handles updated values from AudioWorkletProcessor    this.port.onmessage = event => {      if (event.data.volume)        this._volume = event.data.volume;    }    this.port.start();  }  get updatingInterval() {    return this._updatingInterval;  }  set updatingInterval (intervalValue) {    this._updatingInterval = intervalValue;    this.port.postMessage({ updatingInterval: intervalValue });  }  draw () {    /* Draw the meter based on the volume value. */  }}// The application can use the node when this promise resolves.let context = new AudioContext();let importAudioWorkletNode = context.audioWorklet.addModule('vumeterprocessor.js');

registerProcessor('VUMeter', class extends AudioWorkletProcessor {  static meterSmoothingFactor = 0.9;  static meterMinimum = 0.00001;  constructor (options) {    super(options);    this._volume = 0;    this._updatingInterval = options.updatingInterval;    this._nextUpdateFrames = this.interval;    this.port.onmessage = event => {      if (event.data.updatingInterval)        this._updatingInterval = event.data.updatingInterval;    }    this.port.start();  }  get interval () {    return this._updatingInterval / 1000 * sampleRate;  }  process (inputs, outputs, parameters) {    // Note that the input will be down-mixed to mono; however, if no inputs are    // connected then zero channels will be passed in.    if (inputs[0].length > 0) {      let buffer = inputs[0][0];      let bufferLength = buffer.length;      let sum = 0, x = 0, rms = 0;      // Calculated the squared-sum.      for (let i = 0; i < bufferLength; ++i) {        x = buffer[i];        sum += x * x;      }      // Calculate the RMS level and update the volume.      rms = Math.sqrt(sum / bufferLength);      this.volume = Math.max(rms, this._volume * meterSmoothingFactor);      // Update and sync the volume property with the main thread.      this._nextUpdateFrame -= bufferLength;      if (this._nextUpdateFrame < 0) {        this._nextUpdateFrame += this.interval;        this.port.postMessage({ volume: this._volume });      }    }    // Keep on processing if the volume is above a threshold, so that    // disconnecting inputs does not immediately cause the meter to stop    // computing its smoothed value.    return this._volume >= meterMinimum;  }});

<script src="vumeternode.js"></script><script>  importAudioWorkletNode.then(() => {    let context = new AudioContext();    let oscillator = new Oscillator(context);    let vuMeterNode = new VUMeterNode(context, { updatingInterval: 50 });    oscillator.connect(vuMeterNode);    function drawMeter () {      vuMeterNode.draw();      requestAnimationFrame(drawMeter);    }    drawMeter();  });</script>

2. Processing model

2.1. Background

Real-time audio systems that require low latency are often implemented using callback functions, where the operating system calls the program back when more audio has to be computed in order for the playback to stay uninterrupted. Such callback is called on a high priority thread (often the highest priority on the system). This means that a program that deals with audio only executes code from this callback, as any buffering between a rendering thread and the callback would naturally add latency or make the system less resilient to glitches.

For this reason, the traditional way of executing asynchronous operations on the Web Platform, the event loop, does not work here, as the thread is not continuously executing. Additionally, a lot of unnecessary and potentially blocking operations are available from traditional execution contexts (Windows and Workers), which is not something that is desirable to reach an acceptable level of performance.

Additionally, the Worker model makes creating a dedicated thread necessary for a script execution context, while all AudioNodes usually share the same execution context.

Note: This section specifies how the end result should look like, not how it should be implemented. In particular, instead of using message queue, implementors can use memory that is shared between threads, as long as the memory operations are not reordered.

2.2. Control Thread and Rendering Thread

The Web Audio API MUST be implemented using a control thread, and a rendering thread.

The control thread is the thread from which the AudioContext is instantiated, and from which authors manipulate the audio graph, that is, from where the operation on a BaseAudioContext are invoked. The rendering thread is the thread on which the actual audio output is computed, in reaction to the calls from the control thread. It can be a real-time, callback-based audio thread, if computing audio for an AudioContext, or a normal thread if rendering and audio graph offline using an OfflineAudioContext.

The control thread uses a traditional event loop, as described in [HTML].

The rendering thread uses a specialized rendering loop, described in the section Rendering an audio graph

Communication from the control thread to the rendering thread is done using control message passing. Communication in the other direction is done using regular event loop tasks.

Each AudioContext has a single control message queue, that is a list of control messages that are operations running on the control thread.

Queuing a control message means adding the message to the end of the control message queue of an AudioContext.

Control messages in a control message queue are ordered by time of insertion. The oldest message is therefore the one at the front of the control message queue.

Swapping a control message queue Q_A with another control message queue Q_B means executing the following steps:

Let Q_C be a new, empty control message queue.
Move all the control messages Q_A to Q_C.
Move all the control messages Q_B to Q_A.
Move all the control messages Q_C to Q_B.

Note: For example, successfuly calling start() on an AudioBufferSourceNode source adds a control message to the control message queue of the AudioContext source.context.

2.3. Asynchronous Operations

Calling methods on AudioNodes is effectively asynchronous, and MUST to be done in two phases, a synchronous part and an asynchronous part. For each method, some part of the execution happens on the control thread (for example, throwing an exception in case of invalid parameters), and some part happens on the rendering thread (for example, changing the value of an AudioParam).

In the description of each operation on AudioNodes and AudioContexts, the synchronous section is marked with a ⌛. All the other operations are executed in parallel, as described in [HTML].

The synchronous section is executed on the control thread, and happens immediately. If it fails, the method execution is aborted, possibly throwing an exception. If it succeeds, a control message, encoding the operation to be executed on the rendering thread is enqueued on the control message queue of this rendering thread.

The synchronous and asynchronous sections order with respect to other events MUST be the same: given two operation A and B with respective synchronous and asynchronous section A_Sync and A_Async, and B_Sync and B_Async, if A happens before B, then A_Sync happens before B_Sync, and A_Async happens before B_Async. In other words, synchronous and asynchronous sections can’t be reordered.

2.4. Rendering an Audio Graph

Rendering an audio graph is done in blocks of 128 samples-frames. A block of 128 samples-frames is called a render quantum, and the render quantum size is 128.

Operations that happen atomically on a given thread can only be executed when no other atomic operation is running on another thread.

The algorithm for rendering a block of audio from an AudioContext G with a control message queue Q is comprised of multiple steps and explained in further detail in the algorithm of rendering a graph.

In practice, the AudioContext rendering thread is often running off a system level audio callback, that executes in an isochronous fashion. This callback passes in a buffer that has to be filled with the audio that will be output. The size of the buffer is often larger than a rendering quantum. In this case, multiple invocations of the rendering algorithm will be called in a rapid succession, in the same callback, before returning. After some time, the underlying audio system will call the callback again, and the algorithm will be executed again. This is an implementation detail that should not be observable, apart from the latency implications.

OfflineAudioContext will execute the algorithm continuously, until length frames have been rendered.

The system level audio callback is also queued as a task in the control message queue. The UA MUST perform the following algorithms to process render quanta to fulfill such task by filling up the requested buffer size. Along with the primary message queue, the rendering thread has another task queue for microtasks for any microtask operation such as resolution of Promises in the AudioWorkletGlobalScope.

The following step MUST be performed once before the rendering loop starts.

Set the internal slot [[current frame]] of the BaseAudioContext to 0. Also set currentTime to 0.

Let render result be false.
Process the control message queue.
1. Let Q_rendering be an empty control message queue. Atomically swap Q_rendering with the current control message queue.
2. While there are messages in Q_rendering, execute the following steps:
  1. Execute the asynchronous section of the oldest message of Q_rendering.
  2. Remove the oldest message of Q_rendering.
Process a render quantum.
1. If the rendering thread state of the AudioContext is not running, return false.
2. Order the AudioNodes of the AudioContext to be processed.
  1. Let ordered node list be an empty list of AudioNodes. It will contain an ordered list of AudioNodes when this ordering algorithm terminates.
  2. Let nodes be the set of all nodes created by this AudioContext, and still alive.
  3. Let cycle breakers be an empty set of DelayNodes. It will contain all the DelayNodes that are part of a cycle.
  4. For each AudioNode node in nodes:
    1. If node is a DelayNode that is part of a cycle, add it to cycle breakers and remove it from nodes.
  5. If nodes contains cycles, mute all the AudioNodes that are part of this cycle, and remove them from nodes.
  6. While there are unmarked AudioNodes in nodes:
    1. Choose an AudioNode node in nodes.
    2. Visit node.
    Visiting a node means performing the following steps:
    1. If node is marked, abort these steps.
    2. Mark node.
    3. Visit each AudioNode connected to the input of node.
  7. Add node to the beginning of ordered node list.
  8. Reverse the order of ordered node list.
3. For each DelayNode in a cycle, make available for reading a block of audio from the DelayNode buffer.
4. Compute the value(s) of the AudioListener's AudioParams for this block.
5. For each AudioNode, in the order determined previously:
  1. For each AudioParam of this AudioNode, execute these steps:
    1. If this AudioParam has any AudioNode connected to it, sum the buffers made available for reading by all AudioNode connected to this AudioParam, down mix the resulting buffer down to a mono channel, and call this buffer the input AudioParam buffer.
    2. Compute the value(s) of this AudioParam for this block.
    3. Queue a control message to set the [[current value]] slot of this AudioParam to the last value computed in the preceding step.
  2. If this AudioNode has any AudioNodes connected to its input, sum the buffers made available for reading by all AudioNodes connected to this AudioNode. The resulting buffer is called the input buffer. Up or down-mix it to match if number of input channels of this AudioNode.
  3. If this AudioNode is a source node, compute a block of audio, and make it available for reading.
  4. If this AudioNode is an AudioWorkletNode, invoke associated AudioWorkletProcessor's process() method to compute a block of audio with the argument of input buffer, output buffer and input AudioParam buffer.
    1. Any Promise resolved within the execution of process method will be queued into the microtask queue in the AudioWorkletGlobalScope.
  5. Else, if this AudioNode is a destination node, record the input of this AudioNode.
  6. Else, process the input buffer, and make available for reading the resulting buffer.
6. Atomically perform the following steps:
  1. Increment [[current frame]] by the render quantum size.
  2. Set currentTime to [[current frame]] divided by sampleRate.
7. Set render result to true.
Perform a microtask checkpoint.
Return render result.

Muting an AudioNode means that its output MUST be silence for the rendering of this audio block.

Making a buffer available for reading from an AudioNode means putting it in a state where other AudioNodes connected to this AudioNode can safely read from it.

Note: For example, implementations can choose to allocate a new buffer, or have a more elaborate mechanism, reusing an existing buffer that is now unused.

Recording the input of an AudioNode means copying the input data of this AudioNode for future usage.

Computing a block of audio means running the algorithm for this AudioNode to produce 128 sample-frames.

Processing an input buffer means running the algorithm for an AudioNode, using an input buffer and the value(s) of the AudioParam(s) of this AudioNode as the input for this algorithm.

3. Mixer Gain Structure

3.1. Background

One of the most important considerations when dealing with audio processing graphs is how to adjust the gain (volume) at various points. For example, in a standard mixing board model, each input bus has pre-gain, post-gain, and send-gains. Submix and master out busses also have gain control. The gain control described here can be used to implement standard mixing boards as well as other architectures.

3.2. Summing Inputs

The inputs to AudioNodes have the ability to accept connections from multiple outputs. The input then acts as a unity gain summing junction with each output signal being added with the others:

In cases where the channel layouts of the outputs do not match, a mix (usually up-mix) will occur according to the mixing rules.

No clipping is applied at the inputs or outputs of the AudioNode to allow a maximum of dynamic range within the audio graph.

3.3. Gain Control

In many scenarios, it’s important to be able to control the gain for each of the output signals. The GainNode gives this control:

A graph featuring volume control for each voice

Using these two concepts of unity gain summing junctions and GainNodes, it’s possible to construct simple or complex mixing scenarios.

3.4. Example: Mixer with Send Busses

In a routing scenario involving multiple sends and submixes, explicit control is needed over the volume or "gain" of each connection to a mixer. Such routing topologies are very common and exist in even the simplest of electronic gear sitting around in a basic recording studio.

Here’s an example with two send mixers and a main mixer. Although possible, for simplicity’s sake, pre-gain control and insert effects are not illustrated:

A graph showing a full mixer with send busses.

This diagram is using a shorthand notation where "send 1", "send 2", and "main bus" are actually inputs to AudioNodes, but here are represented as summing busses, where the intersections g2_1, g3_1, etc. represent the "gain" or volume for the given source on the given mixer. In order to expose this gain, a GainNode is used:

Here’s how the above diagram could be constructed in JavaScript:

var context = 0;var compressor = 0;var reverb = 0;var delay = 0;var s1 = 0;var s2 = 0;var source1 = 0;var source2 = 0;var g1_1 = 0;var g2_1 = 0;var g3_1 = 0;var g1_2 = 0;var g2_2 = 0;var g3_2 = 0;// Setup routing graphfunction setupRoutingGraph() {    context = new AudioContext();    compressor = context.createDynamicsCompressor();    // Send1 effect    reverb = context.createConvolver();    // Convolver impulse response may be set here or later    // Send2 effect    delay = context.createDelay();    // Connect final compressor to final destination    compressor.connect(context.destination);    // Connect sends 1 & 2 through effects to main mixer    s1 = context.createGain();    reverb.connect(s1);    s1.connect(compressor);    s2 = context.createGain();    delay.connect(s2);    s2.connect(compressor);    // Create a couple of sources    source1 = context.createBufferSource();    source2 = context.createBufferSource();    source1.buffer = manTalkingBuffer;    source2.buffer = footstepsBuffer;    // Connect source1    g1_1 = context.createGain();    g2_1 = context.createGain();    g3_1 = context.createGain();    source1.connect(g1_1);    source1.connect(g2_1);    source1.connect(g3_1);    g1_1.connect(compressor);    g2_1.connect(reverb);    g3_1.connect(delay);    // Connect source2    g1_2 = context.createGain();    g2_2 = context.createGain();    g3_2 = context.createGain();    source2.connect(g1_2);    source2.connect(g2_2);    source2.connect(g3_2);    g1_2.connect(compressor);    g2_2.connect(reverb);    g3_2.connect(delay);    // We now have explicit control over all the volumes g1_1, g2_1, ..., s1, s2    g2_1.gain.value = 0.2; // For example, set source1 reverb gain    // Because g2_1.gain is an "AudioParam",    // an automation curve could also be attached to it.    // A "mixing board" UI could be created in canvas or WebGL controlling these gains.}

4. Dynamic Lifetime

4.1. Background

This section is non-normative. Please see AudioContext lifetime and AudioNode lifetime for normative requirements.

In addition to allowing the creation of static routing configurations, it should also be possible to do custom effect routing on dynamically allocated voices which have a limited lifetime. For the purposes of this discussion, let’s call these short-lived voices "notes". Many audio applications incorporate the ideas of notes, examples being drum machines, sequencers, and 3D games with many one-shot sounds being triggered according to game play.

In a traditional software synthesizer, notes are dynamically allocated and released from a pool of available resources. The note is allocated when a MIDI note-on message is received. It is released when the note has finished playing either due to it having reached the end of its sample-data (if non-looping), it having reached a sustain phase of its envelope which is zero, or due to a MIDI note-off message putting it into the release phase of its envelope. In the MIDI note-off case, the note is not released immediately, but only when the release envelope phase has finished. At any given time, there can be a large number of notes playing but the set of notes is constantly changing as new notes are added into the routing graph, and old ones are released.

The audio system automatically deals with tearing-down the part of the routing graph for individual "note" events. A "note" is represented by an AudioBufferSourceNode, which can be directly connected to other processing nodes. When the note has finished playing, the context will automatically release the reference to the AudioBufferSourceNode, which in turn will release references to any nodes it is connected to, and so on. The nodes will automatically get disconnected from the graph and will be deleted when they have no more references. Nodes in the graph which are long-lived and shared between dynamic voices can be managed explicitly. Although it sounds complicated, this all happens automatically with no extra handling required.

4.2. Example

dynamic allocation — A graph featuring a subgraph that will be releases early.

The low-pass filter, panner, and second gain nodes are directly connected from the one-shot sound. So when it has finished playing the context will automatically release them (everything within the dotted line). If there are no longer any references to the one-shot sound and connected nodes, then they will be immediately removed from the graph and deleted. The streaming source, has a global reference and will remain connected until it is explicitly disconnected. Here’s how it might look in JavaScript:

var context = 0;var compressor = 0;var gainNode1 = 0;var streamingAudioSource = 0;// Initial setup of the "long-lived" part of the routing graphfunction setupAudioContext() {    context = new AudioContext();    compressor = context.createDynamicsCompressor();    gainNode1 = context.createGain();    // Create a streaming audio source.    var audioElement = document.getElementById('audioTagID');    streamingAudioSource = context.createMediaElementSource(audioElement);    streamingAudioSource.connect(gainNode1);    gainNode1.connect(compressor);    compressor.connect(context.destination);}// Later in response to some user action (typically mouse or key event)// a one-shot sound can be played.function playSound() {    var oneShotSound = context.createBufferSource();    oneShotSound.buffer = dogBarkingBuffer;    // Create a filter, panner, and gain node.    var lowpass = context.createBiquadFilter();    var panner = context.createPanner();    var gainNode2 = context.createGain();    // Make connections    oneShotSound.connect(lowpass);    lowpass.connect(panner);    panner.connect(gainNode2);    gainNode2.connect(compressor);    // Play 0.75 seconds from now (to play immediately pass in 0)    oneShotSound.start(context.currentTime + 0.75);}

5. Channel Up-Mixing and Down-Mixing

This section is normative.

§3 Mixer Gain Structure describes how an input to an AudioNode can be connected from one or more outputs of an AudioNode. Each of these connections from an output represents a stream with a specific non-zero number of channels. An input has mixing rules for combining the channels from all of the connections to it. As a simple example, if an input is connected from a mono output and a stereo output, then the mono connection will usually be up-mixed to stereo and summed with the stereo connection. But, of course, it’s important to define the exact mixing rules for every input to every AudioNode. The default mixing rules for all of the inputs have been chosen so that things "just work" without worrying too much about the details, especially in the very common case of mono and stereo streams. Of course, the rules can be changed for advanced use cases, especially multi-channel.

To define some terms, up-mixing refers to the process of taking a stream with a smaller number of channels and converting it to a stream with a larger number of channels. down-mixing refers to the process of taking a stream with a larger number of channels and converting it to a stream with a smaller number of channels.

An AudioNode input needs to mix all the outputs connected to this input. As part of this process it computes an internal value computedNumberOfChannels representing the actual number of channels of the input at any given time.

For each input of an AudioNode, an implementation MUST:

Compute computedNumberOfChannels.
For each connection to the input:
1. up-mix or down-mix the connection to computedNumberOfChannels according to the ChannelInterpretation value given by the node’s channelInterpretation attribute.
2. Mix it together with all of the other mixed streams (from other connections). This is a straight-forward mixing together of each of the corresponding channels from each connection.

5.1. Speaker Channel Layouts

When channelInterpretation is "speakers" then the up-mixing and down-mixing is defined for specific channel layouts.

Mono (one channel), stereo (two channels), quad (four channels), and 5.1 (six channels) MUST be supported. Other channel layouts may be supported in future version of this specification.

5.2. Channel Ordering

Channel ordering is defined by the following table. Individual multichannel formats MAY not support all intermediate channels. Implementations MUST present the channels provided in the order defined below, skipping over those channels not present.

Order	Label	Mono	Stereo	Quad	5.1
0	SPEAKER_FRONT_LEFT	0	0	0	0
1	SPEAKER_FRONT_RIGHT		1	1	1
2	SPEAKER_FRONT_CENTER				2
3	SPEAKER_LOW_FREQUENCY				3
4	SPEAKER_BACK_LEFT			2	4
5	SPEAKER_BACK_RIGHT			3	5
6	SPEAKER_FRONT_LEFT_OF_CENTER
7	SPEAKER_FRONT_RIGHT_OF_CENTER
8	SPEAKER_BACK_CENTER
9	SPEAKER_SIDE_LEFT
10	SPEAKER_SIDE_RIGHT
11	SPEAKER_TOP_CENTER
12	SPEAKER_TOP_FRONT_LEFT
13	SPEAKER_TOP_FRONT_CENTER
14	SPEAKER_TOP_FRONT_RIGHT
15	SPEAKER_TOP_BACK_LEFT
16	SPEAKER_TOP_BACK_CENTER
17	SPEAKER_TOP_BACK_RIGHT

5.3. Up Mixing Speaker Layouts

Mono up-mix:

  1 -> 2 : up-mix from mono to stereo
    output.L = input;
    output.R = input;

  1 -> 4 : up-mix from mono to quad
    output.L = input;
    output.R = input;
    output.SL = 0;
    output.SR = 0;

  1 -> 5.1 : up-mix from mono to 5.1
    output.L = 0;
    output.R = 0;
    output.C = input; // put in center channel
    output.LFE = 0;
    output.SL = 0;
    output.SR = 0;

Stereo up-mix:

  2 -> 4 : up-mix from stereo to quad
    output.L = input.L;
    output.R = input.R;
    output.SL = 0;
    output.SR = 0;

  2 -> 5.1 : up-mix from stereo to 5.1
    output.L = input.L;
    output.R = input.R;
    output.C = 0;
    output.LFE = 0;
    output.SL = 0;
    output.SR = 0;

Quad up-mix:

  4 -> 5.1 : up-mix from quad to 5.1
    output.L = input.L;
    output.R = input.R;
    output.C = 0;
    output.LFE = 0;
    output.SL = input.SL;
    output.SR = input.SR;

5.4. Down Mixing Speaker Layouts

A down-mix will be necessary, for example, if processing 5.1 source material, but playing back stereo.

Mono down-mix:

  2 -> 1 : stereo to mono
    output = 0.5 * (input.L + input.R);

  4 -> 1 : quad to mono
    output = 0.25 * (input.L + input.R + input.SL + input.SR);

  5.1 -> 1 : 5.1 to mono
    output = sqrt(0.5) * (input.L + input.R) + input.C + 0.5 * (input.SL + input.SR)

Stereo down-mix:

  4 -> 2 : quad to stereo
    output.L = 0.5 * (input.L + input.SL);
    output.R = 0.5 * (input.R + input.SR);

  5.1 -> 2 : 5.1 to stereo
    output.L = L + sqrt(0.5) * (input.C + input.SL)
    output.R = R + sqrt(0.5) * (input.C + input.SR)

Quad down-mix:

  5.1 -> 4 : 5.1 to quad
    output.L = L + sqrt(0.5) * input.C
    output.R = R + sqrt(0.5) * input.C
    output.SL = input.SL
    output.SR = input.SR

5.5. Channel Rules Examples

// Set gain node to explicit 2-channels (stereo).gain.channelCount = 2;gain.channelCountMode = "explicit";gain.channelInterpretation = "speakers";// Set "hardware output" to 4-channels for DJ-app with two stereo output busses.context.destination.channelCount = 4;context.destination.channelCountMode = "explicit";context.destination.channelInterpretation = "discrete";// Set "hardware output" to 8-channels for custom multi-channel speaker array// with custom matrix mixing.context.destination.channelCount = 8;context.destination.channelCountMode = "explicit";context.destination.channelInterpretation = "discrete";// Set "hardware output" to 5.1 to play an HTMLAudioElement.context.destination.channelCount = 6;context.destination.channelCountMode = "explicit";context.destination.channelInterpretation = "speakers";// Explicitly down-mix to mono.gain.channelCount = 1;gain.channelCountMode = "explicit";gain.channelInterpretation = "speakers";

6. Audio Signal Values

The range of all audio signals at a destination node of any audio graph is nominally [-1, 1]. The audio rendition of signal values outside this range, or of the values NaN, positive infinity or negative infinity, is undefined by this specification.

7. Spatialization/Panning

7.1. Background

A common feature requirement for modern 3D games is the ability to dynamically spatialize and move multiple audio sources in 3D space. Game audio engines such as OpenAL, FMOD, Creative’s EAX, Microsoft’s XACT Audio, etc. have this ability.

Using a PannerNode, an audio stream can be spatialized or positioned in space relative to an AudioListener. An AudioContext will contain a single AudioListener. Both panners and listeners have a position in 3D space using a right-handed cartesian coordinate system. The units used in the coordinate system are not defined, and do not need to be because the effects calculated with these coordinates are independent/invariant of any particular units such as meters or feet. PannerNode objects (representing the source stream) have an orientation vector representing in which direction the sound is projecting. Additionally, they have a sound cone representing how directional the sound is. For example, the sound could be omnidirectional, in which case it would be heard anywhere regardless of its orientation, or it can be more directional and heard only if it is facing the listener. AudioListener objects (representing a person’s ears) have forward and up vectors representing in which direction the person is facing.

During rendering, the PannerNode calculates an azimuth and elevation. These values are used internally by the implementation in order to render the spatialization effect. See the Panning Algorithm section for details of how these values are used.

7.2. Azimuth and Elevation

The following algorithm MUST be used to calculate the azimuth and elevation for the PannerNode:

// Let |context| be a BaseAudioContext and let |panner| be a// PannerNode created in |context|.// Calculate the source-listener vector.let listener = context.listener;let sourcePosition = new Vec3(panner.positionX.value, panner.positionY.value,                              panner.positionZ.value);let listenerPosition =    new Vec3(listener.positionX.value, listener.positionY.value,             listener.positionZ.value);let sourceListener = sourcePosition.diff(listenerPosition).normalize();if (sourceListener.magnitude == 0) {  // Handle degenerate case if source and listener are at the same point.  azimuth = 0;  elevation = 0;  return;}// Align axes.let listenerForward = new Vec3(listener.forwardX.value, listener.forwardY.value,                               listener.forwardZ.value);let listenerUp =    new Vec3(listener.upX.value, listener.upY.value, listener.upZ.value);let listenerRight = listenerForward.cross(listenerUp);if (listenerRight.magnitude == 0) {  // Handle the case where listener’s 'up' and 'forward' vectors are linearly  // dependent, in which case 'right' cannot be determined  azimuth = 0;  elevation = 0;  return;}// Determine a unit vector orthogonal to listener’s right, forwardlet listenerRightNorm = listenerRight.normalize();let listenerForwardNorm = listenerForward.normalize();let up = listenerRightNorm.cross(listenerForwardNorm);let upProjection = sourceListener.dot(up);let projectedSource = sourceListener.diff(up.scale(upProjection)).normalize();azimuth = 180 * Math.acos(projectedSource.dot(listenerRightNorm)) / Math.PI;// Source in front or behind the listener.let frontBack = projectedSource.dot(listenerForwardNorm);if (frontBack < 0)  azimuth = 360 - azimuth;// Make azimuth relative to "front" and not "right" listener vector.if ((azimuth >= 0) && (azimuth <= 270))  azimuth = 90 - azimuth;else  azimuth = 450 - azimuth;elevation = 90 - 180 * Math.acos(sourceListener.dot(up)) / Math.PI;if (elevation > 90)  elevation = 180 - elevation;else if (elevation < -90)  elevation = -180 - elevation;

7.3. Panning Algorithm

Mono-to-stereo and stereo-to-stereo panning MUST be supported. Mono-to-stereo processing is used when all connections to the input are mono. Otherwise stereo-to-stereo processing is used.

7.3.1. PannerNode "equalpower" Panning

This is a simple and relatively inexpensive algorithm which provides basic, but reasonable results. It is used for the for the PannerNode when the panningModel attribute is set to "equalpower", in which case the elevation value is ignored. This algorithm MUST be implemented using a-rate parameters.

For each sample to be computed by this AudioNode:

Let azimuth be the value computed in the azimuth and elevation section.

The azimuth value is first contained to be within the range [-90, 90] according to:

// First, clamp azimuth to allowed range of [-180, 180].
azimuth = max(-180, azimuth);
azimuth = min(180, azimuth);

// Then wrap to range [-90, 90].
if (azimuth < -90)
  azimuth = -180 - azimuth;
else if (azimuth > 90)
  azimuth = 180 - azimuth;

A normalized value x is calculated from azimuth for a mono input as:

x = (azimuth + 90) / 180;

Or for a stereo input as:

if (azimuth <= 0) { // -90 -> 0
  // Transform the azimuth value from [-90, 0] degrees into the range [-90, 90].
  x = (azimuth + 90) / 90;
} else { // 0 -> 90
  // Transform the azimuth value from [0, 90] degrees into the range [-90, 90].
  x = azimuth / 90;
}

Left and right gain values are calculated as:

gainL = cos(x * Math.PI / 2);
gainR = sin(x * Math.PI / 2);

For mono input, the stereo output is calculated as:

outputL = input * gainL;
outputR = input * gainR;

Else for stereo input, the output is calculated as:

if (azimuth <= 0) {
  outputL = inputL + inputR * gainL;
  outputR = inputR * gainR;
} else {
  outputL = inputL * gainL;
  outputR = inputR + inputL * gainR;
}

Apply the distance gain and cone gain where the computation of the distance is described in Distance Effects and the cone gain is described in Sound Cones:

let distance = distance();
let distanceGain = distanceModel(distance);
let totalGain = coneGain() * distanceGain();
outputL = totalGain * outputL;
outputR = totalGain * outputR;

7.3.2. PannerNode "HRTF" Panning (Stereo Only)

This requires a set of HRTF (Head-related Transfer Function) impulse responses recorded at a variety of azimuths and elevations. The implementation requires a highly optimized convolution function. It is somewhat more costly than "equalpower", but provides more perceptually spatialized sound.

A diagram showing the process of panning a source using HRTF.

7.3.3. StereoPannerNode Panning

For a StereoPannerNode, the following algorithm MUST be implemented.

For each sample to be computed by this AudioNode

Let pan be the computedValue of the pan AudioParam of this StereoPannerNode.
Clamp pan to [-1, 1].
```
pan = max(-1, pan);
pan = min(1, pan);
```

Calculate x by normalizing pan value to [0, 1]. For mono input:

x = (pan + 1) / 2;

For stereo input:

if (pan <= 0)
  x = pan + 1;
else
  x = pan;

Left and right gain values are calculated as:

gainL = cos(x * Math.PI / 2);
gainR = sin(x * Math.PI / 2);

For mono input, the stereo output is calculated as:

outputL = input * gainL;
outputR = input * gainR;

Else for stereo input, the output is calculated as:

if (pan <= 0) {
  outputL = inputL + inputR * gainL;
  outputR = inputR * gainR;
} else {
  outputL = inputL * gainL;
  outputR = inputR + inputL * gainR;
}

7.4. Distance Effects

Sounds which are closer are louder, while sounds further away are quieter. Exactly how a sound’s volume changes according to distance from the listener depends on the distanceModel attribute.

During audio rendering, a distance value will be calculated based on the panner and listener positions according to:

function distance(panner) {  let pannerPosition = new Vec3(panner.positionX.value, panner.positionY.value,                                panner.positionZ.value);  let listener = context.listener;  let listenerPosition =      new Vec3(listener.positionX.value, listener.positionY.value,               listener.positionZ.value);  return pannerPosition.diff(listenerPosition).magnitude;}

distance will then be used to calculate distanceGain which depends on the distanceModel attribute. See the DistanceModelType section for details of how this is calculated for each distance model. The value computed by the DistanceModelType equations are to be clamped to [0, 1].

As part of its processing, the PannerNode scales/multiplies the input audio signal by distanceGain to make distant sounds quieter and nearer ones louder.

7.5. Sound Cones

The listener and each sound source have an orientation vector describing which way they are facing. Each sound source’s sound projection characteristics are described by an inner and outer "cone" describing the sound intensity as a function of the source/listener angle from the source’s orientation vector. Thus, a sound source pointing directly at the listener will be louder than if it is pointed off-axis. Sound sources can also be omni-directional.

The following algorithm MUST be used to calculate the gain contribution due to the cone effect, given the source (the PannerNode) and the listener:

function coneGain() {  let sourceOrientation =      new Vec3(source.orientationX, source.orientationY, source.orientationZ);  if (sourceOrientation.magnitude == 0 ||      ((source.coneInnerAngle == 360) && (source.coneOuterAngle == 360)))    return 1; // no cone specified - unity gain  // Normalized source-listener vector  let sourcePosition = new Vec3(panner.positionX.value, panner.positionY.value,                                panner.positionZ.value);  let listenerPosition =      new Vec3(listener.positionX.value, listener.positionY.value,               listener.positionZ.value);  let sourceToListener = sourcePosition.diff(listenerPosition).normalize();  let normalizedSourceOrientation = sourceOrientation.normalize();  // Angle between the source orientation vector and the source-listener vector  let angle = 180 *              Math.acos(sourceToListener.dot(normalizedSourceOrientation)) /              Math.PI;  let absAngle = Math.abs(angle);  // Divide by 2 here since API is entire angle (not half-angle)  let absInnerAngle = Math.abs(source.coneInnerAngle) / 2;  let absOuterAngle = Math.abs(source.coneOuterAngle) / 2;  let gain = 1;  if (absAngle <= absInnerAngle) {    // No attenuation    gain = 1;  } else if (absAngle >= absOuterAngle) {    // Max attenuation    gain = source.coneOuterGain;  } else {    // Between inner and outer cones    // inner -> outer, x goes from 0 -> 1    var x = (absAngle - absInnerAngle) / (absOuterAngle - absInnerAngle);    gain = (1 - x) + source.coneOuterGain * x;  }  return gain;}

8. Performance Considerations

8.1. Latency

For web applications, the time delay between mouse and keyboard events (keydown, mousedown, etc.) and a sound being heard is important.

This time delay is called latency and is caused by several factors (input device latency, internal buffering latency, DSP processing latency, output device latency, distance of user’s ears from speakers, etc.), and is cumulative. The larger this latency is, the less satisfying the user’s experience is going to be. In the extreme, it can make musical production or game-play impossible. At moderate levels it can affect timing and give the impression of sounds lagging behind or the game being non-responsive. For musical applications the timing problems affect rhythm. For gaming, the timing problems affect precision of gameplay. For interactive applications, it generally cheapens the users experience much in the same way that very low animation frame-rates do. Depending on the application, a reasonable latency can be from as low as 3-6 milliseconds to 25-50 milliseconds.

Implementations will generally seek to minimize overall latency.

Along with minimizing overall latency, implementations will generally seek to minimize the difference between an AudioContext's currentTime and an AudioProcessingEvent's playbackTime. Deprecation of ScriptProcessorNode will make this consideration less important over time.

Additionally, some AudioNodes can add latency to some paths of the audio graph, notably:

The AudioWorkletNode can run a script that buffers internally, adding delay to the signal path.
The DelayNode, whose role is to add controlled latency time.
The BiquadFilterNode and IIRFilterNode filter design can delay incoming samples, as a natural consequence of the causal filtering process.
The ConvolverNode depending on the impulse, can delay incoming samples, as a natural result of the convolution operation.
The DynamicsCompressorNode has a look ahead algorithm that causes delay in the signal path.
The MediaStreamAudioSourceNode, MediaStreamTrackAudioSourceNode and MediaStreamAudioDestinationNode, depending on the implementation, can add buffers internally that add delays.
The ScriptProcessorNode can have buffers between the control thread and the rendering thread.
The WaveShaperNode, when oversampling, and depending on the oversampling technique, add delays to the signal path.

8.2. Audio Buffer Copying

When an acquire the content operation is performed on an AudioBuffer, the entire operation can usually be implemented without copying channel data. In particular, the last step SHOULD be performed lazily at the next getChannelData() call. That means a sequence of consecutive acquire the contents operations with no intervening getChannelData() (e.g. multiple AudioBufferSourceNodes playing the same AudioBuffer) can be implemented with no allocations or copying.

Implementations can perform an additional optimization: if getChannelData() is called on an AudioBuffer, fresh ArrayBuffers have not yet been allocated, but all invokers of previous acquire the content operations on an AudioBuffer have stopped using the AudioBuffer's data, the raw data buffers can be recycled for use with new AudioBuffers, avoiding any reallocation or copying of the channel data.

8.3. AudioParam Transitions

While no automatic smoothing is done when directly setting the value attribute of an AudioParam, for certain parameters, smooth transition are preferable to directly setting the value.

Using the setTargetAtTime() method with a low timeConstant allows authors to perform a smooth transition.

8.4. Audio Glitching

Audio glitches are caused by an interruption of the normal continuous audio stream, resulting in loud clicks and pops. It is considered to be a catastrophic failure of a multi-media system and MUST be avoided. It can be caused by problems with the threads responsible for delivering the audio stream to the hardware, such as scheduling latencies caused by threads not having the proper priority and time-constraints. It can also be caused by the audio DSP trying to do more work than is possible in real-time given the CPU’s speed.

9. Security and Privacy Considerations

The W3C TAG is developing a Self-Review Questionnaire: Security and Privacy for editors of specifications to informatively answer.

Per the Questions to Consider

Does this specification deal with personally-identifiable information?

It would be possible to perform a hearing test using Web Audio API, thus revealing the range of frequencies audible to a person (this decreases with age). It is difficult to see how this could be done without the realization and consent of the user, as it requires active particpation.
Does this specification deal with high-value data?

No. Credit card information and the like is not used in Web Audio. It is possible to use Web Audio to process or analyze voice data, which might be a privacy concern, but access to the user’s microphone is permission-based via getUserMedia.
Does this specification introduce new state for an origin that persists across browsing sessions?

No. AudioWorklet does not persist across browsing sessions.
Does this specification expose persistent, cross-origin state to the web?

Yes, the supported audio sample rate(s) and the output device channel count are exposed.
Does this specification expose any other data to an origin that it doesn’t currently have access to?

Yes. When giving various information on available AudioNodes, the Web Audio API potentially exposes information on characteristic features of the client (such as audio hardware sample-rate) to any page that makes use of the AudioNode interface. Additionally, timing information can be collected through the AnalyserNode or ScriptProcessorNode interface. The information could subsequently be used to create a fingerprint of the client.

Research by Princeton CITP’s Web Transparency and Accountability Project has shown that DynamicsCompressorNode and OscillatorNode can be used to gather entropy from a client to fingerprint a device. This is due to small, and normally inaudible, differences in DSP architecture, resampling strategies and rounding trade-offs between differing implementations. The precise comiler flags used and also the CPU architecture (Arm vs. x86) also contribute to this entropy.

In practice however, this merely allows deduction of information already readilly available by easier means (User Agent string), such as "this is Chrome running on an x86".

Fingerprinting via clock skew has been described by Steven J Murdoch and Sebastian Zander. It might be possible to determine this from getOutputTimestamp. Skew-based fingerprinting has also been demonstrated by Nakibly et. al. for HTML. The Security & Privacy appending of High Resolution Time should be consulted for further information on clock resolution and drift.

Fingerprinting via latency is also possible; it might be possible to deduce this from baseLatency and outputLatency. Mitigation strategies include adding jitter (dithering) and quantization so that the exact skew is incorrectly reported. Note however that most audio systems aim for low latency, to synchronise the audio generated by WebAudio to other audio or video sources or to visual cues (for example in a game, or an audio recording or music making environment). Excessive latency decreases usability and may be an accessibility issue.
Does this specification enable new script execution/loading mechanisms?

No. It does use the [worklets-1] script execution method, defined in that specification.
Does this specification allow an origin access to a user’s location?

No.
Does this specification allow an origin access to sensors on a user’s device?

Not directly. Currently, audio input is not specified in this document, but it will involve gaining access to the client machine’s audio input or microphone. This will require asking the user for permission in an appropriate way, probably via the getUserMedia() API.
Does this specification allow an origin access to aspects of a user’s local computing environment?

Not directly; all requested sample rates are supported, with upsampling if needed. It is possible to use Media Capture and Streams to probe for supported audio sample rates with MediaTrackSupportedConstraints. This requires explicit user consent. This does provide a small measure of fingerprinting. However, in practice most consumer and prosumer devices use one of two standardized sample rates: 44.1kHz (originally used by CD) and 48kHz (originally used by DAT). Highly resource constrained devices may support the speech-quality 11kHz sample rate, and higher-end devices often support 88.2, 96, or even the audiophile 192kHz rate.

Requiring all implementations to upsample to a single, commonly-supported rate such as 48kHz would increase CPU cost for no particular benefit, and requiring higher-end devices to use a lower rate would merely result in Web Audio being labelled as unsuitable for professional use.
Does this specification allow an origin access to other devices?

It typically does not allow access to other networked devices (an exception in a high-end recording studio might be Dante networked devices, although these typically use a separate, dedicated network). It does of necessity allow access to the users audio output device or devices, which are sometimes separate units to the computer.

For voice or sound-actuated devices, Web Audio API might be used to control other devices. In addition, if the sound-operated device is sensitive to near ultrasonic frequencies, such control might not be audible. This possibility also exists with HTML, through either the <audio> or <video> element. At common audio sampling rates, there is (by design) insufficient headroom for much ultrasonic information:

The limit of human hearing is usually stated as 20kHz. For a 44.1kHz sampling rate, the Nyquist limit is 22.05kHz. Given that a true brickwall filter cannot be physically realized, the space between 20kHz and 22.05kHz is used for a rapid rolloff filter to strongly attenuate all frequencies above Nyquist.

At 48kHz sampling rate, there is still rapid attenuation in the 20kHz to 24kHz band (but it is easier to avoid phase ripple errors in the passband).
Does this specification allow an origin some measure of control over a user agent’s native UI?

If the UI has audio components, such as a voice assistant or screenreader, Web Audio API might be used to emulate emulate aspects of the native UI to make an attack seem more like a local system event. This possibility also exists with HTML, through the <audio> element.
Does this specification expose temporary identifiers to the web?

No.
Does this specification distinguish between behavior in first-party and third-party contexts?

No.
How should this specification work in the context of a user agent’s "incognito" mode?

No differently.
Does this specification persist data to a user’s local device?

No.
Does this specification have a "Security Considerations" and "Privacy Considerations" section?

Yes (you are reading it).
Does this specification allow downgrading default security characteristics?

No.

10. Requirements and Use Cases

Please see [webaudio-usecases].

11. Common Definitions for Specification Code

This section describes common functions and classes employed by JavaScript code used within this specification.

// Three dimensional vector class.class Vec3 {  // Construct from 3 coordinates.  constructor(x, y, z) {    this.x = x;    this.y = y;    this.z = z;  }  // Dot product with another vector.  dot(v) {    return (this.x * v.x) + (this.y * v.y) + (this.z * v.z);  }  // Cross product with another vector.  cross(v) {    return new Vec3((this.y * v.z) - (this.z * v.y),      (this.z * v.x) - (this.x * v.z),      (this.x * v.y) - (this.y * v.x));  }  // Difference with another vector.  diff(v) {    return new Vec3(this.x - v.x, this.y - v.y, this.z - v.z);  }  // Get the magnitude of this vector.  get magnitude() {    return Math.sqrt(dot(this));  }  // Get a copy of this vector multiplied by a scalar.  scale(s) {    return new Vec3(this.x * s, this.y * s, this.z * s);  }  // Get a normalized copy of this vector.  normalize() {    let m = magnitude;    if (m == 0) {      return new Vec3(0, 0, 0);    }    return scale(1 / m);  }}

12. Change Log

12.1. Since Working Draft of 19 June 2018

Minor editorial clarifications
Update implementation-report.html
Widen the valid range of detune values so that any value that doesn’t cause 2^(d/1200) to overflow is valid.
PannerNode constructor throws errors
Rephrase algorithm for setting buffer and curve
Refine startRendering algorithm.
Make "queue a task" link to the HTML spec
Specify more precisely, events overlapping with SetValueCurveAtTime
Add implementation report to gh-pages
Honor the given value in outputChannelCount
Initialize bufferDuration outside of process() in ABSN algorithm.
Rework definition of ABSN output behavior to account for playbackRate’s interaction with the start(…duration) argument.
Add mention of video element in ultrasonic attack surface.

12.2. Since Working Draft of 08 December 2015

Add AudioWorklet and related interfaces to support custom nodes. This replaces ScriptProcessorNode, which is now deprecated.
Explicitly say what the channel count, mode, and interpretation values are for all source nodes.
Specify the behavior of Web Audio when a document is unloaded.
Merge the proposed SpatialListener interface into AudioListener.
Rework and clean up algorithms for panning and spatialization and define "magic functions".
Clarify that AudioBufferSourceNode looping is limited by duration argument to start().
Add constructors with options dictionaries for all node types.
Clarify parameter automation method behavior and equations. Handle cases where automation methods may interact with each other.
Support latency hints and arbitrary sample rates in AudioContext constructor.
Clear up ambiguities in definitions of start() and stop() for scheduled sources.
Remove automatic dezippering from AudioParam value setters which now equate to setValueAtTime().
Specify normative behavior of DynamicsCompressorNode.
Specify that AudioParam.value returns the most recent computed value.
Permit AudioBufferSourceNode to specify sub-sample start, duration, loopStart and loopEnd. Respecify algorithms to say exactly how looping works in all scenarios, including dynamic and negative playback rates.
Harmonized behavior of IIRFilterNode with BiquadFilterNode.
Add diagram describing mono-input-to-matrixed-stereo case.
Prevent connecting an AudioNode to an AudioParam of a different AudioContext.
Added Audioparam cancelAndHoldAtTime
Clarify behaviour of AudioParam.cancelScheduledValues().
Add playing reference to MediaElementAudioSourceNodes and MediaStreamAudioSourceNodes.
Refactor BaseAudioContext interface out of AudioContext, OfflineAudioContext.
OfflineAudioContext inherits from BaseAudioContext, not AudioContext.
"StereoPanner" replaced with the correct "StereoPannerNode".
Support chaining on AudioNode.connect() and AudioParam automation methods.
Specify behavior of events following SetTarget events.
Reinstate channelCount declaration for AnalyserNode.
Specify exponential ramp behavior when previous value is 0.
Specify behavior of setValueCurveAtTime parameters.
Add spatialListener attribute to AudioContext.
Remove section titled "Doppler Shift".
Added a list of nodes and reason why they can add latency, in an informative section.
Speced nominal ranges, nyquist, and behavior when outside the range.
Spec the processing model for the Web Audio API.
Merge the SpatialPannerNode into the PannerNode, undeprecating the PannerNode.
Merge the SpatialListener into the AudioListener, undeprecating the AudioListener.
Added latencyHint(s).
Move the constructor from BaseAudioContext to AudioContext where it belongs; BaseAudioContext is not constructible.
Specified the Behavior of automations and nominal ranges.
The playbackRate is widened to +/- infinity.
setValueCurveAtTime is modified so that an implicit call to setValueAtTime is made at the end of the curve duration.
Make setting the value attribute of an AudioParam strictly equivalent of calling setValueAtTime with AudioContext.currentTime.
Add new sections for AudioContextOptions and AudioTimestamp.
Add constructor for all nodes.
Define ConstantSourceNode.
Make the WaveShaperNode have a tail time, depending on the oversampling level.
Allow collecting MediaStreamAudioSourceNode or MediaElementAudioSourceNode when they won’t play ever again.
Add a concept of 'allowed to start' and use it when creating an AudioContext and resuming it from resume() (closes #836).
Add AudioScheduledSourceNode base class for source nodes.
Mark all AudioParams as being k-rate.

13. Acknowledgements

This specification is the collective work of the W3C Audio Working Group.

Members and former members of the Working Group and contributors to the specification are (at the time of writing, and by alphabetical order):
Adenot, Paul (Mozilla Foundation) - Specification Co-editor; Akhgari, Ehsan (Mozilla Foundation); Berkovitz, Joe (Invited Expert, affiliated with Noteflight/Hal Leonard, WG Chair from September 2013 to December 2017); Bossart, Pierre (Intel Corporation); Buffa, Michel (NSAU;) Caceres, Marcos (Invited Expert); Cardoso, Gabriel (INRIA); Carlson, Eric (Apple, Inc.); Chen, Bin (Baidu, Inc.); Choi, Hongchan (Google, Inc.); Geelnard, Marcus (Opera Software); Gehring, Todd (Dolby Laboratories); Goode, Adam (Google, Inc.); Gregan, Matthew (Mozilla Foundation); Hofmann, Bill (Dolby Laboratories); Jägenstedt, Philip (Opera Software); Kalliokoski, Jussi (Invited Expert); Kawai, Ryoya (AMEI); Lilley, Chris (W3C Staff); Lowis, Chris (Invited Expert. WG co-chair from December 2012 to September 2013, affiliated with British Broadcasting Corporation); MacDonald, Alistair (W3C Invited Experts) — WG co-chair from March 2011 to July 2012; Mandyam, Giridhar (Qualcomm Innovation Center, Inc); Michel, Thierry (W3C/ERCIM); Noble, Jer (Apple, Inc.); O’Callahan, Robert(Mozilla Foundation); Onumonu, Anthony (British Broadcasting Corporation); Paradis, Matthew (British Broadcasting Corporation) - WG Chair from September 2013 to present; Raman, T.V. (Google, Inc.); Rogers, Chris (Google, Inc.) – Specification Editor until August 2013; Schepers, Doug (W3C/MIT); Shires, Glen (Google, Inc.); Smith, Michael (W3C/Keio); Thereaux, Olivier (British Broadcasting Corporation); Toy, Raymond (Google, Inc.) - WG Chair from December 2017 - Present, Specification Co-editor; Verdie, Jean-Charles (MStar Semiconductor, Inc.); Wei, James (Intel Corporation); Weitnauer, Michael (IRT); Wilson, Chris (Google,Inc.) - Specification Co-editor; ZERGAOUI, Mohamed (INNOVIMAX)

Web Audio API

W3C Candidate Recommendation, 18 September 2018

Abstract

Status of this document

Introduction

Features

Modular Routing

API Overview

1. The Audio API

1.1. The BaseAudioContext Interface

1.1.1. Attributes

1.1.2. Methods

1.1.3. Callback DecodeSuccessCallback() Parameters

1.1.4. Callback DecodeErrorCallback() Parameters

1.1.5. Lifetime

1.1.6. Lack of Introspection or Serialization Primitives

1.1.7. System Resources Associated with BaseAudioContext Subclasses

1.2. The AudioContext Interface

1.2.1. Constructors

1.2.2. Attributes

1.2.3. Methods

1.2.4. AudioContextOptions

1.2.4.1. Dictionary AudioContextOptions Members

1.2.5. AudioTimestamp

1.2.5.1. Dictionary AudioTimestamp Members

1.3. The OfflineAudioContext Interface

1.3.1. Constructors

1.3.2. Attributes

1.3.3. Methods

1.3.4. OfflineAudioContextOptions

1.3.4.1. Dictionary OfflineAudioContextOptions Members

1.3.5. The OfflineAudioCompletionEvent Interface

1.3.5.1. Attributes

1.3.5.2. OfflineAudioCompletionEventInit

1.3.5.2.1. Dictionary OfflineAudioCompletionEventInit Members

1.4. The AudioBuffer Interface

1.4.1. Constructors

1.4.2. Attributes

1.4.3. Methods

1.4.4. AudioBufferOptions

1.4.4.1. Dictionary AudioBufferOptions Members

1.5. The AudioNode Interface

1.5.1. AudioNode Creation

1.5.2. Attributes

1.5.3. Methods

1.5.4. AudioNodeOptions

1.5.4.1. Dictionary AudioNodeOptions Members

1.5.5. Lifetime

1.6. The AudioParam Interface

1.6.1. Attributes

1.6.2. Methods

1.6.3. Computation of Value

1.6.4. AudioParam Automation Example

1.7. The AudioScheduledSourceNode Interface

1.7.1. Attributes

1.7.2. Methods

1.8. The AnalyserNode Interface

1.8.1. Constructors

1.8.2. Attributes

1.8.3. Methods

1.8.4. AnalyserOptions

1.8.4.1. Dictionary AnalyserOptions Members

1.8.5. Time-Domain Down-Mixing

1.8.6. FFT Windowing and Smoothing over Time

1.9. The AudioBufferSourceNode Interface

1.9.1. Constructors

1.9.2. Attributes

1.9.3. Methods

1.9.4. AudioBufferSourceOptions

1.9.4.1. Dictionary AudioBufferSourceOptions Members

1.9.5. Looping

1.9.6. Playback of AudioBuffer Contents

1.10. The AudioDestinationNode Interface

1.10.1. Attributes

1.11. The AudioListener Interface

1.11.1. Attributes

1.11.2. Methods

1.12. The AudioProcessingEvent Interface - DEPRECATED

1.12.1. Attributes

1.12.2. AudioProcessingEventInit

1.1. The `BaseAudioContext` Interface

1.1.3. Callback `DecodeSuccessCallback()` Parameters

1.1.4. Callback `DecodeErrorCallback()` Parameters

1.1.7. System Resources Associated with `BaseAudioContext` Subclasses

1.2. The `AudioContext` Interface

1.2.4. `AudioContextOptions`

1.2.4.1. Dictionary `AudioContextOptions` Members

1.2.5. `AudioTimestamp`

1.2.5.1. Dictionary `AudioTimestamp` Members

1.3. The `OfflineAudioContext` Interface

1.3.4. `OfflineAudioContextOptions`

1.3.4.1. Dictionary `OfflineAudioContextOptions` Members

1.3.5. The `OfflineAudioCompletionEvent` Interface

1.3.5.2. `OfflineAudioCompletionEventInit`

1.3.5.2.1. Dictionary `OfflineAudioCompletionEventInit` Members

1.4. The `AudioBuffer` Interface

1.4.4. `AudioBufferOptions`

1.4.4.1. Dictionary `AudioBufferOptions` Members

1.5. The `AudioNode` Interface

1.5.4. `AudioNodeOptions`

1.5.4.1. Dictionary `AudioNodeOptions` Members

1.6. The `AudioParam` Interface

1.6.4. `AudioParam` Automation Example

1.7. The `AudioScheduledSourceNode` Interface

1.8. The `AnalyserNode` Interface

1.8.4. `AnalyserOptions`

1.8.4.1. Dictionary `AnalyserOptions` Members

1.9. The `AudioBufferSourceNode` Interface

1.9.4. `AudioBufferSourceOptions`

1.9.4.1. Dictionary `AudioBufferSourceOptions` Members

1.10. The `AudioDestinationNode` Interface

1.11. The `AudioListener` Interface

1.12. The `AudioProcessingEvent` Interface - DEPRECATED

1.12.2. `AudioProcessingEventInit`

1.12.2.1. Dictionary `AudioProcessingEventInit` Members

1.13. The `BiquadFilterNode` Interface

1.13.4. `BiquadFilterOptions`

1.13.4.1. Dictionary `BiquadFilterOptions` Members

1.14. The `ChannelMergerNode` Interface

1.14.2. `ChannelMergerOptions`

1.14.2.1. Dictionary `ChannelMergerOptions` Members

1.15. The `ChannelSplitterNode` Interface

1.15.2. `ChannelSplitterOptions`

1.15.2.1. Dictionary `ChannelSplitterOptions` Members

1.16. The `ConstantSourceNode` Interface

1.16.3. `ConstantSourceOptions`

1.16.3.1. Dictionary `ConstantSourceOptions` Members

1.17. The `ConvolverNode` Interface

1.17.3. `ConvolverOptions`

1.17.3.1. Dictionary `ConvolverOptions` Members

1.18. The `DelayNode` Interface

1.18.3. `DelayOptions`

1.18.3.1. Dictionary `DelayOptions` Members

1.19. The `DynamicsCompressorNode` Interface

1.19.3. `DynamicsCompressorOptions`

1.19.3.1. Dictionary `DynamicsCompressorOptions` Members

1.20. The `GainNode` Interface

1.20.3. `GainOptions`

1.20.3.1. Dictionary `GainOptions` Members

1.21. The `IIRFilterNode` Interface

1.21.3.1. Dictionary `IIRFilterOptions` Members

1.22. The `MediaElementAudioSourceNode` Interface

1.22.3. `MediaElementAudioSourceOptions`

1.22.3.1. Dictionary `MediaElementAudioSourceOptions` Members

1.23. The `MediaStreamAudioDestinationNode` Interface

1.24. The `MediaStreamAudioSourceNode` Interface

1.24.3. `MediaStreamAudioSourceOptions`

1.24.3.1. Dictionary `MediaStreamAudioSourceOptions` Members

1.25. The `MediaStreamTrackAudioSourceNode` Interface

1.25.2. `MediaStreamTrackAudioSourceOptions`

1.25.2.1. Dictionary `MediaStreamTrackAudioSourceOptions` Members

1.26. The `OscillatorNode` Interface