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.
High dynamic range, using 32-bit 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-domain 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:

const context = new AudioContext();

function playSound() {
  const 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.

let context;let compressor;let reverb;let source1, source2, source3;let lowpassFilter;let waveShaper;let panner;let dry1, dry2, dry3;let wet1, wet2, wet3;let mainDry;let mainWet;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 main wet and dry.  mainDry = context.createGain();  mainWet = context.createGain();  // Connect final compressor to final destination.  compressor.connect(context.destination);  // Connect main dry and wet to compressor.  mainDry.connect(compressor);  mainWet.connect(compressor);  // Connect reverb to main wet.  reverb.connect(mainWet);  // 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(mainDry);  wet1.connect(reverb);  // Connect source2  dry2 = context.createGain();  wet2 = context.createGain();  source2.connect(waveShaper);  waveShaper.connect(dry2);  waveShaper.connect(wet2);  dry2.connect(mainDry);  wet2.connect(reverb);  // Connect source3  dry3 = context.createGain();  wet3 = context.createGain();  source3.connect(panner);  panner.connect(dry3);  panner.connect(wet3);  dry3.connect(mainDry);  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() {  const context = new AudioContext();  // Create the low frequency oscillator that supplies the modulation signal  const lfo = context.createOscillator();  lfo.frequency.value = 1.0;  // Create the high frequency oscillator to be modulated  const hfo = context.createOscillator();  hfo.frequency.value = 440.0;  // Create a gain node whose gain determines the amplitude of the modulation signal  const 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.
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

BaseAudioContext

Firefox53+SafariNoneChrome56+

Opera43+Edge79+

Edge (Legacy)NoneIENone

Firefox for Android53+iOS SafariNoneChrome for Android56+Android WebView56+Samsung Internet6.0+Opera Mobile43+

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).

BaseAudioContext are created with an internal slot [[pending promises]] that is an initially empty ordered list of promises.

Each BaseAudioContext has a unique media element event task source. Additionally, a BaseAudioContext has two private slots [[rendering thread state]] and [[control thread state]] that take values from AudioContextState, and that are both initialy set to "suspended".

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 = undefined (DOMException error);

callback DecodeSuccessCallback = undefined (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);
};

1.1.1. Attributes

BaseAudioContext/audioWorklet Firefox76+SafariNoneChrome66+ OperaYesEdge79+ Edge (Legacy)NoneIENone Firefox for Android79+iOS SafariNoneChrome for Android66+Android WebView66+Samsung Internet9.0+Opera MobileYes 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 [HTML] and AudioWorklet.

BaseAudioContext/currentTime In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

BaseAudioContext/destination In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

BaseAudioContext/listener In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ listener, of type AudioListener, readonly

An AudioListener which is used for 3D spatialization.

BaseAudioContext/onstatechange In all current engines. Firefox40+Safari9+Chrome43+ OperaYesEdge79+ Edge (Legacy)14+IENone Firefox for Android40+iOS Safari9+Chrome for AndroidYesAndroid WebViewYesSamsung InternetYesOpera MobileYes 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.

BaseAudioContext/sampleRate In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

BaseAudioContext/state In all current engines. Firefox40+Safari9+Chrome43+ OperaYesEdge79+ Edge (Legacy)14+IENone Firefox for Android40+iOS Safari9+Chrome for AndroidYesAndroid WebViewYesSamsung InternetYesOpera MobileYes state, of type AudioContextState, readonly

Describes the current state of the BaseAudioContext. Getting this attribute returns the contents of the [[control thread state]] slot.

1.1.2. Methods

BaseAudioContext/createAnalyser In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ createAnalyser()

Factory method for an AnalyserNode.

No parameters.

Return type: AnalyserNode

BaseAudioContext/createBiquadFilter In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

BaseAudioContext/createBuffer In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

BaseAudioContext/createBufferSource In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ createBufferSource()

Factory method for a AudioBufferSourceNode.

No parameters.

Return type: AudioBufferSourceNode

BaseAudioContext/createChannelMerger In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

BaseAudioContext/createChannelSplitter In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

BaseAudioContext/createConstantSource Firefox52+SafariNoneChrome56+ Opera43+Edge79+ Edge (Legacy)NoneIENone Firefox for Android52+iOS SafariNoneChrome for Android56+Android WebView56+Samsung Internet6.0+Opera Mobile43+ createConstantSource()

Factory method for a ConstantSourceNode.

No parameters.

Return type: ConstantSourceNode

BaseAudioContext/createConvolver In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ createConvolver()

Factory method for a ConvolverNode.

No parameters.

Return type: ConvolverNode

BaseAudioContext/createDelay In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

BaseAudioContext/createDynamicsCompressor In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ createDynamicsCompressor()

Factory method for a DynamicsCompressorNode.

No parameters.

Return type: DynamicsCompressorNode

BaseAudioContext/createGain In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ createGain()

Factory method for GainNode.

No parameters.

Return type: GainNode

BaseAudioContext/createIIRFilter Firefox50+SafariNoneChrome49+ OperaYesEdge79+ Edge (Legacy)14+IENone Firefox for Android50+iOS SafariNoneChrome for Android49+Android WebView49+Samsung Internet5.0+Opera MobileYes 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

BaseAudioContext/createOscillator In all current engines. Firefox25+Safari6+Chrome20+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ createOscillator()

Factory method for an OscillatorNode.

No parameters.

Return type: OscillatorNode

BaseAudioContext/createPanner In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ createPanner()

Factory method for a PannerNode.

No parameters.

Return type: PannerNode

BaseAudioContext/createPeriodicWave In all current engines. Firefox25+Safari8+Chrome59+ Opera17+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari8+Chrome for Android59+Android WebView59+Samsung Internet7.0+Opera Mobile18+ 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. The default value is 2. 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. The default value is 2. Values of up to 32 must be supported. A `NotSupportedError` must be thrown if the number of channels is not supported.

Return type: ScriptProcessorNode

BaseAudioContext/createStereoPanner Firefox37+SafariNoneChrome42+ OperaNoneEdge79+ Edge (Legacy)12+IENone Firefox for Android37+iOS SafariNoneChrome for AndroidYesAndroid WebViewYesSamsung InternetYesOpera MobileNone createStereoPanner()

Factory method for a StereoPannerNode.

No parameters.

Return type: StereoPannerNode

BaseAudioContext/createWaveShaper In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ createWaveShaper()

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

No parameters.

Return type: WaveShaperNode

BaseAudioContext/decodeAudioData In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

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

If this's relevant global object's associated Document is not fully active then return a promise rejected with "InvalidStateError" DOMException.
Let promise be a new Promise.
If the operation IsDetachedBuffer (described in [ECMASCRIPT]) on audioData is false, execute the following steps:
1. Append promise to [[pending promises]].
2. Detach the audioData ArrayBuffer. This operation is described in [ECMASCRIPT]. If this operations throws, jump to the step 3.
3. Queue a decoding operation to be performed on another thread.
Else, execute the following error steps:
1. Let error be a DataCloneError.
2. Reject promise with error, and remove it from [[pending promises]].
3. Queue a media element 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.

Let can decode be a boolean flag, initially set to true.
Attempt to determine the MIME type of audioData, using MIME Sniffing §6.2 Matching an audio or video type pattern. If the audio or video type pattern matching algorithm returns undefined, set can decode to false.
If can decode is true, attempt to decode the encoded audioData into linear PCM. In case of failure, set can decode to false.
If can decode is false, queue a media element task to execute the following steps:
1. Let error be a DOMException whose name is EncodingError.
  1. Reject promise with error, and remove it from [[pending promises]].
2. 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 BaseAudioContext if it is different from the sample-rate of audioData.
2. queue a media element task to execute the following steps:
  1. Let buffer be an AudioBuffer containing the final result (after possibly performing sample-rate conversion).
  2. Resolve promise with buffer.
  3. 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>

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

AudioContext

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Opera22+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariNoneChrome for Android35+Android WebView37+Samsung Internet3.0+Opera Mobile22+

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.

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

An AudioContext is said to be allowed to start if the user agent allows the context state to transition from "suspended" to "running". A user agent may disallow this initial transition, and to allow it only when the AudioContext's relevant global object has sticky activation.

AudioContext has an internal slot:

[[suspended by user]]: A boolean flag representing whether the context is suspended by user code. The initial value is false.

1.2.1. Constructors

AudioContext/AudioContext Firefox25+SafariNoneChrome35+ Opera22+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariNoneChrome for Android35+Android WebView37+Samsung Internet3.0+Opera Mobile22+ AudioContext(contextOptions)

If the current settings object’s responsible document is NOT fully active, throw an InvalidStateError and abort these steps.

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 [[pending resume promises]] be a slot on this AudioContext, that is an initially 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 context is allowed to start, send a control message to start processing.
Return this AudioContext object.

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

Attempt to acquire system resources. In case of failure, abort the following steps.
Set the [[rendering thread state]] to running on the AudioContext.
queue a media element task to execute the following steps:
1. Set the state attribute of the AudioContext to "running".
2. queue a media element task to fire an 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.constructor(contextOptions) method.
Parameter	Type	Nullable	Optional	Description
`contextOptions`	AudioContextOptions	✘	✔	User-specified options controlling how the `AudioContext` should be constructed.

1.2.2. Attributes

AudioContext/baseLatency Firefox70+SafariNoneChrome58+ Opera45+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android58+Android WebView58+Samsung Internet7.0+Opera Mobile43+ 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.

AudioContext/outputLatency In only one current engine. Firefox70+SafariNoneChromeNone OperaNoneEdgeNone Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for AndroidNoneAndroid WebViewNoneSamsung InternetNoneOpera MobileNone 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

AudioContext/close In all current engines. Firefox40+Safari9+Chrome42+ OperaYesEdge79+ Edge (Legacy)14+IENone Firefox for Android40+iOS Safari9+Chrome for Android43+Android WebView43+Samsung Internet4.0+Opera MobileYes close()

Closes the AudioContext, releasing the system resources being used. 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:

If this's relevant global object's associated Document is not fully active then return a promise rejected with "InvalidStateError" DOMException.
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 closed.
Queue a control message to close 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.

This will stop rendering.
If this control message is being run in a reaction to the document being unloaded, abort this algorithm.

There is no need to notify the control thread in this case.
queue a media element task to execute the following 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 media element task to fire an 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<undefined>

AudioContext/createMediaElementSource In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AudioContext/createMediaStreamDestination In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)NoneIENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ createMediaStreamDestination()

Creates a MediaStreamAudioDestinationNode

No parameters.

Return type: MediaStreamAudioDestinationNode

AudioContext/createMediaStreamSource In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AudioContext/createMediaStreamTrackSource In only one current engine. Firefox68+SafariNoneChromeNone OperaNoneEdgeNone Edge (Legacy)NoneIENone Firefox for Android68+iOS SafariNoneChrome for AndroidNoneAndroid WebViewNoneSamsung InternetNoneOpera MobileNone 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

AudioContext/getOutputTimestamp Firefox70+SafariNoneChrome57+ Opera44+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android57+Android WebView57+Samsung Internet7.0+Opera Mobile43+ getOutputTimestamp()

Returns a new AudioTimestamp instance containing two related audio stream position values for the context: 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-3]).

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) {
  const timestamp = context.getOutputTimestamp();
  const 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

AudioContext/resume In all current engines. Firefox40+Safari9+Chrome41+ OperaYesEdge79+ Edge (Legacy)14+IENone Firefox for AndroidYesiOS Safari9+Chrome for Android41+Android WebView41+Samsung Internet4.0+Opera MobileYes resume()

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

When resume is called, execute these steps:

If this's relevant global object's associated Document is not fully active then return a promise rejected with "InvalidStateError" DOMException.
Let promise be a new Promise.
If the [[control thread state]] on the AudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
Set [[suspended by user]] to false.
If the context is not allowed to start, append promise to [[pending promises]] and [[pending resume promises]] and abort these steps, returning promise.
Set the [[control thread state]] on the AudioContext to running.
Queue a control message to resume the AudioContext.
Return promise.

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

Attempt to acquire system resources.
Set the [[rendering thread state]] on the AudioContext to running.
Start rendering the audio graph.
In case of failure, queue a media element task to execute the following steps:
1. Reject all promises from [[pending resume promises]] in order, then clear [[pending resume promises]].
2. Additionally, remove those promises from [[pending promises]].
queue a media element task to execute the following steps:
1. Resolve all promises from [[pending resume promises]] in order.
2. Clear [[pending resume promises]]. Additionally, remove those promises from [[pending promises]].
3. Resolve promise.
4. If the state attribute of the AudioContext is not already "running":
  1. Set the state attribute of the AudioContext to "running".
  2. queue a media element task to fire an event named statechange at the AudioContext.

No parameters.

Return type: Promise<undefined>

AudioContext/suspend In all current engines. Firefox40+Safari9+Chrome43+ OperaYesEdge79+ Edge (Legacy)14+IENone Firefox for Android40+iOS Safari9+Chrome for Android43+Android WebView43+Samsung Internet4.0+Opera MobileYes 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:

If this's relevant global object's associated Document is not fully active then return a promise rejected with "InvalidStateError" DOMException.
Let promise be a new Promise.
If the [[control thread state]] on the AudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
Append promise to [[pending promises]].
Set [[suspended by user]] to true.
Set the [[control thread state]] 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 media element task to execute the following 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 media element task to fire an 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<undefined>

1.2.4. `AudioContextOptions`

AudioContextOptions

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Opera?Edge79+

Edge (Legacy)NoneIENone

Firefox for Android61+iOS SafariNoneChrome for Android60+Android WebView60+Samsung Internet8.0+Opera Mobile?

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

AudioContextOptions/latencyHint Firefox61+SafariNoneChrome60+ Opera?Edge79+ Edge (Legacy)NoneIENone Firefox for Android61+iOS SafariNoneChrome for Android60+Android WebView60+Samsung Internet8.0+Opera Mobile? 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.

AudioContextOptions/sampleRate Firefox61+SafariNoneChrome74+ OperaNoneEdge79+ Edge (Legacy)NoneIENone Firefox for Android61+iOS SafariNoneChrome for Android74+Android WebView74+Samsung Internet11.0+Opera Mobile? 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-3]).

1.3. The `OfflineAudioContext` Interface

OfflineAudioContext

Firefox25+SafariNoneChrome35+

Opera22+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariNoneChrome for Android35+Android WebView4.4.3+Samsung Internet3.0+Opera Mobile22+

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]
interface OfflineAudioContext : BaseAudioContext {
  constructor(OfflineAudioContextOptions contextOptions);
  constructor(unsigned long numberOfChannels, unsigned long length, float sampleRate);
  Promise<AudioBuffer> startRendering();
  Promise<undefined> resume();
  Promise<undefined> suspend(double suspendTime);
  readonly attribute unsigned long length;
  attribute EventHandler oncomplete;
};

1.3.1. Constructors

OfflineAudioContext/OfflineAudioContext Firefox53+SafariNoneChrome35+ Opera22+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariNoneChrome for Android35+Android WebView4.4.3+Samsung Internet3.0+Opera Mobile22+ OfflineAudioContext(contextOptions)

If the current settings object’s responsible document is NOT fully active, throw an InvalidStateError and abort these steps.

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.constructor(contextOptions) method.
Parameter	Type	Nullable	Optional	Description
`contextOptions`				The initial parameters needed to construct this context.

OfflineAudioContext(numberOfChannels, length, sampleRate)

The OfflineAudioContext can be 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.constructor(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

OfflineAudioContext/length FirefoxYesSafariNoneChrome51+ Opera38+Edge79+ Edge (Legacy)14+IENone Firefox for AndroidYesiOS SafariNoneChrome for Android51+Android WebView51+Samsung Internet5.0+Opera Mobile41+ 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.
OfflineAudioContext/oncomplete In all current engines. Firefox25+Safari6+Chrome25+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari?Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ oncomplete, of type EventHandler: An EventHandler of type OfflineAudioCompletionEvent. It is the last event fired on an OfflineAudioContext.

1.3.3. Methods

OfflineAudioContext/startRendering In all current engines. Firefox25+Safari6+Chrome25+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari?Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ startRendering()

Given the current connections and scheduled changes, starts rendering audio.

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 this's relevant global object's associated Document is not fully active then return a promise rejected with "InvalidStateError" DOMException.
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.
Append promise to [[pending promises]].
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 rendering if necessary.
If a suspended context is resumed, continue to render the buffer.
Once the rendering is complete, queue a media element task to execute the following steps:
1. Resolve the promise created by startRendering() with [[rendered buffer]].
2. queue a media element task to fire an event named complete using an instance of OfflineAudioCompletionEvent whose renderedBuffer property is set to [[rendered buffer]].

No parameters.

Return type: Promise<AudioBuffer>

OfflineAudioContext/resume In only one current engine. FirefoxNoneSafariNoneChrome49+ Opera36+Edge79+ Edge (Legacy)18IENone Firefox for AndroidNoneiOS SafariNoneChrome for Android49+Android WebView49+Samsung Internet5.0+Opera Mobile36+ resume()

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

When resume is called, execute these steps:

If this's relevant global object's associated Document is not fully active then return a promise rejected with "InvalidStateError" DOMException.
Let promise be a new Promise.
Abort these steps and reject promise with InvalidStateError when any of following conditions is true:
- The [[control thread state]] on the OfflineAudioContext is closed.
- The [[rendering started]] slot on the OfflineAudioContext is false.
Set the [[control thread state]] flag on the OfflineAudioContext to running.
Queue a control message to resume the OfflineAudioContext.
Return promise.

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

Set the [[rendering thread state]] on the OfflineAudioContext to running.
Start rendering the audio graph.
In case of failure, queue a media element task to reject promise and abort the remaining steps.
queue a media element task to execute the following steps:
1. Resolve promise.
2. If the state attribute of the OfflineAudioContext is not already "running":
  1. Set the state attribute of the OfflineAudioContext to "running".
  2. queue a media element task to fire an event named statechange at the OfflineAudioContext.

No parameters.

Return type: Promise<undefined>

OfflineAudioContext/suspend In only one current engine. FirefoxNoneSafariNoneChrome49+ Opera36+Edge79+ Edge (Legacy)18IENone Firefox for AndroidNoneiOS SafariNoneChrome for Android49+Android WebView49+Samsung Internet5.0+Opera Mobile36+ 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 up 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 up 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<undefined>

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

OfflineAudioCompletionEvent

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

OfflineAudioContext/complete_event

In all current engines.

Firefox25+Safari6+Chrome25+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS Safari?Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+

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

OfflineAudioCompletionEvent/OfflineAudioCompletionEvent

In all current engines.

Firefox53+SafariYesChrome57+

Opera42+Edge79+

Edge (Legacy)NoneIENone

Firefox for Android53+iOS SafariYesChrome for Android57+Android WebView57+Samsung Internet6.0+Opera Mobile42+

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

1.3.5.1. Attributes

OfflineAudioCompletionEvent/renderedBuffer In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AudioBuffer/AudioBuffer

Firefox53+SafariNoneChrome55+

Opera42+Edge79+

Edge (Legacy)NoneIENone

Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+

This interface represents a memory-resident audio asset. It can contain one or more channels with each channel appearing to be 32-bit floating-point linear PCM values with a nominal range of \([-1,1]\) but the values are not limited to this range. 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.

AudioBuffer

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

[Exposed=Window]
interface AudioBuffer {
  constructor (AudioBufferOptions options);
  readonly attribute float sampleRate;
  readonly attribute unsigned long length;
  readonly attribute double duration;
  readonly attribute unsigned long numberOfChannels;
  Float32Array getChannelData (unsigned long channel);
  undefined copyFromChannel (Float32Array destination,
                             unsigned long channelNumber,
                             optional unsigned long bufferOffset = 0);
  undefined copyToChannel (Float32Array source,
                           unsigned long channelNumber,
                           optional unsigned long bufferOffset = 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.constructor() method.
Parameter	Type	Nullable	Optional	Description
`options`	AudioBufferOptions	✘	✘	An `AudioBufferOptions` that determine the properties for this `AudioBuffer`.

1.4.2. Attributes

AudioBuffer/duration In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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]].

AudioBuffer/length In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ length, of type unsigned long, readonly

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

AudioBuffer/numberOfChannels In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ numberOfChannels, of type unsigned long, readonly

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

AudioBuffer/sampleRate In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AudioBuffer/copyFromChannel Firefox25+SafariNoneChrome43+ Opera30+Edge79+ Edge (Legacy)13+IENone Firefox for Android25+iOS SafariNoneChrome for Android43+Android WebView43+Samsung Internet4.0+Opera Mobile30+ copyFromChannel(destination, channelNumber, bufferOffset)

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 bufferOffset. Then the number of frames copied from buffer to destination is \(\max(0, \min(N_b - k, N_f))\). 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 channels of the `AudioBuffer`, an `IndexSizeError` MUST be thrown.
`bufferOffset`	unsigned long	✘	✔	An optional offset, defaulting to 0. Data from the `AudioBuffer` starting at this offset is copied to the `destination`.

Return type: undefined

AudioBuffer/copyToChannel Firefox25+SafariNoneChrome43+ Opera30+Edge79+ Edge (Legacy)13+IENone Firefox for Android25+iOS SafariNoneChrome for Android43+Android WebView43+Samsung Internet4.0+Opera Mobile30+ copyToChannel(source, channelNumber, bufferOffset)

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 bufferOffset. Then the number of frames copied from source to the buffer is \(\max(0, \min(N_b - k, N_f))\). 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 channels of the `AudioBuffer`, an `IndexSizeError` MUST be thrown.
`bufferOffset`	unsigned long	✘	✔	An optional offset, defaulting to 0. Data from the `source` is copied to the `AudioBuffer` starting at this offset.

Return type: undefined

AudioBuffer/getChannelData In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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 `[[number of channels]]` 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.

Note: Because AudioBuffer can only be created via createBuffer() or via the AudioBuffer constructor, this cannot throw.
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 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.

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 of all connections to that input. Please see § 4 Channel Up-Mixing and Down-Mixing for normative requirements and details.

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.

AudioNode

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

[Exposed=Window]
interface AudioNode : EventTarget {
  AudioNode connect (AudioNode destinationNode,
                     optional unsigned long output = 0,
                     optional unsigned long input = 0);
  undefined connect (AudioParam destinationParam, optional unsigned long output = 0);
  undefined disconnect ();
  undefined disconnect (unsigned long output);
  undefined disconnect (AudioNode destinationNode);
  undefined disconnect (AudioNode destinationNode, unsigned long output);
  undefined disconnect (AudioNode destinationNode,
                        unsigned long output,
                        unsigned long input);
  undefined disconnect (AudioParam destinationParam);
  undefined 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 factory method, called on a BaseAudioContext c, execute these steps:

Let node 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.
Call the constructor for n on node with c and option as arguments.
Return node

Initializing an object o 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.
For each member of dict passed in, execute these steps, with k the key of the member, and v its value. If any exceptions is thrown when executing these steps, abort the iteration and propagate the exception to the caller of the algorithm (constructor or factory method).
1. If k is the name of an AudioParam on this interface, set the value attribute of this AudioParam to v.
2. 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 § 4 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. AudioNode Tail-Time

An AudioNode can have a tail-time. This means that even when the AudioNode is fed silence, the output can be non-silent.

AudioNodes have a non-zero tail-time if they have internal processing state such that input in the past affects the future output. AudioNodes may continue to produce non-silent output for the calculated tail-time even after the input transitions from non-silent to silent.

1.5.3. AudioNode Lifetime

AudioNode can be actively processing during a render quantum, if any of the following conditions hold.

An AudioScheduledSourceNode is actively processing if and only if it is playing for at least part of the current rendering quantum.
A MediaElementAudioSourceNode is actively processing if and only if its mediaElement is playing for at least part of the current rendering quantum.
A MediaStreamAudioSourceNode or a MediaStreamTrackAudioSourceNode are actively processing when the associated MediaStreamTrack object has a readyState attribute equal to "live", a muted attribute equal to false and an enabled attribute equal to true.
A DelayNode in a cycle is actively processing only when the absolute value of any output sample for the current render quantum is greater than or equal to \( 2^{-126} \).
A ScriptProcessorNode is actively processing when its input or output is connected.
An AudioWorkletNode is actively processing when its AudioWorkletProcessor's [[callable process]] returns true and either its active source flag is true or any AudioNode connected to one of its inputs is actively processing.
All other AudioNodes start actively processing when any AudioNode connected to one of its inputs is actively processing, and stops actively processing when the input that was received from other actively processing AudioNode no longer affects the output.

Note: This takes into account AudioNodes that have a tail-time.

AudioNodes that are not actively processing output a single channel of silence.

1.5.4. Attributes

AudioNode/channelCount In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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 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.

AudioWorkletNode

See § 1.32.3.3.2 Configuring Channels with AudioWorkletNodeOptions Configuring Channels with AudioWorkletNodeOptions.

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 be greater than two, and a NotSupportedError exception MUST be thrown for any attempt to change it to a value greater than two.

DynamicsCompressorNode

The channel count cannot be greater than two, and a NotSupportedError exception MUST be thrown for any attempt to change it 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 it to a value greater than two.

ScriptProcessorNode

The channel count cannot be changed, and an NotSupportedError 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 it to a value greater than two.

See § 4 Channel Up-Mixing and Down-Mixing for more information on this attribute.

AudioNode/channelCountMode In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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 set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".
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 NotSupportedError 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 § 4 Channel Up-Mixing and Down-Mixing section for more information on this attribute.

AudioNode/channelInterpretation In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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 § 4 Channel Up-Mixing and Down-Mixing for more information on this attribute.

AudioNode/context In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ context, of type BaseAudioContext, readonly

The BaseAudioContext which owns this AudioNode.

AudioNode/numberOfInputs In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

AudioNode/numberOfOutputs In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

1.5.5. Methods

AudioNode/connect In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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. Multiple `AudioNode`s can be connected to the same `AudioNode`, this is described in Channel Upmixing and down mixing section.
`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-bounds, 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`.

Return type: AudioNode

Connects the AudioNode to an AudioParam, controlling the parameter value with an a-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 the `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-bounds, an `IndexSizeError` exception MUST be thrown.

Return type: undefined

AudioNode/disconnect In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ disconnect()

Disconnects all outgoing connections from the AudioNode.

No parameters.

Return type: undefined

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: undefined

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: undefined

disconnect(destinationNode, output)

Disconnects a specific output of the AudioNode from any and all inputs 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-bounds, an `IndexSizeError` exception MUST be thrown.

Return type: undefined

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 input to 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-bounds, 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: undefined

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: undefined

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-bounds, an `IndexSizeError` exception MUST be thrown.

Return type: undefined

1.5.6. `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.

AudioNodeOptions

Firefox53+Safari?Chrome55+

Opera42+Edge79+

Edge (Legacy)NoneIENone

Firefox for Android53+iOS Safari?Chrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+

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

1.5.6.1. 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.

1.6. The `AudioParam` Interface

AudioParam

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

AudioParam controls an individual aspect of an AudioNode's functionality, 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 each individual 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".

AudioParam/defaultValue In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ defaultValue, of type float, readonly

Initial value for the value attribute.

AudioParam/maxValue In all current engines. Firefox53+Safari6+Chrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariYesChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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.

AudioParam/minValue In all current engines. Firefox53+Safari6+Chrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariYesChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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.

AudioParam/value In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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. See § 1.6.3 Computation of Value for the algorithm for the value that is returned.

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

AudioParam/cancelAndHoldAtTime In only one current engine. FirefoxNoneSafariNoneChrome57+ Opera44+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android57+Android WebView57+Samsung Internet7.0+Opera Mobile43+ 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

AudioParam/cancelScheduledValues In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

For a setValueCurveAtTime(), let \(T_0\) and \(T_D\) be the corresponding startTime and duration, respectively of this event. Then if cancelTime is in the range \([T_0, T_0 + T_D]\), the event is removed from the timeline.

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

AudioParam/exponentialRampToValueAtTime FirefoxNoneSafari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for AndroidNoneiOS SafariYesChrome for AndroidNoneAndroid WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AudioParam/linearRampToValueAtTime FirefoxNoneSafari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for AndroidNoneiOS Safari?Chrome for AndroidNoneAndroid WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AudioParam/setTargetAtTime In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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 [[current 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

AudioParam/setValueAtTime In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AudioParam/setValueCurveAtTime In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

The computation of the value of an AudioParam consists of two parts:

the paramIntrinsicValue value that is computed from the value attribute and any automation events.
the paramComputedValue that is the final value controlling the audio DSP and is computed by the audio rendering thread during each render quantum.

These values MUST be computed as follows:

paramIntrinsicValue 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. If automation events are removed from a given time range, then the paramIntrinsicValue 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.
Set [[current value]] to the value of paramIntrinsicValue at the beginning of this render quantum.
paramComputedValue is the sum of the paramIntrinsicValue value and the value of the input AudioParam buffer. If the sum is NaN, replace the sum with the defaultValue.
If this AudioParam is a compound parameter, compute its final value with other AudioParams.
Set computedValue to paramComputedValue.

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

const curveLength = 44100;const curve = new Float32Array(curveLength);for (const i = 0; i < curveLength; ++i)  curve[i] = Math.sin(Math.PI * i / curveLength);const t0 = 0;const t1 = 0.1;const t2 = 0.2;const t3 = 0.3;const t4 = 0.325;const t5 = 0.5;const t6 = 0.6;const t7 = 0.7;const t8 = 1.0;const 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

AudioScheduledSourceNode

In all current engines.

Firefox53+Safari14+Chrome57+

Opera44+Edge79+

Edge (Legacy)NoneIENone

Firefox for Android53+iOS Safari14+Chrome for Android57+Android WebView57+Samsung Internet7.0+Opera Mobile43+

The interface represents the common features of 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.

An AudioScheduledSourceNode is said to be playing when its associated BaseAudioContext's currentTime is greater or equal to the time the AudioScheduledSourceNode is set to start, and less than the time it’s set to stop.

AudioScheduledSourceNodes are created with an internal boolean slot [[source started]], initially set to false.

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

1.7.1. Attributes

AudioScheduledSourceNode/ended_event In all current engines. Firefox25+Safari7+Chrome30+ Opera17+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari7+Chrome for Android30+Android WebView37+Samsung Internet2.0+Opera Mobile18+ AudioScheduledSourceNode/onended In all current engines. Firefox25+Safari7+Chrome30+ Opera17+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari7+Chrome for Android30+Android WebView37+Samsung Internet2.0+Opera Mobile18+ onended, of type EventHandler

A property used to set the EventHandler (described in HTML[HTML]) for the ended event that is dispatched for 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

AudioScheduledSourceNode/start In all current engines. Firefox25+Safari6.1+Chrome24+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari7+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ start(when)

Schedules a sound to playback at an exact time.

When this method is called, execute these steps:

If this AudioScheduledSourceNode internal slot [[source started]] is true, an InvalidStateError exception MUST be thrown.
Check for any errors that must be thrown due to parameter constraints described below. If any exception is thrown during this step, abort those steps.
Set the internal slot [[source started]] on this AudioScheduledSourceNode to true.
Queue a control message to start the AudioScheduledSourceNode, including the parameter values in the message.
Send a control message to the associated AudioContext to start running its rendering thread only when all the following conditions are met:
1. The context’s [[control thread state]] is "suspended".
2. The context is allowed to start.
3. [[suspended by user]] flag is false.
NOTE: This allows start() to start an AudioContext that would otherwise not be allowed to start.

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: undefined

AudioScheduledSourceNode/stop In all current engines. Firefox25+Safari6.1+Chrome24+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari7+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ 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 this AudioScheduledSourceNode internal slot [[source started]] is not true, 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 message.

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: undefined

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	No

AnalyserNode

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

[Exposed=Window]
interface AnalyserNode : AudioNode {
  constructor (BaseAudioContext context, optional AnalyserOptions options = {});
  undefined getFloatFrequencyData (Float32Array array);
  undefined getByteFrequencyData (Uint8Array array);
  undefined getFloatTimeDomainData (Float32Array array);
  undefined 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/AnalyserNode Firefox53+SafariNoneChrome55+ Opera42+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+ AnalyserNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the AnalyserNode.constructor() 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

AnalyserNode/fftSize In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

AnalyserNode/frequencyBinCount In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ frequencyBinCount, of type unsigned long, readonly

Half the FFT size.

AnalyserNode/maxDecibels In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

AnalyserNode/minDecibels In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

AnalyserNode/smoothingTimeConstant In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AnalyserNode/getByteFrequencyData In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ getByteFrequencyData(array)

Get a reference to the bytes held by the Uint8Array passed as an argument. Copies the current frequency data to those bytes. 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: undefined

AnalyserNode/getByteTimeDomainData In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ getByteTimeDomainData(array)

Get a reference to the bytes held by the Uint8Array passed as an argument. Copies the current time-domain data (waveform data) into those bytes. 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: undefined

AnalyserNode/getFloatFrequencyData In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ getFloatFrequencyData(array)

Get a reference to the bytes held by the Float32Array passed as an argument. Copies the current frequency data into those bytes. 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: undefined

AnalyserNode/getFloatTimeDomainData Firefox25+SafariNoneChrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariNoneChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ getFloatTimeDomainData(array)

Get a reference to the bytes held by the Float32Array passed as an argument. Copies the current time-domain data (waveform data) into those bytes. 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: undefined

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 real and imaginary frequency data.
Smooth over time the frequency domain data.
Convert 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

$$
  X[k] = \frac{1}{N} \sum_{n = 0}^{N - 1} \hat{x}[n]\, W^{-kn}_{N}
$$

for \(k = 0, \dots, N/2-1\) where \(W_N = e^{2\pi i/N}\).

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

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

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

AudioBufferSourceNode/AudioBufferSourceNode

Firefox53+SafariNoneChrome55+

Opera42+Edge79+

Edge (Legacy)NoneIENone

Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+

AudioBufferSourceNode

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

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() descriptions.

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

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

In addition, if the buffer has more than one channel, then the AudioBufferSourceNode output must change to a single channel of silence at the beginning of a render quantum after the time at which any one of the following conditions holds:

the end of the buffer has been reached;
the duration has been reached;
the stop time has been reached.

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

1.9.1. Constructors

AudioBufferSourceNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the AudioBufferSourceNode.constructor() 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

AudioBufferSourceNode/buffer In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

AudioBufferSourceNode/detune Firefox40+SafariNoneChrome44+ Opera31+Edge79+ Edge (Legacy)13+IENone Firefox for Android40+iOS SafariNoneChrome for Android44+Android WebView44+Samsung Internet4.0+Opera Mobile32+ 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

AudioBufferSourceNode/loop In all current engines. Firefox25+Safari6+Chrome15+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

AudioBufferSourceNode/loopEnd In all current engines. Firefox25+Safari6+Chrome24+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ 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.

AudioBufferSourceNode/loopStart In all current engines. Firefox25+Safari6+Chrome24+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ 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.

AudioBufferSourceNode/playbackRate In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

AudioBufferSourceNode/start In all current engines. Firefox25+Safari6.1+Chrome24+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari7+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ start(when, offset, duration)

Schedules a sound to playback at an exact time.

When this method is called, execute these steps:

If this AudioBufferSourceNode internal slot [[source started]] is true, an InvalidStateError exception MUST be thrown.
Check for any errors that must be thrown due to parameter constraints described below. If any exception is thrown during this step, abort those steps.
Set the internal slot [[source started]] on this AudioBufferSourceNode to true.
Queue a control message to start the AudioBufferSourceNode, including the parameter values in the message.
Acquire the contents of the buffer if the buffer has been set.
Send a control message to the associated AudioContext to start running its rendering thread only when all the following conditions are met:
1. The context’s [[control thread state]] is suspended.
2. The context is allowed to start.
3. [[suspended by user]] flag is false.
NOTE: This allows start() to start an AudioContext that would otherwise not be allowed to start.

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`, `playbackRate` is positive or zero, and `loop` is `true`, playback will begin at `loopEnd`. If `offset` is greater than `loopStart`, `playbackRate` is negative, and `loop` is `true`, playback will begin at `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: undefined

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:

A starting offset, which can be expressed with sub-sample precision.
Loop points, which can be expressed with sub-sample precision and can vary dynamically during playback.
Playback rate and detuning parameters, which combine to yield a single computedPlaybackRate that can assume finite 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 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  const output = []; // accumulates rendered sample frames  // Combine the two k-rate parameters affecting playback rate  const 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.      if (loop && computedPlaybackRate >= 0 && offset >= actualLoopEnd) {        offset = actualLoopEnd;      }      if (computedPlaybackRate < 0 && loop && offset < actualLoopStart) {        offset = actualLoopStart;      }      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.  Schedule a change to set the number of output channels to 1.  }  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

AudioDestinationNode

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

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	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	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

AudioDestinationNode/maxChannelCount In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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 § 6 Spatialization/Panning for more details about spatialization.

The positionX, positionY, and 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, and 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 § 6 Spatialization/Panning section.

AudioListener

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

[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;
  undefined setPosition (float x, float y, float z);
  undefined setOrientation (float x, float y, float z, float xUp, float yUp, float zUp);
};

1.11.1. Attributes

AudioListener/forwardX In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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`"

AudioListener/forwardY In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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`"

AudioListener/forwardZ In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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`"

AudioListener/positionX In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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`"

AudioListener/positionY In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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`"

AudioListener/positionZ In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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`"

AudioListener/upX In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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`"

AudioListener/upY In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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`"

AudioListener/upZ In only one current engine. FirefoxNoneSafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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.

setOrientation() describes which direction the listener is pointing in the 3D cartesian coordinate space. Both a forward vector and an up vector are provided. 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 § 6 Spatialization/Panning.

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

The xUp, yUp, and 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	Description
`x`	float	✘	✘	forward x direction fo the `AudioListener`
`y`	float	✘	✘	forward y direction fo the `AudioListener`
`z`	float	✘	✘	forward z direction fo the `AudioListener`
`xUp`	float	✘	✘	up x direction fo the `AudioListener`
`yUp`	float	✘	✘	up y direction fo the `AudioListener`
`zUp`	float	✘	✘	up z direction fo the `AudioListener`

Return type: undefined

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.

setPosition() 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, and 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	Description
`x`	float	✘	✘	x-coordinate of the position of the `AudioListener`
`y`	float	✘	✘	y-coordinate of the position of the `AudioListener`
`z`	float	✘	✘	z-coordinate of the position of the `AudioListener`

1.11.3. Processing

Because AudioListener's parameters can be connected with AudioNodes and they can also affect the output of PannerNodes in the same graph, the node ordering algorithm should take the AudioListener into consideration when computing the order of processing. For this reason, all the PannerNodes in the graph have the AudioListener as input.

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]
interface AudioProcessingEvent : Event {
  constructor (DOMString type, AudioProcessingEventInit eventInitDict);
  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	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. 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. 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.

BiquadFilterNode/BiquadFilterNode

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BiquadFilterNode

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

1.13.1. Constructors

BiquadFilterNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the BiquadFilterNode.constructor() 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

BiquadFilterNode/Q In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ Q, of type AudioParam, readonly

The Q factor of the filter.

For lowpass and highpass filters the Q value is interpreted to be in dB. For these filters the nominal range is \([-Q_{lim}, Q_{lim}]\) where \(Q_{lim}\) is the largest value for which \(10^{Q/20}\) does not overflow. This is approximately \(770.63678\).

For the bandpass, notch, allpass, and peaking filters, this value is a linear value. The value is related to the bandwidth of the filter and hence should be a positive value. The nominal range is \([0, 3.4028235e38]\), the upper limit being the most-positive-single-float.

This is not used for the lowshelf and highshelf filters.

Parameter	Value	Notes
`defaultValue`	1
`minValue`	most-negative-single-float	Approximately -3.4028235e38, but see above for the actual limits for different filters
`maxValue`	most-positive-single-float	Approximately 3.4028235e38, but see above for the actual limits for different filters
`automationRate`	"`a-rate`"

BiquadFilterNode/detune In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ detune, of type AudioParam, readonly

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

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`"

BiquadFilterNode/frequency In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ frequency, of type AudioParam, readonly

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

Parameter	Value	Notes
`defaultValue`	350
`minValue`	0
`maxValue`	Nyquist frequency
`automationRate`	"`a-rate`"

BiquadFilterNode/gain In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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`	\(\approx 1541\)	This value is approximately \(40\ \log_{10} \mathrm{FLT\_MAX}\) where FLT_MAX is the largest `float` value.
`automationRate`	"`a-rate`"

BiquadFilterNode/type In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

BiquadFilterNode/getFrequencyResponse In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ getFrequencyResponse(frequencyHz, magResponse, phaseResponse)

Given the [[current value]] from each of the filter parameters, 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, in Hz, 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: undefined

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 BiquadFilterNode processes audio with a transfer function of

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

which is equivalent to a time-domain equation of:

$$
a_0 y(n) + a_1 y(n-1) + a_2 y(n-2) =
  b_0 x(n) + b_1 x(n-1) + b_2 x(n-2)
$$

The initial filter state is 0.

Note: While fixed filters are stable, it is possible to create unstable biquad filters using automations of AudioParams. It is the developers responsibility to manage this.

Note: The UA may produce a warning to notify the user that NaN values have occurred in the filter state. This is usually indicative of an unstable filter.

The coefficients in the transfer function above are different for each node type. The following intermediate variables 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	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 when any of the inputs is actively processing. If none of the inputs are actively processing, then output is a single channel of silence.

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.

ChannelMergerNode/ChannelMergerNode

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ChannelMergerNode

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

1.14.1. Constructors

ChannelMergerNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the ChannelMergerNode.constructor() 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	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.

ChannelSplitterNode/ChannelSplitterNode

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ChannelSplitterNode

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

1.15.1. Constructors

ChannelSplitterNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the ChannelSplitterNode.constructor() 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

ConstantSourceNode

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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	No

ConstantSourceNode/ConstantSourceNode

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

1.16.1. Constructors

ConstantSourceNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the ConstantSourceNode.constructor() 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

ConstantSourceNode/offset Firefox52+SafariNoneChrome56+ OperaNoneEdge79+ Edge (Legacy)NoneIENone Firefox for Android52+iOS SafariNoneChrome for AndroidNoneAndroid WebViewNoneSamsung InternetNoneOpera MobileNone 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

ConvolverNode

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Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

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	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.

ConvolverNode/ConvolverNode

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

1.17.1. Constructors

ConvolverNode(context, options)

When the constructor is called with a BaseAudioContext context and an option object options, execute these steps:

Set the attributes normalize to the inverse of the value of disableNormalization.
If buffer is present, set the buffer attribute to its value.

Note: This means that the buffer will be normalized according to the value of the normalize attribute.
Let o be new AudioNodeOptions dictionary.
If channelCount is present in options, set channelCount on o with the same value.
If channelCountMode is present in options, set channelCountMode on o with the same value.
If channelInterpretation is present in options, set channelInterpretation on o with the same value.
Initialize the AudioNode this, with c and o as argument.

Arguments for the ConvolverNode.constructor() 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

ConvolverNode/buffer In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ buffer, of type AudioBuffer, nullable

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.

When setting the buffer attribute, execute the following steps synchronously:

If the buffer number of channels is not 1, 2, 4, or if the sample-rate of the buffer is not the same as the sample-rate of its associated BaseAudioContext, a NotSupportedError MUST be thrown.
Acquire the content of the AudioBuffer.

Note: If the buffer is set to an new buffer, audio may glitch. If this is undesirable, it is recommended to create a new ConvolverNode to replace the old, possibly cross-fading between the two.

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

ConvolverNode/normalize In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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) {  const GainCalibration = 0.00125;  const GainCalibrationSampleRate = 44100;  const MinPower = 0.000125;  // Normalize by RMS power.  const numberOfChannels = buffer.numberOfChannels;  const length = buffer.length;  let power = 0;  for (let i = 0; i < numberOfChannels; i++) {    let channelPower = 0;    const channelData = buffer.getChannelData(i);    for (let j = 0; j < length; j++) {      const 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;  let 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.

If this node is not actively processing, the output is a single channel of silence.

Note: The diagrams below show the outputs when actively processing.

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

1.18. The `DelayNode` Interface

DelayNode

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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	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.

DelayNode/DelayNode

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

1.18.1. Constructors

DelayNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the DelayNode.constructor() 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

DelayNode/delayTime In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

1.18.4. Processing

A DelayNode has an internal buffer that holds delayTime seconds of audio.

The processing of a DelayNode is broken down in two parts: writing to the delay line, and reading from the delay line. This is done via two internal AudioNodes (that are not available to authors and exist only to ease the description of the inner workings of the node). Both are created from a DelayNode.

Creating a DelayWriter for a DelayNode means creating an object that has the same interface as an AudioNode, and that writes the input audio into the internal buffer of the DelayNode. It has the same input connections as the DelayNode it was created from.

Creating a DelayReader for a DelayNode means creating an object that has the same interface as an AudioNode, and that can read the audio data from the internal buffer of the DelayNode. It is connected to the same AudioNodes as the DelayNode it was created from. A DelayReader is a source node.

When processing an input buffer, a DelayWriter MUST write the audio to the internal buffer of the DelayNode.

When producing an output buffer, a DelayReader MUST yield exactly the audio that was written to the corresponding DelayWriter delayTime seconds ago.

Note: This means that channel count changes are reflected after the delay time has passed.

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	Yes	This node has a tail-time such that this node continues to output non-silent audio with zero input due to the look-ahead delay.

DynamicsCompressorNode/DynamicsCompressorNode

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DynamicsCompressorNode

In all current engines.

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[Exposed=Window]
interface DynamicsCompressorNode : AudioNode {
  constructor (BaseAudioContext context,
               optional DynamicsCompressorOptions options = {});
  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)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Let [[internal reduction]] be a private slot on this, that holds a floating point number, in decibels. Set [[internal reduction]] to 0.0.

Arguments for the DynamicsCompressorNode.constructor() 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

DynamicsCompressorNode/attack In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

DynamicsCompressorNode/knee In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

DynamicsCompressorNode/ratio In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

DynamicsCompressorNode/reduction In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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]].

DynamicsCompressorNode/release In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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

DynamicsCompressorNode/threshold In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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 a new class, EnvelopeFollower, that instantiates a special object that behaves like an AudioNode, described below:

const delay = new DelayNode(context, {delayTime: 0.006});
const gain = new GainNode(context);
const 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. Subtract detector average from 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 be the product of compressor gain and 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 be the inverse of full range gain.
Return the result of taking the 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 and sampled at the time of processing of this block (as k-rate parameters).
Calculate the sum of threshold plus knee also sampled at the time of processing of this block (as k-rate parameters).
Let knee end threshold have the value of this sum converted to linear units.
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 knee end threshold, 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 \(10^{v/20}\).

1.20. The `GainNode` Interface

Changing the gain of an audio signal is a fundamental operation 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	No

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

GainNode/GainNode

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GainNode

In all current engines.

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Opera15+Edge79+

Edge (Legacy)12+IENone

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

1.20.1. Constructors

GainNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the GainNode.constructor() 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

GainNode/gain In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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	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.

IIRFilterNode/IIRFilterNode

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Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+

IIRFilterNode

Firefox50+SafariNoneChrome49+

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Firefox for Android50+iOS SafariNoneChrome for Android49+Android WebView49+Samsung Internet5.0+Opera Mobile36+

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

1.21.1. Constructors

IIRFilterNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

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

1.21.2. Methods

IIRFilterNode/getFrequencyResponse Firefox50+SafariNoneChrome49+ Opera36+Edge79+ Edge (Legacy)14+IENone Firefox for Android50+iOS SafariNoneChrome for Android49+Android WebView49+Samsung Internet5.0+Opera Mobile36+ 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.

Arguments for the IIRFilterNode.getFrequencyResponse() method.
Parameter	Type	Nullable	Optional	Description
`frequencyHz`	Float32Array	✘	✘	This parameter specifies an array of frequencies, in Hz, 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: undefined

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. See feedforward argument of createIIRFilter() for other constraints.
feedback, of type sequence<double>: The feedback coefficients for the IIRFilterNode. This member is required. 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.

Note: The UA may produce a warning to notify the user that NaN values have occurred in the filter state. This is usually indicative of an unstable filter.

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.

If the sample rate of the HTMLMediaElement differs from the sample rate of the associated AudioContext, then the output from the HTMLMediaElement must be resampled to match the context’s sample rate.

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.

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

MediaElementAudioSourceNode/MediaElementAudioSourceNode

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MediaElementAudioSourceNode

In all current engines.

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

1.22.1. Constructors

MediaElementAudioSourceNode(context, options)

initialize the AudioNode this, with context and options as arguments.

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

1.22.2. Attributes

MediaElementAudioSourceNode/mediaElement In all current engines. Firefox70+Safari6+ChromeYes OperaYesEdgeYes Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS Safari6+Chrome for AndroidYesAndroid WebViewYesSamsung InternetYesOpera MobileYes mediaElement, of type HTMLMediaElement, readonly: The HTMLMediaElement used when constructing this MediaElementAudioSourceNode.

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	No

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

MediaStreamAudioDestinationNode/MediaStreamAudioDestinationNode

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Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+

MediaStreamAudioDestinationNode

In all current engines.

Firefox25+Safari6+Chrome14+

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Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

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

1.23.1. Constructors

MediaStreamAudioDestinationNode(context, options)

Initialize the AudioNode this, with context and options as arguments.

Arguments for the MediaStreamAudioDestinationNode.constructor() 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

MediaStreamAudioDestinationNode/stream In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)18IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

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. When the MediaStreamTrack ends, this AudioNode outputs one channel of silence.

If the sample rate of the MediaStreamTrack differs from the sample rate of the associated AudioContext, then the output of the MediaStreamTrack is resampled to match the context’s sample rate.

MediaStreamAudioSourceNode

In all current engines.

Firefox25+Safari11+Chrome23+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS Safari11+Chrome for AndroidYesAndroid WebView37+Samsung InternetYesOpera Mobile14+

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

1.24.1. Constructors

MediaStreamAudioSourceNode/MediaStreamAudioSourceNode Firefox53+SafariNoneChrome55+ Opera42+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+ MediaStreamAudioSourceNode(context, options)

If the mediaStream member of options does not reference a MediaStream that has at least one MediaStreamTrack whose kind attribute has the value "audio", throw an InvalidStateError and abort these steps. Else, let this stream be inputStream.
Let tracks be the list of all MediaStreamTracks of inputStream that have a kind of "audio".
Sort the elements in tracks based on their id attribute using an ordering on sequences of code unit values.
Initialize the AudioNode this, with context and options as arguments.
Set an internal slot [[input track]] on this MediaStreamAudioSourceNode to be the first element of tracks. This is the track used as the input audio for this MediaStreamAudioSourceNode.

After construction, any change to the MediaStream that was passed to the constructor do not affect the underlying output of this AudioNode.

The slot [[input track]] is only used to keep a reference to the MediaStreamTrack.

Note: This means that when removing the track chosen by the constructor of the MediaStreamAudioSourceNode from the MediaStream passed into this constructor, the MediaStreamAudioSourceNode will still take its input from the same track.

Note: The behaviour for picking the track to output is arbitrary for legacy reasons. MediaStreamTrackAudioSourceNode can be used instead to be explicit about which track to use as input.

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

1.24.2. Attributes

MediaStreamAudioSourceNode/mediaStream In all current engines. Firefox70+Safari11+Chrome23+ OperaYesEdge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS Safari11+Chrome for AndroidYesAndroid WebView37+Samsung InternetYesOpera MobileYes mediaStream, of type MediaStream, readonly: The MediaStream used when constructing this MediaStreamAudioSourceNode.

1.24.3. `MediaStreamAudioSourceOptions`

This specifies the options for constructing a MediaStreamAudioSourceNode.

MediaStreamAudioSourceOptions

In all current engines.

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Firefox for Android53+iOS Safari?Chrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile14+

dictionary MediaStreamAudioSourceOptions {
  required MediaStream mediaStream;
};

1.24.3.1. Dictionary `MediaStreamAudioSourceOptions` Members

MediaStreamAudioSourceOptions/mediaStream Firefox53+Safari?Chrome55+ Opera?Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS Safari?Chrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile? 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.

If the sample rate of the MediaStreamTrack differs from the sample rate of the associated AudioContext, then the output of the mediaStreamTrack is resampled to match the context’s sample rate.

MediaStreamTrackAudioSourceNode

In only one current engine.

Firefox68+SafariNoneChromeNone

OperaNoneEdgeNone

Edge (Legacy)NoneIENone

Firefox for Android68+iOS SafariNoneChrome for AndroidNoneAndroid WebViewNoneSamsung InternetNoneOpera MobileNone

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

1.25.1. Constructors

MediaStreamTrackAudioSourceNode/MediaStreamTrackAudioSourceNode In only one current engine. Firefox68+SafariNoneChromeNone OperaNoneEdgeNone Edge (Legacy)NoneIENone Firefox for Android68+iOS SafariNoneChrome for AndroidNoneAndroid WebViewNoneSamsung InternetNoneOpera MobileNone MediaStreamTrackAudioSourceNode(context, options)

If the mediaStreamTrack's kind attribute is not "audio", throw an InvalidStateError and abort these steps.
Initialize the AudioNode this, with context and options as arguments.

Arguments for the MediaStreamTrackAudioSourceNode.constructor() method.
Parameter	Type	Nullable	Optional	Description
`context`	AudioContext	✘	✘	The `AudioContext` 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.

MediaStreamTrackAudioSourceOptions

In only one current engine.

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dictionary MediaStreamTrackAudioSourceOptions {
  required MediaStreamTrack mediaStreamTrack;
};

1.25.2.1. Dictionary `MediaStreamTrackAudioSourceOptions` Members

MediaStreamTrackAudioSourceOptions/mediaStreamTrack In only one current engine. Firefox68+SafariNoneChromeNone OperaNoneEdgeNone Edge (Legacy)NoneIENone Firefox for Android68+iOS SafariNoneChrome for AndroidNoneAndroid WebViewNoneSamsung InternetNoneOpera MobileNone 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].

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	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

OscillatorNode

In all current engines.

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Firefox for Android25+iOS Safari6+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+

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

1.26.1. Constructors

OscillatorNode/OscillatorNode Firefox53+SafariNoneChrome55+ Opera42+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+ OscillatorNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the OscillatorNode.constructor() 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`.

1.26.2. Attributes

OscillatorNode/detune In all current engines. Firefox25+Safari6+Chrome20+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ 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`"

OscillatorNode/frequency In all current engines. Firefox25+Safari6+Chrome20+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ 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`"

OscillatorNode/type In all current engines. Firefox25+Safari6+Chrome20+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android25+Android WebView37+Samsung Internet1.5+Opera Mobile14+ 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.

1.26.3. Methods

OscillatorNode/setPeriodicWave In all current engines. Firefox25+Safari8+Chrome30+ Opera17+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari8+Chrome for Android30+Android WebView37+Samsung Internet2.0+Opera Mobile18+ setPeriodicWave(periodicWave)

Sets an arbitrary custom periodic waveform given a PeriodicWave.

Arguments for the OscillatorNode.setPeriodicWave() method.
Parameter	Type	Nullable	Optional	Description
`periodicWave`	PeriodicWave	✘	✘	custom waveform to be used by the oscillator

Return type: undefined

1.26.4. `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;
};

1.26.4.1. 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".

1.26.5. Basic Waveform Phase

The idealized mathematical waveforms for the various oscillator types are defined below. 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 disableNormalization set to false, 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:

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

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

1.27. 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 BaseAudioContext'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	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 the tail-time is zero.

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.

If the node is actively processing, the output of this node is hard-coded to stereo (2 channels) and cannot be configured. If the node is not actively processing, then the output is a single channel of silence.

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:

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

where \(d’_{ref} = \min\left(d_{ref}, d_{max}\right)\) and \(d’_{max} = \max\left(d_{ref}, d_{max}\right)\). 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 \(\left[d’_{ref},\, d’_{max}\right]\).

"inverse"

An inverse distance model which calculates distanceGain according to:

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

That is, \(d\) is clamped to the interval \(\left[d_{ref},\, \infty\right)\). 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:

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

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

PannerNode

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

[Exposed=Window]
interface PannerNode : AudioNode {
  constructor (BaseAudioContext context, optional PannerOptions options = {});
  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;
  undefined setPosition (float x, float y, float z);
  undefined setOrientation (float x, float y, float z);
};

1.27.1. Constructors

PannerNode/PannerNode Firefox53+SafariNoneChrome55+ Opera42+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+ PannerNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the PannerNode.constructor() 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`.

1.27.2. Attributes

PannerNode/coneInnerAngle In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ coneInnerAngle, of type double

A parameter for directional audio sources that 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].

PannerNode/coneOuterAngle In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ coneOuterAngle, of type double

A parameter for directional audio sources that 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].

PannerNode/coneOuterGain In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ coneOuterGain, of type double

A parameter for directional audio sources that 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.

PannerNode/distanceModel In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ distanceModel, of type DistanceModelType

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

PannerNode/maxDistance In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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.

PannerNode/orientationX Firefox50+SafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for Android50+iOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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.

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

PannerNode/orientationY Firefox50+SafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for Android50+iOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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

PannerNode/orientationZ Firefox50+SafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for Android50+iOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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

PannerNode/panningModel In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ panningModel, of type PanningModelType

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

PannerNode/positionX Firefox50+SafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for Android50+iOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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

PannerNode/positionY Firefox50+SafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for Android50+iOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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

PannerNode/positionZ Firefox50+SafariNoneChrome52+ Opera39+Edge79+ Edge (Legacy)NoneIENone Firefox for Android50+iOS SafariNoneChrome for Android52+Android WebView52+Samsung Internet6.0+Opera Mobile41+ 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

PannerNode/refDistance In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ refDistance, of type double

A reference distance for reducing volume as source moves further from the listener. For distances less than this, the volume is not reduced. The default value is 1. A RangeError exception MUST be thrown if this is set to a negative value.

PannerNode/rolloffFactor In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS Safari6+Chrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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 \([0, \infty)\).

Note that the clamping happens as part of the processing of the distance computation. The attribute reflects the value that was set and is not modified.

1.27.3. 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: undefined

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: undefined

1.27.4. `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;
};

1.27.4.1. 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 \(x\)-component value for the orientationX AudioParam.
orientationY, of type float, defaulting to 0: The initial \(y\)-component value for the orientationY AudioParam.
orientationZ, of type float, defaulting to 0: The initial \(z\)-component 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 \(x\)-coordinate value for the positionX AudioParam.
positionY, of type float, defaulting to 0: The initial \(y\)-coordinate value for the positionY AudioParam.
positionZ, of type float, defaulting to 0: The initial \(z\)-coordinate 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.

1.27.5. Channel Limitations

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

1.28. 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.

PeriodicWave/PeriodicWave

Firefox53+Safari?Chrome55+

Opera42+Edge79+

Edge (Legacy)NoneIENone

Firefox for Android53+iOS Safari?Chrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+

PeriodicWave

In all current engines.

Firefox25+Safari8+Chrome30+

Opera17+Edge79+

Edge (Legacy)18IENone

Firefox for Android26+iOS Safari8+Chrome for Android30+Android WebView37+Samsung Internet2.0+Opera Mobile18+

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

1.28.1. Constructors

PeriodicWave(context, options)

Let p be a new PeriodicWave object. Let [[real]] and [[imag]] be two internal slots of type Float32Array, and let [[normalize]] be an internal slot.
Process options according to one of the following cases:
1. If both options.real and options.imag are present
  1. If the lengths of options.real and options.imag are different or if either length is less than 2, throw an IndexSizeError and abort this algorithm.
  2. Set [[real]] and [[imag]] to new arrays with the same length as options.real.
  3. Copy all elements from options.real to [[real]] and options.imag to [[imag]].
2. If only options.real is present
  1. If length of options.real is less than 2, throw an IndexSizeError and abort this algorithm.
  2. Set [[real]] and [[imag]] to arrays with the same length as options.real.
  3. Copy options.real to [[real]] and set [[imag]] to all zeros.
3. If only options.imag is present
  1. If length of options.imag is less than 2, throw an IndexSizeError and abort this algorithm.
  2. Set [[real]] and [[imag]] to arrays with the same length as options.imag.
  3. Copy options.imag to [[imag]] and set [[real]] to all zeros.
4. Otherwise
  1. Set [[real]] and [[imag]] to zero-filled arrays of length 2.
  2. Set element at index 1 of [[imag]] to 1.
  Note: When setting this PeriodicWave on an OscillatorNode, this is equivalent to using the built-in type "sine".
Set element at index 0 of both [[real]] and [[imag]] to 0. (This sets the DC component to 0.)
Initialize [[normalize]] to the inverse of the disableNormalization attribute of the PeriodicWaveConstraints on the PeriodicWaveOptions.
Return p.

Arguments for the PeriodicWave.constructor() 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`.

1.28.2. `PeriodicWaveConstraints`

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

dictionary PeriodicWaveConstraints {
  boolean disableNormalization = false;
};

1.28.2.1. 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.

1.28.3. `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;
};

1.28.3.1. 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. The second element (index 1) represents the fundamental frequency. The third represents the first overtone and so on.
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. The second element (index 1) represents the fundmental frequency. The third represents the first overtone and so on.

1.28.4. Waveform Generation

The createPeriodicWave() method takes two arrays to specify the Fourier coefficients of the PeriodicWave. Let \(a\) and \(b\) represent the [[real]] and [[imag]] arrays of length \(L\), respectively. 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.

1.28.5. 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.

1.28.6. 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"

$$
  b[n] = \frac{8\sin\dfrac{n\pi}{2}}{(\pi n)^2}
$$

1.29. 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	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;
};

1.29.1. 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.

1.30. The `StereoPannerNode` Interface

This interface represents a processing node which positions an incoming audio stream in a stereo image using a low-cost 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	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.

StereoPannerNode

Firefox37+SafariNoneChrome41+

Opera28+Edge79+

Edge (Legacy)12+IENone

Firefox for Android37+iOS SafariNoneChrome for Android41+Android WebView41+Samsung Internet4.0+Opera Mobile28+

[Exposed=Window]
interface StereoPannerNode : AudioNode {
  constructor (BaseAudioContext context, optional StereoPannerOptions options = {});
  readonly attribute AudioParam pan;
};

1.30.1. Constructors

StereoPannerNode/StereoPannerNode Firefox53+SafariNoneChrome55+ Opera42+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+ StereoPannerNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Arguments for the StereoPannerNode.constructor() 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`.

1.30.2. Attributes

StereoPannerNode/pan Firefox37+SafariNoneChrome41+ Opera28+Edge79+ Edge (Legacy)12+IENone Firefox for Android37+iOS SafariNoneChrome for Android41+Android WebView41+Samsung Internet4.0+Opera Mobile28+ 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`"

1.30.3. `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;
};

1.30.3.1. Dictionary `StereoPannerOptions` Members

pan, of type float, defaulting to 0: The initial value for the pan AudioParam.

1.30.4. 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.

1.31. 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	Maybe	There is a tail-time 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.

enum OverSampleType {
  "none",
  "2x",
  "4x"
};

Enumeration description
"`none`"	Don’t oversample
"`2x`"	Oversample two times
"`4x`"	Oversample four times

WaveShaperNode

In all current engines.

Firefox25+Safari6+Chrome14+

Opera15+Edge79+

Edge (Legacy)12+IENone

Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+

[Exposed=Window]
interface WaveShaperNode : AudioNode {
  constructor (BaseAudioContext context, optional WaveShaperOptions options = {});
  attribute Float32Array? curve;
  attribute OverSampleType oversample;
};

1.31.1. Constructors

WaveShaperNode/WaveShaperNode Firefox53+SafariNoneChrome55+ Opera42+Edge79+ Edge (Legacy)NoneIENone Firefox for Android53+iOS SafariNoneChrome for Android55+Android WebView55+Samsung Internet6.0+Opera Mobile42+ WaveShaperNode(context, options)

When the constructor is called with a BaseAudioContext c and an option object option, the user agent MUST initialize the AudioNode this, with context and options as arguments.

Also, let [[curve set]] be an internal slot of this WaveShaperNode. Initialize this slot to false. If options is given and specifies a curve, set [[curve set]] to true.

Arguments for the WaveShaperNode.constructor() 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`.

1.31.2. Attributes

WaveShaperNode/curve In all current engines. Firefox25+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android25+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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. The output is determined by the following algorithm.

Let \(x\) be the input sample, \(y\) be the corresponding output of the node, \(c_k\) be the \(k\)'th element of the curve, and \(N\) be the length of the curve.

Let

$$
  \begin{align*}
  v &= \frac{N-1}{2}(x + 1) \\
  k &= \lfloor v \rfloor \\
  f &= v - k
  \end{align*}
$$

Then

$$
  \begin{align*}
  y &=
    \begin{cases}
    c_0 & v \lt 0 \\
    c_{N-1} & v \ge N - 1 \\
    (1-f)\,c_k + fc_{k+1} & \mathrm{otherwise}
    \end{cases}
  \end{align*}
$$

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.

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.

Note: The use of a curve that produces a non-zero output value for zero input value will cause this node to produce a DC signal even if there are no inputs connected to this node. This will persist until the node is disconnected from downstream nodes.

WaveShaperNode/oversample In all current engines. Firefox26+Safari6+Chrome14+ Opera15+Edge79+ Edge (Legacy)12+IENone Firefox for Android26+iOS SafariYesChrome for Android18+Android WebView37+Samsung Internet1.0+Opera Mobile14+ 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, or 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.

1.31.3. `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";
};

1.31.3.1. 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.

1.32. The `AudioWorklet` Interface

AudioWorklet

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[Exposed=Window, SecureContext]
interface AudioWorklet : Worklet {
};

1.32.1. 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 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.

The AudioWorklet's worklet global scope type is AudioWorkletGlobalScope.

The AudioWorklet's worklet destination type is "audioworklet".

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 constructor and the active instances created from the constructor.

AudioWorklet has one internal slot:

node name to parameter descriptor map which is a map containing an identical set of string keys from node name to processor constructor 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.
    const input = inputs[0];
    const output = outputs[0];
    output[0].set(input[0]);

    // Process only while there are active inputs.
    return false;
  }
};

registerProcessor('bypass-processor', BypassProcessor);

// The main global scope
const context = new AudioContext();
context.audioWorklet.addModule('bypass-processor.js').then(() => {
  const 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.

1.32.2. 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.

Exactly 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 [HTML]. Overriding the default specified in [HTML], AudioWorkletGlobalScopes must not be terminated arbitrarily by the user agent.

An AudioWorkletGlobalScope has the following internal slots:

node name to processor constructor map which is a map that stores key-value pairs of processor name → AudioWorkletProcessorConstructor instance. Initially this map is empty and populated when the registerProcessor() method is called.
pending processor construction data stores temporary data generated by the AudioWorkletNode constructor for the instantiation of the corresponding AudioWorkletProcessor. The pending processor construction data contains the following items:
- node reference which is initially empty. This storage is for an AudioWorkletNode reference that is transferred from the AudioWorkletNode constructor.
- transferred port which is initially empty. This storage is for a deserialized MessagePort that is transferred from the AudioWorkletNode constructor.

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: An 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.

AudioWorkletGlobalScope

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callback AudioWorkletProcessorConstructor = AudioWorkletProcessor (object options);

[Global=(Worklet, AudioWorklet), Exposed=AudioWorklet]
interface AudioWorkletGlobalScope : WorkletGlobalScope {
  undefined registerProcessor (DOMString name,
                               AudioWorkletProcessorConstructor processorCtor);
  readonly attribute unsigned long long currentFrame;
  readonly attribute double currentTime;
  readonly attribute float sampleRate;
};

1.32.2.1. 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.

1.32.2.2. Methods

AudioWorkletGlobalScope/registerProcessor Firefox76+SafariNoneChrome66+ Opera53+Edge79+ Edge (Legacy)NoneIENone Firefox for AndroidNoneiOS SafariNoneChrome for Android66+Android WebView66+Samsung Internet9.0+Opera Mobile47+ registerProcessor(name, processorCtor)

Registers a class constructor derived from AudioWorkletProcessor.

When the registerProcessor(name, processorCtor) method is called, perform the following steps. If an exception is thrown in any step, abort the remaining steps.

If name is an empty string, throw a NotSupportedError.
If name alredy exists as a key in the node name to processor constructor map, throw a NotSupportedError.
If the result of IsConstructor(argument=processorCtor) is false, throw a TypeError .
Let prototype be the result of Get(O=processorCtor, P="prototype").
If the result of Type(argument=prototype) is not Object, throw a TypeError .
Let parameterDescriptorsValue be the result of Get(O=processorCtor, P="parameterDescriptors").
If parameterDescriptorsValue is not undefined, execute the following steps:
1. Let parameterDescriptorSequence be the result of the conversion from parameterDescriptorsValue to an IDL value of type sequence<AudioParamDescriptor>.
2. Let paramNames be an empty Array.
3. For each descriptor of parameterDescriptorSequence:
  1. Let paramName be the value of the member name in descriptor. Throw a NotSupportedError if paramNames already contains paramName value.
  2. Append paramName to the paramNames array.
  3. Let defaultValue be the value of the member defaultValue in descriptor.
  4. Let minValue be the value of the member minValue in descriptor.
  5. Let maxValue be the value of the member maxValue in descriptor.
  6. If the expresstion minValue <= defaultValue <= maxValue is false, throw an InvalidStateError.
Append the key-value pair name → processorCtor to node name to processor constructor map of the associated AudioWorkletGlobalScope.
queue a media element task to append the key-value pair name → parameterDescriptorSequence 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 after registration.

Arguments for the AudioWorkletGlobalScope.registerProcessor(name, processorCtor) method.
Parameter	Type	Nullable	Optional	Description
`name`	DOMString	✘	✘	A string key that represents a class constructor to be registered. This key is used to look up the constructor of `AudioWorkletProcessor` during construction of an `AudioWorkletNode`.
`processorCtor`	AudioWorkletProcessorConstructor	✘	✘	A class constructor extended from `AudioWorkletProcessor`.

Return type: undefined

1.32.2.3. The instantiation of `AudioWorkletProcessor`

At the end of the AudioWorkletNode construction, A struct named processor construction data will be prepared for cross-thread transfer. This struct contains the following items:

name which is a DOMString that is to be looked up in the node name to processor constructor map.
node which is a reference to the AudioWorkletNode created.
options which is a serialized AudioWorkletNodeOptions given to the AudioWorkletNode's constructor.
port which is a serialized MessagePort paired with the AudioWorkletNode's port.

Upon the arrival of the transferred data on the AudioWorkletGlobalScope, the rendering thread will invoke the algorithm below:

Let constructionData be the processor construction data transferred from the control thread.
Let processorName, nodeReference and serializedPort be constructionData’s name, node, and port respectively.
Let serializedOptions be constructionData’s options.
Let deserializedPort be the result of StructuredDeserialize(serializedPort, the current Realm).
Let deserializedOptions be the result of StructuredDeserialize(serializedOptions, the current Realm).
Let processorCtor be the result of looking up processorName on the AudioWorkletGlobalScope's node name to processor constructor map.
Store nodeReference and deserializedPort to node reference and transferred port of this AudioWorkletGlobalScope's pending processor construction data respectively.
Construct a callback function from processorCtor with the argument of deserializedOptions. If any exceptions are thrown in the callback, queue a task to the control thread to fire an event named processorerror using ErrorEvent at nodeReference.
Empty the pending processor construction data slot.

1.32.3. The `AudioWorkletNode` Interface

This interface represents a user-defined AudioNode which lives on the control thread. The user can create an AudioWorkletNode from a 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	See notes	Any tail-time is handled by the node itself

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.

All tasks posted from an AudioWorkletNode are posted to the task queue of its associated BaseAudioContext.

[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.

AudioWorkletNode

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[Exposed=Window, SecureContext]
interface AudioWorkletNode : AudioNode {
  constructor (BaseAudioContext context, DOMString name,
               optional AudioWorkletNodeOptions options = {});
  readonly attribute AudioParamMap parameters;
  readonly attribute MessagePort port;
  attribute EventHandler onprocessorerror;
};

1.32.3.1. Constructors

AudioWorkletNode/AudioWorkletNode Firefox76+SafariNoneChrome66+ Opera?Edge79+ Edge (Legacy)NoneIENone Firefox for Android79+iOS SafariNoneChrome for Android66+Android WebView66+Samsung Internet9.0+Opera Mobile? AudioWorkletNode(context, name, options)

Arguments for the AudioWorkletNode.constructor() method.
Parameter	Type	Nullable	Optional	Description
`context`	BaseAudioContext	✘	✘	The `BaseAudioContext` this new `AudioWorkletNode` will be associated with.
`name`	DOMString	✘	✘	A string that is a key for the `BaseAudioContext`’s node name to parameter descriptor map.
`options`	AudioWorkletNodeOptions	✘	✔	Optional initial parameters value for this `AudioWorkletNode`.

When the constructor is called, the user agent MUST perform the following steps on the control thread:

When the AudioWorkletNode constructor is invoked with context, nodeName, options:

If nodeName does not exist as a key in the BaseAudioContext’s node name to parameter descriptor map, throw a InvalidStateError exception and abort these steps.
Let node be this value.
Initialize the AudioNode node with context and options as arguments.
Configure input, output and output channels of node with options. Abort the remaining steps if any exception is thrown.
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 serializedProcessorPort be the result of StructuredSerializeWithTransfer(processorPortOnThisSide, « processorPortOnThisSide »).
Convert options dictionary to optionsObject.
Let serializedOptions be the result of StructuredSerialize(optionsObject).
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 name member in descriptor.
  2. Let audioParam be a new AudioParam instance with automationRate, defaultValue, minValue, and maxValue having values equal to the values of corresponding members on descriptor.
  3. Append a key-value pair paramName → audioParam to audioParamMap’s entries.
3. If parameterData is present on options, perform the following steps:
  1. Let parameterData be the value of parameterData.
  2. For each paramName → paramValue of parameterData:
    1. If there exists a map entry on audioParamMap with key paramName, let audioParamInMap be such entry.
    2. Set value property of audioParamInMap to paramValue.
4. Set node’s parameters to audioParamMap.
Queue a control message to invoke the constructor of the corresponding AudioWorkletProcessor with the processor construction data that consists of: nodeName, node, serializedOptions, and serializedProcessorPort.

1.32.3.2. Attributes

AudioWorkletNode/onprocessorerror Firefox76+SafariNoneChrome67+ Opera?Edge79+ Edge (Legacy)NoneIENone Firefox for Android79+iOS SafariNoneChrome for Android67+Android WebView67+Samsung Internet9.0+Opera Mobile? onprocessorerror, of type EventHandler

When an unhandled exception is thrown from the processor’s constructor, process method, or any user-defined class method, the processor will queue a media element task to fire an event named processorerror using ErrorEvent at the associated AudioWorkletNode.

The ErrorEvent is created and initialized appropriately with its message, filename, lineno, colno attributes on the control thread.

Note that once a unhandled exception is thrown, the processor will output silence throughout its lifetime.

AudioWorkletNode/parameters Firefox76+SafariNoneChrome67+ OperaYesEdge79+ Edge (Legacy)NoneIENone Firefox for Android79+iOS SafariNoneChrome for Android67+Android WebView67+Samsung Internet9.0+Opera MobileYes 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 constructor at the instantiation.

AudioWorkletNode/port Firefox76+SafariNoneChrome67+ OperaYesEdge79+ Edge (Legacy)NoneIENone Firefox for Android79+iOS SafariNoneChrome for Android67+Android WebView67+Samsung Internet9.0+Opera MobileYes port, of type MessagePort, readonly

Every AudioWorkletNode has an associated port which is the MessagePort. It is connected to the port on the corresponding AudioWorkletProcessor object allowing bidirectional communication between the AudioWorkletNode and its AudioWorkletProcessor.

Note: Authors that register a event listener on the "message" event of this port should call close on either end of the MessageChannel (either in the AudioWorkletProcessor or the AudioWorkletNode side) to allow for resources to be collected.

1.32.3.3. `AudioWorkletNodeOptions`

The AudioWorkletNodeOptions dictionary can be used to initialize attibutes in the instance of an AudioWorkletNode.

dictionary AudioWorkletNodeOptions : AudioNodeOptions {
  unsigned long numberOfInputs = 1;
  unsigned long numberOfOutputs = 1;
  sequence<unsigned long> outputChannelCount;
  record<DOMString, double> parameterData;
  object processorOptions;
};

1.32.3.3.1. Dictionary `AudioWorkletNodeOptions` Members

numberOfInputs, of type unsigned long, defaulting to 1: This is used to initialize the value of the AudioNode numberOfInputs attribute.
numberOfOutputs, of type unsigned long, defaulting to 1: This is used to initialize the value of the AudioNode numberOfOutputs attribute.
outputChannelCount, of type sequence<unsigned long>: This array is used to configure the number of channels in each output.
parameterData, of type record<DOMString, double>: This is a list of user-defined key-value pairs that are used to set the initial value of an AudioParam with the matched name in the AudioWorkletNode.
processorOptions, of type object: This holds any user-defined data that may be used to initialize custom properties in an AudioWorkletProcessor instance that is associated with the AudioWorkletNode.

1.32.3.3.2. Configuring Channels with `AudioWorkletNodeOptions`

The following algorithm describes how an AudioWorkletNodeOptions can be used to configure various channel configurations.

Let node be an AudioWorkletNode instance that is given to this algorithm.
If both numberOfInputs and numberOfOutputs are zero, throw a NotSupportedError and abort the remaining steps.
If outputChannelCount is present,
1. If any value in outputChannelCount is zero or greater than the implementation’s maximum number of channels, throw a NotSupportedError and abort the remaining steps.
2. If the length of outputChannelCount does not equal numberOfOutputs, throw an IndexSizeError and abort the remaining steps.
3. If both numberOfInputs and numberOfOutputs are 1, set the channel count of the node output to the one value in outputChannelCount.
4. Otherwise set the channel count of the kth output of the node to the kth element of outputChannelCount sequence and return.
If outputChannelCount is not present,
1. If both numberOfInputs and numberOfOutputs are 1, set the initial channel count of the node output to 1 and return.
  
  NOTE: For this case, the output chanel count will change to computedNumberOfChannels dynamically based on the input and the channelCountMode at runtime.
2. Otherwise set the channel count of each output of the node to 1 and return.

1.32.4. The `AudioWorkletProcessor` Interface

This interface represents an audio processing code that runs on the audio rendering thread. It lives in the AudioWorkletGlobalScope, and the definition of the class manifests the actual audio processing. Note that the an AudioWorkletProcessor construction can only happen as a result of an AudioWorkletNode contruction.

AudioWorkletProcessor

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[Exposed=AudioWorklet]
interface AudioWorkletProcessor {
  constructor ();
  readonly attribute MessagePort port;
};

callback AudioWorkletProcessCallback =
  boolean (FrozenArray<FrozenArray<Float32Array>> inputs,
           FrozenArray<FrozenArray<Float32Array>> outputs,
           object parameters);

AudioWorkletProcessor has two internal slots:

[[node reference]]: A reference to the associated AudioWorkletNode.
[[callable process]]: A boolean flag representing whether process() is a valid function that can be invoked.

1.32.4.1. Constructors

AudioWorkletProcessor/AudioWorkletProcessor Firefox76+SafariNoneChrome64+ Opera?Edge79+ Edge (Legacy)NoneIENone Firefox for Android79+iOS SafariNoneChrome for Android64+Android WebView64+Samsung Internet9.0+Opera Mobile? AudioWorkletProcessor()

When the constructor for AudioWorkletProcessor is invoked, the following steps are performed on the rendering thread.

Let nodeReference be the result of looking up node reference on the pending processor construction data of the current AudioWorkletGlobalScope. Throw a TypeError exception if the slot is empty.
Let processor be the this value.
Set processor’s [[node reference]] to nodeReference.
Set processor’s [[callable process]] to true.
Let deserializedPort be the result of looking up transferred port from the pending processor construction data.
Set processor’s port to deserializedPort.
Empty the pending processor construction data slot.

1.32.4.2. Attributes

AudioWorkletProcessor/port Firefox76+SafariNoneChrome64+ Opera?Edge79+ Edge (Legacy)NoneIENone Firefox for Android79+iOS SafariNoneChrome for Android64+Android WebView64+Samsung Internet9.0+Opera Mobile? port, of type MessagePort, readonly

Every AudioWorkletProcessor has an associated port which is a MessagePort. It is connected to the port on the corresponding AudioWorkletNode object allowing bidirectional communication between an AudioWorkletNode and its AudioWorkletProcessor.

1.32.4.3. Callback `AudioWorkletProcessCallback`

Users can define a custom audio processor by extending AudioWorkletProcessor. The subclass MUST define an AudioWorkletProcessCallback named process() that implements the audio processing algorithm and may have a static property named parameterDescriptors which is an iterable of AudioParamDescriptors.

The process() callback function is handled as specified when rendering a graph.

The return value of this callback controls the lifetime of the AudioWorkletProcessor's associated AudioWorkletNode.

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 the AudioWorkletNode is actively processing.
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.

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.
    const input = inputs[0];
    const output = outputs[0];
    const 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];
      }
    }
  }
}

1.32.4.3.1. Callback `AudioWorkletProcessCallback` Parameters

The following describes the parameters to the AudioWorkletProcessCallback function.

In general, the inputs and outputs arrays will be reused between calls so that no memory allocation is done. However, if the topology changes, because, say, the number of channels in the input or the output changes, new arrays are reallocated. New arrays are also reallocated if any part of the inputs or outputs arrays are transferred.

inputs, of type FrozenArray<FrozenArray<Float32Array>>

The input audio buffer from the incoming connections provided by the user agent. 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 there are no actively processing AudioNodes connected to the \(n\)th input of the AudioWorkletNode for 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, of type FrozenArray<FrozenArray<Float32Array>>

The output audio buffer that is to be consumed by the user agent. outputs[n][m] is a Float32Array object containing the audio samples for \(m\)th channel of \(n\)th output. Each of the Float32Arrays are zero-filled. The number of channels in the output will match computedNumberOfChannels only when the node has a single output.

parameters, of type object

An ordered map of name → parameterValues. parameters["name"] returns parameterValues, which is a FrozenArray<Float32Array> with the automation values of the name AudioParam.

For each array, the array contains the computedValue of 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.

This object is frozen according the the following steps

Let parameter be the ordered map of the name and parameter values.
SetIntegrityLevel(parameter, frozen)

This frozen ordered map computed in the algorithm is passed to the parameters argument.

Note: This means the object cannot be modified and hence the same object can be used for successive calls unless length of an array changes.

1.32.4.4. `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";
};

1.32.4.4.1. Dictionary `AudioParamDescriptor` Members

There are constraints on the values for these members. See the algorithm for handling an AudioParamDescriptor for the constraints.

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.
maxValue, of type float, defaulting to 3.4028235e38: Represents the maximum value.
minValue, of type float, defaulting to -3.4028235e38: Represents the minimum value.
name, of type DOMString: Represents the name of the parameter.

1.32.5. 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.

1.32.6. AudioWorklet Examples

1.32.6.1. 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.

const context = new AudioContext();context.audioWorklet.addModule('bitcrusher.js').then(() => {  const osc = new OscillatorNode(context);  const 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.  const bitcrusher = new AudioWorkletNode(context, 'bitcrusher', {    parameterData: {bitDepth: 8}  });  osc.connect(bitcrusher).connect(amp).connect(context.destination);  osc.start();});

class Bitcrusher 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) {    // The initial parameter value can be set by passing |options|    // to the processor’s constructor.    super(options);    this._phase = 0;    this._lastSampleValue = 0;  }  process (inputs, outputs, parameters) {    const input = inputs[0];    const output = outputs[0];    const bitDepth = parameters.bitDepth;    const frequencyReduction = parameters.frequencyReduction;    if (bitDepth.length > 1) {      // The bitDepth parameter array has 128 sample values.      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 % frequencyReduction.length];          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.      const 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 % frequencyReduction.length];          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.  }});registerProcessor('bitcrusher', Bitcrusher);

Note: In the definition of AudioWorkletProcessor class, an InvalidStateError will be thrown if the author-supplied constructor has an explicit return value that is not this or does not properly call super().

1.32.6.2. 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.

/* vumeter-node.js: Main global scope */export default class VUMeterNode extends AudioWorkletNode {  constructor (context, updateIntervalInMS) {    super(context, 'vumeter', {      numberOfInputs: 1,      numberOfOutputs: 0,      channelCount: 1,      processorOptions: {        updateIntervalInMS: updateIntervalInMS || 16.67;      }    });    // States in AudioWorkletNode    this._updateIntervalInMS = updateIntervalInMS;    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 updateInterval() {    return this._updateIntervalInMS;  }  set updateInterval(updateIntervalInMS) {    this._updateIntervalInMS = updateIntervalInMS;    this.port.postMessage({updateIntervalInMS: updateIntervalInMS});  }  draw () {    // Draws the VU meter based on the volume value    // every |this._updateIntervalInMS| milliseconds.  }};

/* vumeter-processor.js: AudioWorkletGlobalScope */const SMOOTHING_FACTOR = 0.9;const MINIMUM_VALUE = 0.00001;registerProcessor('vumeter', class extends AudioWorkletProcessor {  constructor (options) {    super();    this._volume = 0;    this._updateIntervalInMS = options.processorOptions.updateIntervalInMS;    this._nextUpdateFrame = this._updateIntervalInMS;    this.port.onmessage = event => {      if (event.data.updateIntervalInMS)        this._updateIntervalInMS = event.data.updateIntervalInMS;    }  }  get intervalInFrames () {    return this._updateIntervalInMS / 1000 * sampleRate;  }  process (inputs, outputs, parameters) {    const input = inputs[0];    // Note that the input will be down-mixed to mono; however, if no inputs are    // connected then zero channels will be passed in.    if (input.length > 0) {      const samples = input[0];      let sum = 0;      let rms = 0;      // Calculated the squared-sum.      for (let i = 0; i < samples.length; ++i)        sum += samples[i] * samples[i];      // Calculate the RMS level and update the volume.      rms = Math.sqrt(sum / samples.length);      this._volume = Math.max(rms, this._volume * SMOOTHING_FACTOR);      // Update and sync the volume property with the main thread.      this._nextUpdateFrame -= samples.length;      if (this._nextUpdateFrame < 0) {        this._nextUpdateFrame += this.intervalInFrames;        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 >= MINIMUM_VALUE;  }});

/* index.js: Main global scope, entry point */import VUMeterNode from './vumeter-node.js';const context = new AudioContext();context.audioWorklet.addModule('vumeter-processor.js').then(() => {  const oscillator = new OscillatorNode(context);  const vuMeterNode = new VUMeterNode(context, 25);  oscillator.connect(vuMeterNode);  oscillator.start();  function drawMeter () {    vuMeterNode.draw();    requestAnimationFrame(drawMeter);  }  drawMeter();});

2. Processing model

2.1. Background

This section is non-normative.

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 a callback is ideally 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. Crossing thread boundaries or adding some 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 computing audio for 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 rendering thread.

Queuing a control message means adding the message to the end of the control message queue of an BaseAudioContext.

Note: For example, successfuly calling start() on an AudioBufferSourceNode source adds a control message to the control message queue of the associated BaseAudioContext.

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.

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 BaseAudioContexts, 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 a BaseAudioContext 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.

An OfflineAudioContext is not required to have a system-level audio callback, but behaves as if it did with the callback happening as soon as the previous callback is finished.

The 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 control message queue, each AudioContext has a regular task queue, called its associated task queue for tasks that are posted to the rendering thread from the control thread. An additional microtask checkpoint is performed after processing a render quantum to run any microtasks that might have been queued during the execution of the process methods of AudioWorkletProcessor.

All tasks posted from an AudioWorkletNode are posted to the associated task queue of its associated BaseAudioContext.

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.

The following steps MUST be performed when rendering a render quantum.

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 the BaseAudioContext's associated task queue.
1. Let task queue be the BaseAudioContext's associated task queue.
2. Let task count be the number of tasks in the in task queue
3. While task count is not equal to 0, execute the following steps:
  1. Let oldest task be the first runnable task in task queue, and remove it from task queue.
  2. Set the rendering loop’s currently running task to oldest task.
  3. Perform oldest task’s steps.
  4. Set the rendering loop currently running task back to null.
  5. Decrement task count
  6. Perform a microtask checkpoint.
Process a render quantum.
1. If the [[rendering thread state]] of the BaseAudioContext is not running, return false.
2. Order the AudioNodes of the BaseAudioContext to be processed.
  1. Let ordered node list be an empty list of AudioNodes and AudioListener. It will contain an ordered list of AudioNodes and the AudioListener when this ordering algorithm terminates.
  2. Let nodes be the set of all nodes created by this BaseAudioContext, and still alive.
  3. Add the AudioListener to nodes.
  4. Let cycle breakers be an empty set of DelayNodes. It will contain all the DelayNodes that are part of a cycle.
  5. 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.
  6. For each DelayNode delay in cycle breakers:
    1. Let delayWriter and delayReader respectively be a DelayWriter and a DelayReader, for delay. Add delayWriter and delayReader to nodes. Disconnect delay from all its input and outputs.
      
      Note: This breaks the cycle: if a DelayNode is in a cycle, its two ends can be considered separately, because delay lines cannot be smaller than one render quantum when in a cycle.
  7. If nodes contains cycles, mute all the AudioNodes that are part of this cycle, and remove them from nodes.
  8. Consider all elements in nodes to be unmarked. While there are unmarked elements in nodes:
    1. Choose an element 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. If node is an AudioNode, Visit each AudioNode connected to the input of node.
    4. For each AudioParam param of node:
      
      For each AudioNode param input node connected to param:
      
      Visit param input node
    5. Add node to the beginning of ordered node list.
  9. Reverse the order of ordered node list.
3. Compute the value(s) of the AudioListener's AudioParams for this block.
4. For each AudioNode, in ordered node list:
  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 according to § 1.6.3 Computation of Value.
  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, execute these substeps:
    1. Let processor be the associated AudioWorkletProcessor instance of AudioWorkletNode.
    2. Let O be the ECMAScript object corresponding to processor.
    3. Let processCallback be an uninitialized variable.
    4. Let completion be an uninitialized variable.
    5. Prepare to run script with the current settings object.
    6. Prepare to run a callback with the current settings object.
    7. Let getResult be Get(O, "process").
    8. If getResult is an abrupt completion, set completion to getResult and jump to the step labeled return.
    9. Set processCallback to getResult.[[Value]].
    10. If ! IsCallable(processCallback) is false, then:
      1. Set completion to new Completion {[[Type]]: throw, [[Value]]: a newly created TypeError object, [[Target]]: empty}.
      2. Jump to the step labeled return.
    11. Set [[callable process]] to true.
    12. Perform the following substeps:
      1. Let args be a Web IDL arguments list consisting of inputs, outputs, and parameters.
      2. Let esArgs be the result of converting args to an ECMAScript arguments list.
      3. Let callResult be the Call(processCallback, O, esArgs). This operation computes a block of audio with esArgs. Upon a successful function call, a buffer containing copies of the elements of the Float32Arrays passed via the outputs is made available for reading. Any Promise resolved within this call will be queued into the microtask queue in the AudioWorkletGlobalScope.
      4. If callResult is an abrupt completion, set completion to callResult and jump to the step labeled return.
      5. Set processor’s active source flag to ToBoolean(callResult.[[Value]]).
    13. Return: at this point completion will be set to an ECMAScript completion value.
      1. Clean up after running a callback with the current settings object.
      2. Clean up after running script with the current settings object.
      3. If completion is an abrupt completion:
        
        Set [[callable process]] to false.
        
        Set processor’s active source flag to false.
        
        Make a silent output buffer available for reading.
        
        Queue a task to the control thread fire an ErrorEvent named processorerror at the associated AudioWorkletNode.
  5. 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.
5. Atomically perform the following steps:
  1. Increment [[current frame]] by the render quantum size.
  2. Set currentTime to [[current frame]] divided by sampleRate.
6. 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.

2.5. Unloading a document

Additional unloading document cleanup steps are defined for documents that use BaseAudioContext:

Reject all the promises of [[pending promises]] with InvalidStateError, for each AudioContext and OfflineAudioContext whose relevant global object is the same as the document’s associated Window.
Stop all decoding threads.
Queue a control message to close() the AudioContext or OfflineAudioContext.

3. Dynamic Lifetime

3.1. Background

Note: The normative description of AudioContext and AudioNode lifetime characteristics is described by the AudioContext lifetime and AudioNode lifetime.

This section is non-normative.

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.

3.2. 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:

let context = 0;let compressor = 0;let gainNode1 = 0;let 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.    const 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() {    const oneShotSound = context.createBufferSource();    oneShotSound.buffer = dogBarkingBuffer;    // Create a filter, panner, and gain node.    const lowpass = context.createBiquadFilter();    const panner = context.createPanner();    const 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);}

4. Channel Up-Mixing and Down-Mixing

This section is normative.

An AudioNode 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 summing together of each of the corresponding channels that have been up-mixed or down-mixed in step 1 for each connection.

4.1. 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.

4.2. 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

4.3. Implication of tail-time on input and output channel count

When an AudioNode has a non-zero tail-time, and an output channel count that depends on the input channels count, the AudioNode's tail-time must be taken into account when the input channel count changes.

When there is a decrease in input channel count, the change in output channel count MUST happen when the input that was received with greater channel count no longer affects the output.

When there is an increase in input channel count, the behavior depends on the AudioNode type:

For a DelayNode or a DynamicsCompressorNode, the number of output channels MUST increase when the input that was received with greater channel count begins to affect the output.
For other AudioNodes that have a tail-time, the number of output channels MUST increase immediately.

Note: For a ConvolverNode, this only applies to the case where the impulse response is mono. Otherwise, the ConvolverNode always outputs a stereo signal regardless of its input channel count.

Note: Intuitively, this allows not losing stereo information as part of processing: when multiple input render quanta of different channel count contribute to an output render quantum then the output render quantum’s channel count is a superset of the input channel count of the input render quantums.

4.4. 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;

4.5. 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

4.6. 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";

5. Audio Signal Values

5.1. Audio sample format

Linear pulse code modulation (linear PCM) describes a format where the audio values are sampled at a regular interval, and where the quantization levels between two successive values are linearly uniform.

Whenever signal values are exposed to script in this specification, they are in linear 32-bit floating point pulse code modulation format (linear 32-bit float PCM), often in the form of Float32Array objects.

5.2. Rendering

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.

6. Spatialization/Panning

6.1. Background

A common feature requirement for modern 3D games is the ability to dynamically spatialize and move multiple audio sources in 3D space. For example OpenAL has this ability.

Using a PannerNode, an audio stream can be spatialized or positioned in space relative to an AudioListener. A BaseAudioContext 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.

The coordinate system for spatialization is shown in the diagram below, with the default values shown. The locations for the AudioListener and PannerNode are moved from the default positions so we can see things better.

panner-coord — Diagram of the coordinate system with AudioListener and PannerNode attributes shown.

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.

6.2. Azimuth and Elevation

The following algorithm MUST be used to calculate the azimuth and elevation for the PannerNode. The implementation must appropriately account for whether the various AudioParams below are "a-rate" or "k-rate".

// Let |context| be a BaseAudioContext and let |panner| be a// PannerNode created in |context|.// Calculate the source-listener vector.const listener = context.listener;const sourcePosition = new Vec3(panner.positionX.value, panner.positionY.value,                                panner.positionZ.value);const listenerPosition =    new Vec3(listener.positionX.value, listener.positionY.value,             listener.positionZ.value);const 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.const listenerForward = new Vec3(listener.forwardX.value, listener.forwardY.value,                                 listener.forwardZ.value);const listenerUp =    new Vec3(listener.upX.value, listener.upY.value, listener.upZ.value);const 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, forwardconst listenerRightNorm = listenerRight.normalize();const listenerForwardNorm = listenerForward.normalize();const up = listenerRightNorm.cross(listenerForwardNorm);const upProjection = sourceListener.dot(up);const projectedSource = sourceListener.diff(up.scale(upProjection)).normalize();azimuth = 180 * Math.acos(projectedSource.dot(listenerRightNorm)) / Math.PI;// Source in front or behind the listener.const frontBack = projectedSource.dot(listenerForwardNorm);if (frontBack < 0)  azimuth = 360 - azimuth;// Make azimuth relative to "forward" 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;

6.3. 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.

6.3.1. 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 the appropriate rate as specified by the automationRate. If any of the PannerNode's AudioParams or the AudioListener's AudioParams are "a-rate", a-rate processing must be used.

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;

6.3.2. 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.

6.3.3. 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;
}

6.4. 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) {  const pannerPosition = new Vec3(panner.positionX.value, panner.positionY.value,                                  panner.positionZ.value);  const listener = context.listener;  const 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.

As part of its processing, the PannerNode scales/multiplies the input audio signal by distanceGain to make distant sounds quieter and nearer ones louder.

6.5. 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 diagram ilustrates the relationship between the source’s cone with respect to the listener. In the diagram, coneInnerAngle = 50 and coneOuterAngle = 120. That is, the inner cone extends 25 deg on each side of the direction vector. Similarly, the outer cone is 60 deg on each side.

cone-diagram — Cone angles for a source in relationship to the source orientation and the listeners position and orientation.

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() {  const 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  const sourcePosition = new Vec3(panner.positionX.value, panner.positionY.value,                                  panner.positionZ.value);  const listenerPosition =      new Vec3(listener.positionX.value, listener.positionY.value,               listener.positionZ.value);  const sourceToListener = sourcePosition.diff(listenerPosition).normalize();  const normalizedSourceOrientation = sourceOrientation.normalize();  // Angle between the source orientation vector and the source-listener vector  const angle = 180 *                Math.acos(sourceToListener.dot(normalizedSourceOrientation)) /                Math.PI;  const absAngle = Math.abs(angle);  // Divide by 2 here since API is entire angle (not half-angle)  const absInnerAngle = Math.abs(source.coneInnerAngle) / 2;  const 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    const x = (absAngle - absInnerAngle) / (absOuterAngle - absInnerAngle);    gain = (1 - x) + source.coneOuterGain * x;  }  return gain;}

7. Performance Considerations

7.1. 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.

7.2. 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.

7.3. 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.

7.4. 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.

8. Security and Privacy Considerations

Per the Self-Review Questionnaire: Security and Privacy §questions:

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. See AudioContext.
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 compiler flags used and also the CPU architecture (ARM vs. x86) contribute to this entropy.

In practice however, this merely allows deduction of information already readily available by easier means (User Agent string), such as "this is browser X running on platform Y". However, to reduce the possibility of additional fingerprinting, we mandate browsers take action to mitigate fingerprinting issues that might be possible from the output of any node.

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 High-Resolution Time §7 Privacy and Security section 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.

Fingerprinting via the sample rate of the AudioContext is also possible. We recommend the following steps to be taken to minimize this:
1. 44.1 kHz and 48 kHz are allowed as default rates; the system will choose between them for best applicability. (Obviously, if the audio device is natively 44.1, 44.1 will be chosen, etc., but also the system may choose the most "compatible" rate—e.g. if the system is natively 96kHz, 48kHz would likely be chosen, not 44.1kHz.
2. The system should resample to one of those two rates for devices that are natively at different rates, despite the fact that this may cause extra battery drain due to resampled audio. (Again, the system will choose the most compatible rate—e.g. if the native system is 16kHz, it’s expected that 48kHz would be chosen.)
3. It is expected (though not mandated) that browsers would offer a user affordance to force use of the native rate—e.g. by setting a flag in the browser on the device. This setting would not be exposed in the API.
4. It is also expected behavior that a different rate could be explicitly requested in the constructor for AudioContext (this is already in the specification; it normally causes the audio rendering to be done at the requested sampleRate, and then up- or down-sampled to the device output), and if that rate is natively supported, the rendering could be passed straight through. This would enable apps to render to higher rates without user intervention (although it’s not observable from Web Audio that the audio output is not downsampled on output)—for example, if MediaDevices capabilities were read (with user intervention) and indicated a higher rate was supported.
Fingerprinting via the number of output channels for the AudioContext is possible as well. We recommend that maxChannelCount be set to two (stereo). Stereo is by far the most common number of channels.
Does this specification enable new script execution/loading mechanisms?

No. It does use the [HTML] 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.

Additionally, the security and privacy considerations from the Media Capture and Streams specification should be noted. In particular, analysis of ambient audio or playing unique audio may enable identification of user location down to the level of a room or even simultaneous occupation of a room by disparate users or devices. Access to both audio output and audio input might also enable communication between otherwise partitioned contexts in one browser.
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 user’s 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 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?

Not 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.

9. Requirements and Use Cases

Please see [webaudio-usecases].

10. 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() {    const m = magnitude;    if (m == 0) {      return new Vec3(0, 0, 0);    }    return scale(1 / m);  }}

11. Change Log

11.1. Since Proposed Recommendation of 6 May 2021

Styling, status and boilerplate updates for Recommendation
Updated links to RFC2119, High-Resolution Time, and Audio EQ Cookbook
A spelling error was corrected

11.2. Since Candidate Recommendation Snapshot of 14 January 2021

PR 2333: Update links to point to W3C versions
PR 2334: Use bikeshed to link to ErrorEvent
PR 2331: Add MIMESniff to normative references
PR 2328: MediaStream must be resampled to match the context sample rate
PR 2318: Restore empty of pending processor construction data after successful initialization of AudioWorkProcessor#port.
PR 2317: Standardize h3/h4 interface and dictionary markup
PR 2312: Rework description of control thread state and rendering thread state
PR 2311: Adjust the steps to process a context’s regular task queue
PR 2310: OscillatorNode output is mono
PR 2308: Refine phrasing for "allowed to start"
PR 2307: Replace "queue a task" with "queue a media element task"
PR 2306: Move some steps from AudioWorkletProcessor constructor to the instantiation algorithm
PR 2304: Add required components for ES operations in the rendering loop
PR 2301: Define when and how regular tasks are processed wrt the processing model
PR 2286: Clean up ABSN start algorithm
PR 2277: Fix compression curve diagram
PR 2273: Clarify units used in threshold & knee value calculations
PR 2256: ABSN extrapolates the last output
PR 2250: Use FrozenArray for AudioWorkletProcessor process()
PR 2298: Bikeshed HTML validation issues

11.3. Since Candidate Recommendation of 11 June 2020

PR 2202: Fixed wrong optionality of IIRFilterNode options
Issue 2191: Restrict sounds beyond normal hearing
PR 2210: Return rejected promise when the document is not fully active, for operations returning promises
Issue 2191: Destination of request created by addModule
Issue 2213: The message queue is for message running on the rendering thread.
Issue 2216: Use inclusive language in the spec
PR 2219: Update more terminology in images and markdown documents
Issue 2206: PannerNode.rollOffFactor with "linear" distance model is not clamped to [0, 1] in main browser engines
Issue 2169: AudioParamDescriptor has member constraints that are redundant
Issue 1457: [privacy] Exposing data to an origin: fingerprinting
Issue 2061: Privacy re-review of latest changes
Issue 2225: Describe "Planar versus interleaved buffers"
Issue 2231: WaveShaper [[curve set]] not defined
Issue 2240: Align with Web IDL specification
Issue 2242: LInk to undefined instead of using <code>
Issue 2227: Clarify buffer.copyToChannel() must be called before source.buffer = buffer else nothing is played
PR 2253: Fix duplicated IDs for decode callbacks
Issue 2252: When are promises in "[[pending resume promises]]" resolved?
PR 2266: Prohibit arbitrary termination of AudioWorkletGlobalScopes

11.4. Since Candidate Recommendation of 18 September 2018

Issue 2193: Incorrect azimuth comparison in spatialization algorithm
Issue 2192: Waveshaper curve interpolation algorithm incorrect
Issue 2171: Allow not having get parameterDescriptors in an AudioWorkletProcessor
Issue 2184: PannerNode refDistance description unclear
Issue 2165: AudioScheduledSourceNode start algorithm incomplete
Issue 2155: Restore changes accidentally reverted in bikeshed conversion
Issue 2154: Exception for changing channelCountMode on ScriptProcessorNode does not match browsers
Issue 2153: Exception for changing channelCount on ScriptProcessorNode does not match browsers
Issue 2152: close() steps don’t make sense
Issue 2150: AudioBufferOptions requires throwing NotFoundError in cases that can’t happen
Issue 2149: MediaStreamAudioSourceNode constructor has weird check for AudioContext
Issue 2148: IIRFilterOptions description makes impossible demands
Issue 2147: PeriodicWave constructor examines lengths of things that might not be there
Issue 2113: BiquadFilter gain lower bound can be lower.
Issue 2096: Lifetime of pending processor construction data and exceptions in instantiation of AudioWorkletProcessor
Issue 2087: Minor issues with BiquadFilter AudioParams
Issue 2083: Missing text in WaveShaperNode?
Issue 2082: WaveShaperNode curve interpolation incomplete
Issue 2074: Should the AudioWorkletNode constructor invoke the algorithm for initializing an object that inherits from AudioNode?
Issue 2073: Inconsistencies in constructor descriptions and factory method initialization
Issue 2072: Clarification on AudioBufferSourceNode looping, and loop points
Issue 2071: cancelScheduledValues with setValueCurveAtTime
Issue 2060: Would it be helpful to restrict use of AudioWorkletProcessor.port().postMessage() in order to facilitate garbage collection?
Issue 2051: Update to constructor operations
Issue 2050: Restore ConvolverNode channel mixing configurability (up to 2 channels)
Issue 2045: Should the check on process() be removed from AudioWorkletGlobalScope.registerProcessor()?
Issue 2044: Remove options parameter from AudioWorkletProcessor constructor WebIDL
Issue 2036: Remove options parameter of AudioWorkletProcessor constructor
Issue 2035: De-duplicate initial value setting on AudioWorkletNode AudioParams
Issue 2027: Revise "processor construction data" algorithm
Issue 2021: AudioWorkletProcessor constructor leads to infinite recursion
Issue 2018: There are still issues with the setup of an AudioWorkletNode’s parameters
Issue 2016: Clarify parameters in AudioWorkletProcessor.process()
Issue 2011: AudioWorkletNodeOptions.processorOptions should not default to null.
Issue 1989: Please update to Web IDL changes to optional dictionary defaulting
Issue 1984: Handling of exceptions in audio worklet is not very clear
Issue 1976: AudioWorkletProcessor’s [[node reference]] seems to be write-only
Issue 1972: parameterDescriptors handling during AudioWorkletNode initialization is probably wrong
Issue 1971: AudioWorkletNode options serialization is underdefined
Issue 1970: "active source" flag handling is a weird monkeypatch
Issue 1969: It would be clearer if the various validation of AudioWorkletNodeOptions were an explicit step or set of steps
Issue 1966: parameterDescriptors is not looked up by the AudioWorkletProcessor constructor
Issue 1963: NewTarget check for AudioWorkletProcessor isn’t actually possible with a Web IDL constructor
Issue 1947: Spec is inconsistent about whether parameterDescriptors is an array or an iterable
Issue 1946: Population of "node name to parameter descriptor map" needs to be defined
Issue 1945: registerProcessor is doing odd things with threads and JS values
Issue 1943: Describe how WaveShaperNode shapes the input with the curve
Issue 1935: length of AudioWorkletProcessor.process() parameter sequences with inactive inputs
Issue 1932: Make AudioWorkletNode output buffer available for reading
Issue 1925: front vs forward
Issue 1902: Mixer Gain Structure section not needed
Issue 1906: Steps in rendering algorithm
Issue 1905: Rendering callbacks are observable
Issue 1904: Strange Note in algorithm for swapping a control message queue
Issue 1903: Funny sentence about priority and latency
Issue 1901: AudioWorkletNode state property?
Issue 1900: AudioWorkletProcessor NewTarget undefined
Issue 1899: Missing synchronous markers
Issue 1897: WaveShaper curve value setter allows multiple sets
Issue 1896: WaveShaperNode constructor says curve set is initialized to false
Issue #1471: AudioNode Lifetime section seems to attempt to make garbage collection observable
Issue #1893: Active processing for Panner/Convolver/ChannelMerger
Issue #1894: Funny text in PannerNode.orientationX
Issue #1866: References to garbage collection
Issue #1851: Parameter values used for BiquadFilterNode::getFrequencyResponse
Issue #1905: Rendering callbacks are observable
Issue #1879: ABSN playback algorithm offset
Issue #1882: Biquad lowpass/highpass Q
Issue #1303: MediaElementAudioSourceNode information in a funny place
Issue #1896: WaveShaperNode constructor says curve set is initialized to false
Issue #1897: WaveShaper curve value setter allows multiple sets.
Issue #1880: setOrientation description has confusing paragraph
Issue #1855: createScriptProcessor parameter requirements
Issue #1857: Fix typos and bad phrasing
Issue #1788: Unclear what value is returned by AudioParam.value
Issue #1852: Fix error condition of AudioNode.disconnect(destinationNode, output, input)
Issue #1841: Recovering from unstable biquad filters?
Issue #1777: Picture of the coordinate system for panner node
Issue #1802: Clarify interaction between user-invoked suspend and autoplay policy
Issue #1822: OfflineAudioContext.suspend can suspend before the given time
Issue #1772: Sorting tracks alphabetically is underspecified
Issue #1797: Specification is incomplete for AudioNode.connect()
Issue #1805: Exception ordering on error
Issue #1790: Automation example chart has an error (reversed function arguments
Fix rendering algorithm iteration and cycle breaking
Issue #1719: channel count changes in filter nodes with tail time
Issue #1563: Make decodeAudioData more precise
Issue #1481: Tighten spec on ABSN output channels?
Issue #1762: Setting convolver buffer more than once?
Issue #1758: Explicitly include time-domain processing code for BiquadFilterNode
Issue #1770: Link to correct algorithm for StereoPannerNode, mention algorithm is equal-power
Issue #1753: Have a single AudioWorkletGlobalScope per BaseAudioContext
Issue #1746: AnalyserNode: Clarify how much time domain data we’re supposed to keep around
Issue #1741: Sample rate of AudioBuffer
Issue #1745: Clarify unit of fftSize
Issue #1743: Missing normative reference to Fetch
Use "get a reference to the bytes" algorithm as needed.
Specify rules for determining output chanel count.
Clarified rendering algorithm for AudioListener.

11.5. 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.

11.6. 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.

12. 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); Becker, Steven (Microsoft Corporation); Berkovitz, Joe (Invited Expert, affiliated with Noteflight/Hal Leonard) - WG co-chair from September 2013 to December 2017); Bossart, Pierre (Intel Corporation); Borins, Myles (Google, Inc); Buffa, Michel (NSAU); Caceres, Marcos (Invited Expert); Cardoso, Gabriel (INRIA); Carlson, Eric (Apple, Inc); Chen, Bin (Baidu, Inc); Choi, Hongchan (Google, Inc) - Specification Co-editor; Collichio, Lisa (Qualcomm); Geelnard, Marcus (Opera Software); Gehring, Todd (Dolby Laboratories); Goode, Adam (Google, Inc); Gregan, Matthew (Mozilla Foundation); Hikawa, Kazuo (AMEI); Hofmann, Bill (Dolby Laboratories); Jägenstedt, Philip (Google, Inc); Jeong, Paul Changjin (HTML5 Converged Technology Forum); Kalliokoski, Jussi (Invited Expert); Lee, WonSuk (Electronics and Telecommunications Research Institute); Kakishita, Masahiro (AMEI); Kawai, Ryoya (AMEI); Kostiainen, Anssi (Intel Corporation); 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); Nair, Varun (Facebook); Needham, Chris (British Broadcasting Corporation); Noble, Jer (Apple, Inc); O’Callahan, Robert(Mozilla Foundation); Onumonu, Anthony (British Broadcasting Corporation); Paradis, Matthew (British Broadcasting Corporation) - WG co-chair from September 2013 to present; Pozdnyakov, Mikhail (Intel Corporation); Raman, T.V. (Google, Inc); Rogers, Chris (Google, Inc); Schepers, Doug (W3C/MIT); Schmitz, Alexander (JS Foundation); Shires, Glen (Google, Inc); Smith, Jerry (Microsoft Corporation); Smith, Michael (W3C/Keio); Thereaux, Olivier (British Broadcasting Corporation); Toy, Raymond (Google, Inc.) - WG co-chair from December 2017 - Present; Toyoshima, Takashi (Google, Inc); Troncy, Raphael (Institut Telecom); Verdie, Jean-Charles (MStar Semiconductor, Inc.); Wei, James (Intel Corporation); Weitnauer, Michael (IRT); Wilson, Chris (Google,Inc); Zergaoui, Mohamed (INNOVIMAX)

Web Audio API

W3C Recommendation, 17 June 2021

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. AudioNode Tail-Time

1.5.3. AudioNode Lifetime

1.5.4. Attributes

1.5.5. Methods

1.5.6. AudioNodeOptions

1.5.6.1. Dictionary AudioNodeOptions Members

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

1.11.3. Processing

1.12. The AudioProcessingEvent Interface - DEPRECATED

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

1.5.6. `AudioNodeOptions`

1.5.6.1. 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. `IIRFilterOptions`

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`