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:

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

masterDry

;






let

masterWet

;










function

setupRoutingGraph

()


{





context

=


new

AudioContext

();










//
Create
the
effects
nodes.





lowpassFilter

=

context
.

createBiquadFilter

();





waveShaper

=

context
.

createWaveShaper

();





panner

=

context
.

createPanner

();





compressor

=

context
.

createDynamicsCompressor

();





reverb

=

context
.

createConvolver

();










//
Create
master
wet
and
dry.





masterDry

=

context
.

createGain

();





masterWet

=

context
.

createGain

();










//
Connect
final
compressor
to
final
destination.





compressor
.

connect

(

context
.

destination

);










//
Connect
master
dry
and
wet
to
compressor.





masterDry
.

connect

(

compressor

);





masterWet
.

connect

(

compressor

);










//
Connect
reverb
to
master
wet.





reverb
.

connect

(

masterWet

);










//
Create
a
few
sources.





source1

=

context
.

createBufferSource

();





source2

=

context
.

createBufferSource

();





source3

=

context
.

createOscillator

();









source1
.

buffer

=

manTalkingBuffer

;





source2
.

buffer

=

footstepsBuffer

;





source3
.

frequency
.

value

=


440


;










//
Connect
source1





dry1

=

context
.

createGain

();





wet1

=

context
.

createGain

();





source1
.

connect

(

lowpassFilter

);





lowpassFilter
.

connect

(

dry1

);





lowpassFilter
.

connect

(

wet1

);





dry1
.

connect

(

masterDry

);





wet1
.

connect

(

reverb

);










//
Connect
source2





dry2

=

context
.

createGain

();





wet2

=

context
.

createGain

();





source2
.

connect

(

waveShaper

);





waveShaper
.

connect

(

dry2

);





waveShaper
.

connect

(

wet2

);





dry2
.

connect

(

masterDry

);





wet2
.

connect

(

reverb

);










//
Connect
source3





dry3

=

context
.

createGain

();





wet3

=

context
.

createGain

();





source3
.

connect

(

panner

);





panner
.

connect

(

dry3

);





panner
.

connect

(

wet3

);





dry3
.

connect

(

masterDry

);





wet3
.

connect

(

reverb

);










//
Start
the
sources
now.





source1
.

start

(


0


);





source2
.

start

(


0


);





source3
.

start

(


0


);






}

Modular routing also permits the output of AudioNode s 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.

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 AudioNode s.

• An AudioNode interface, which represents audio sources, audio outputs, and intermediate processing modules. AudioNode s can be dynamically connected together in a modular fashion . AudioNode s 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

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.

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

callback DecodeErrorCallback = void (DOMException error);
callback DecodeSuccessCallback = void (AudioBuffer decodedData);
[Exposed=Window]
interface BaseAudioContext : EventTarget {
  readonly attribute AudioDestinationNode destination;
  readonly attribute float sampleRate;
  readonly attribute double currentTime;
  readonly attribute AudioListener listener;
  readonly attribute AudioContextState state;
  [SameObject, SecureContext]
  readonly attribute AudioWorklet audioWorklet;
  attribute EventHandler onstatechange;
  AnalyserNode createAnalyser ();
  BiquadFilterNode createBiquadFilter ();
  AudioBuffer createBuffer (unsigned long numberOfChannels,
                            unsigned long length,
                            float sampleRate);
  AudioBufferSourceNode createBufferSource ();
  ChannelMergerNode createChannelMerger (optional unsigned long numberOfInputs = 6);
  ChannelSplitterNode createChannelSplitter (
    optional unsigned long numberOfOutputs = 6);
  ConstantSourceNode createConstantSource ();
  ConvolverNode createConvolver ();
  DelayNode createDelay (optional double maxDelayTime = 1.0);
  DynamicsCompressorNode createDynamicsCompressor ();
  GainNode createGain ();
  IIRFilterNode createIIRFilter (sequence<double> feedforward,
                                 sequence<double> feedback);
  OscillatorNode createOscillator ();
  PannerNode createPanner ();
  PeriodicWave createPeriodicWave (sequence<float> real,
                                   sequence<float> imag,
                                   optional PeriodicWaveConstraints constraints = {});
  ScriptProcessorNode createScriptProcessor(
    optional unsigned long bufferSize = 0,
    optional unsigned long numberOfInputChannels = 2,
    optional unsigned long numberOfOutputChannels = 2);
  StereoPannerNode createStereoPanner ();
  WaveShaperNode createWaveShaper ();
  Promise<AudioBuffer> decodeAudioData (
    ArrayBuffer audioData,
    optional DecodeSuccessCallback? successCallback,
    optional DecodeErrorCallback? errorCallback);
};

1.1.1. Attributes

audioWorklet , of type AudioWorklet , readonly

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

currentTime , of type double , readonly

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

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

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

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

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

destination , of type AudioDestinationNode , readonly

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

listener , of type AudioListener , readonly

An AudioListener which is used for 3D spatialization .

onstatechange , of type EventHandler

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

sampleRate , of type float , readonly

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

state , of type AudioContextState , readonly

Describes the current state of the AudioContext . Its value is identical to control thread state .

1.1.2. Methods

createAnalyser()

Factory method for an AnalyserNode .

No parameters.

Return type: AnalyserNode

createBiquadFilter()

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

No parameters.

Return type: BiquadFilterNode

createBuffer(numberOfChannels, length, sampleRate)

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

Arguments for the BaseAudioContext.createBuffer() method.

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

Return type: AudioBuffer

createBufferSource()

Factory method for a AudioBufferSourceNode .

No parameters.

Return type: AudioBufferSourceNode

createChannelMerger(numberOfInputs)

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

Arguments for the BaseAudioContext.createChannelMerger(numberOfInputs) method.

Parameter	Type	Nullable	Optional	Description
numberOfInputs	unsigned long	✘	✔	Determines the number of inputs. Values of up to 32 MUST be supported. If not specified, then 6 will be used.

Return type: ChannelMergerNode

createChannelSplitter(numberOfOutputs)

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

Arguments for the BaseAudioContext.createChannelSplitter(numberOfOutputs) method.

Parameter	Type	Nullable	Optional	Description
numberOfOutputs	unsigned long	✘	✔	The number of outputs. Values of up to 32 MUST be supported. If not specified, then 6 will be used.

Return type: ChannelSplitterNode

createConstantSource()

Factory method for a ConstantSourceNode .

No parameters.

Return type: ConstantSourceNode

createConvolver()

Factory method for a ConvolverNode .

No parameters.

Return type: ConvolverNode

createDelay(maxDelayTime)

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

Arguments for the BaseAudioContext.createDelay(maxDelayTime) method.

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

Return type: DelayNode

createDynamicsCompressor()

Factory method for a DynamicsCompressorNode .

No parameters.

Return type: DynamicsCompressorNode

createGain()

Factory method for GainNode .

No parameters.

Return type: GainNode

createIIRFilter(feedforward, feedback)

Arguments for the BaseAudioContext.createIIRFilter() method.

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

Return type: IIRFilterNode

createOscillator()

Factory method for an OscillatorNode .

No parameters.

Return type: OscillatorNode

createPanner()

Factory method for a PannerNode .

No parameters.

Return type: PannerNode

createPeriodicWave(real, imag, constraints)

Factory method to create a PeriodicWave .

When calling this method, execute these steps:

1. If real and imag are not of the same length, an IndexSizeError MUST be thrown.

2. Let o be a new object of type PeriodicWaveOptions .

3. Respectively set the real and imag parameters passed to this factory method to the attributes of the same name on o .

4. Set the disableNormalization attribute on o to the value of the disableNormalization attribute of the constraints attribute passed to the factory method.

5. Construct a new PeriodicWave p , passing the BaseAudioContext this factory method has been called on as a first argument, and o .

6. 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

createStereoPanner()

Factory method for a StereoPannerNode .

No parameters.

Return type: StereoPannerNode

createWaveShaper()

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

No parameters.

Return type: WaveShaperNode

decodeAudioData(audioData, successCallback, errorCallback)

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

Although the primary method of interfacing with this function is via its promise return value, the callback parameters are provided for legacy reasons.

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

1. Let promise be a new Promise.

2. 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.

3. 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 task to invoke errorCallback with error .

4. 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 thread s can run in parallel to service multiple calls to decodeAudioData .

1. Let can decode be a boolean flag, initially set to true.

2. 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 .

3. If can decode is true , attempt to decode the encoded audioData into linear PCM . In case of failure, set can decode to false .

4. If can decode is false , queue a task to execute the following step, on the control thread ’s event loop:

   1. Let error be a DOMException whose name is EncodingError .

      2. Reject promise with error , and remove it from [[pending promises]] .

   2. If errorCallback is not missing, invoke errorCallback with error .

5. Otherwise:

   1. Take the result, representing the decoded linear PCM audio data, and resample it to the sample-rate of the AudioContext if it is different from the sample-rate of audioData .

   2. Queue a task on the control thread ’s event loop to execute the following steps:

      1. Let buffer be an AudioBuffer containing the final result (after possibly 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 AudioContext s, after which any attempt to create a new AudioContext will fail, throwing NotSupportedError .

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

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

1.2. The AudioContext Interface

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

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

[Exposed=Window]
interface AudioContext : BaseAudioContext {
  constructor (optional AudioContextOptions contextOptions = {});
  readonly attribute double baseLatency;
  readonly attribute double outputLatency;
  AudioTimestamp getOutputTimestamp ();
  Promise<void> resume ();
  Promise<void> suspend ();
  Promise<void> 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 delay this initial transition, 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(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:

1. Set a control thread state to suspended on the AudioContext .

2. Set a rendering thread state to suspended on the AudioContext .

3. Let [[pending resume promises]] be a slot on this AudioContext , that is an initially empty ordered list of promises.

4. 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.

5. If the context is allowed to start , send a control message to start processing.

6. Return this AudioContext object.

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

1. Attempt to acquire system resources . In case of failure, abort the following steps.

2. Set the rendering thread state to running on the AudioContext .

3. Queue a task on the control thread event loop, to execute these steps:

   1. Set the state attribute of the AudioContext to " running ".

   2. Queue a task to fire a simple event named statechange at the AudioContext .

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

Arguments for the AudioContext.constructor(contextOptions) method.

Parameter	Type	Nullable	Optional	Description
contextOptions	AudioContextOptions	✘	✔	User-specified options controlling how the AudioContext should be constructed.

1.2.2. Attributes

baseLatency , of type double , readonly

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

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

outputLatency , of type double , readonly

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

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

1.2.3. Methods

close()

Closes the AudioContext , releasing the system resources 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:

1. Let promise be a new Promise.

2. If the control thread state flag on the AudioContext is closed reject the promise with InvalidStateError , abort these steps, returning promise .

3. Set the control thread state flag on the AudioContext to closed .

4. Queue a control message to close the AudioContext .

5. Return promise .

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

1. Attempt to release system resources .

2. Set the rendering thread state to suspended .

   This will stop rendering.

3. 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.

4. Queue a task on the control thread ’s event loop, to execute these steps:

   1. Resolve promise .

   2. If the state attribute of the AudioContext is not already " closed ":

      1. Set the state attribute of the AudioContext to " closed ".

      2. Queue a task to fire a simple event named statechange at the AudioContext .

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

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

No parameters.

Return type: Promise < void >

createMediaElementSource(mediaElement)

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

Arguments for the AudioContext.createMediaElementSource() method.

Parameter	Type	Nullable	Optional	Description
mediaElement	HTMLMediaElement	✘	✘	The media element that will be re-routed.

Return type: MediaElementAudioSourceNode

createMediaStreamDestination()

Creates a MediaStreamAudioDestinationNode

No parameters.

Return type: MediaStreamAudioDestinationNode

createMediaStreamSource(mediaStream)

Creates a MediaStreamAudioSourceNode .

Arguments for the AudioContext.createMediaStreamSource() method.

Parameter	Type	Nullable	Optional	Description
mediaStream	MediaStream	✘	✘	The media stream that will act as source.

Return type: MediaStreamAudioSourceNode

createMediaStreamTrackSource(mediaStreamTrack)

Creates a MediaStreamTrackAudioSourceNode .

Arguments for the AudioContext.createMediaStreamTrackSource() method.

Parameter	Type	Nullable	Optional	Description
mediaStreamTrack	MediaStreamTrack	✘	✘	The MediaStreamTrack that will act as source. The value of its kind attribute must be equal to "audio" , or an InvalidStateError exception MUST be thrown.

Return type: MediaStreamTrackAudioSourceNode

getOutputTimestamp()

Returns a new AudioTimestamp instance containing two 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-2] ).

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

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

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

function outputPerformanceTime(contextTime) {
  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

resume()

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

When resume is called, execute these steps:

1. Let promise be a new Promise.

2. If the control thread state on the AudioContext is closed reject the promise with InvalidStateError , abort these steps, returning promise .

3. Set [[suspended by user]] to false .

4. If the context is not allowed to start , append promise to [[pending promises]] and [[pending resume promises]] and abort these steps, returning promise .

5. Set the control thread state on the AudioContext to running .

6. Queue a control message to resume the AudioContext .

7. Return promise .

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

1. Attempt to acquire system resources .

2. Set the rendering thread state on the AudioContext to running .

3. Start rendering the audio graph .

4. In case of failure, queue a task on the control thread to execute the following, and abort these steps:

   1. Reject all promises from [[pending resume promises]] in order, then clear [[pending resume promises]] .

   2. Additionally, remove those promises from [[pending promises]] .

5. Queue a task on the control thread ’s event loop, to execute these steps:

   1. Clear [[pending resume promises]] . Additionally, remove those promises from [[pending promises]] .

   2. Resolve promise .

   3. If the state attribute of the AudioContext is not already " running ":

      1. Set the state attribute of the AudioContext to " running ".

      2. Queue a task to fire a simple event named statechange at the AudioContext .

No parameters.

Return type: Promise < void >

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:

1. Let promise be a new Promise.

2. If the control thread state on the AudioContext is closed reject the promise with InvalidStateError , abort these steps, returning promise .

3. Append promise to [[pending promises]] .

4. Set [[suspended by user]] to true .

5. Set the control thread state on the AudioContext to suspended .

6. Queue a control message to suspend the AudioContext .

7. Return promise .

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

1. Attempt to release system resources .

2. Set the rendering thread state on the AudioContext to suspended .

3. Queue a task on the control thread ’s event loop, to execute these steps:

   1. Resolve promise .

   2. If the state attribute of the AudioContext is not already " suspended ":

      1. Set the state attribute of the AudioContext to " suspended ".

      2. Queue a task to fire a simple event named statechange at the AudioContext .

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

No parameters.

Return type: Promise < void >

1.2.4. AudioContextOptions

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

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

1.2.4.1. Dictionary AudioContextOptions Members

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

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

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

sampleRate , of type float

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

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

1.2.5. AudioTimestamp

dictionary AudioTimestamp {
  double contextTime;
  DOMHighResTimeStamp performanceTime;
};

1.2.5.1. Dictionary AudioTimestamp Members

contextTime , of type double

Represents a point in the time coordinate system of BaseAudioContext’s currentTime .

performanceTime , of type DOMHighResTimeStamp

Represents a point in the time coordinate system of a Performance interface implementation (described in [hr-time-2] ).

1.3. The OfflineAudioContext Interface

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

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

1.3.1. Constructors

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:

1. Set the control thread state for c to "suspended" .

2. Set the rendering thread state for c to "suspended" .

3. 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

length , of type unsigned long , readonly

The size of the buffer in sample-frames. This is the same as the value of the length parameter for the constructor.

oncomplete , of type EventHandler

An EventHandler of type OfflineAudioCompletionEvent . It is the last event fired on an OfflineAudioContext .

1.3.3. Methods

startRendering()

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

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 :

1. If the [[rendering started]] slot on the OfflineAudioContext is true , return a rejected promise with InvalidStateError , and abort these steps.
2. Set the [[rendering started]] slot of the OfflineAudioContext to true .
3. Let promise be a new promise.
4. 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 .
5. If an exception was thrown during the preceding AudioBuffer constructor call, reject promise with this exception.
6. Otherwise, in the case that the buffer was successfully constructed, begin offline rendering .
7. Append promise to [[pending promises]] .
8. Return promise .

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

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

No parameters.

Return type: Promise < AudioBuffer >

resume()

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

When resume is called, execute these steps:

1. Let promise be a new Promise.

2. 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 .

3. Set the control thread state flag on the OfflineAudioContext to running .

4. Queue a control message to resume the OfflineAudioContext .

5. Return promise .

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

1. Set the rendering thread state on the OfflineAudioContext to running .

2. Start rendering the audio graph .

3. In case of failure, queue a task on the control thread to reject promise and abort these steps:

4. Queue a task on the control thread ’s event loop, to execute these 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 task to fire a simple event named statechange at the OfflineAudioContext .

No parameters.

Return type: Promise < void >

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 < void >

1.3.4. OfflineAudioContextOptions

This specifies the options to use in constructing an OfflineAudioContext .

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

1.3.4.1. Dictionary OfflineAudioContextOptions Members

length , of type unsigned long

The length of the rendered AudioBuffer in sample-frames.

numberOfChannels , of type unsigned long , defaulting to 1

The number of channels for this OfflineAudioContext .

sampleRate , of type float

The sample rate for this OfflineAudioContext .

1.3.5. The OfflineAudioCompletionEvent Interface

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

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

1.3.5.1. Attributes

renderedBuffer , of type AudioBuffer , readonly

An AudioBuffer containing the rendered audio data.

1.3.5.2. OfflineAudioCompletionEventInit

dictionary OfflineAudioCompletionEventInit : EventInit {
  required AudioBuffer renderedBuffer;
};

1.3.5.2.1. Dictionary OfflineAudioCompletionEventInit Members

renderedBuffer , of type AudioBuffer

Value to be assigned to the renderedBuffer attribute of the event.

1.4. The AudioBuffer Interface

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

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

AudioBuffer has four internal slots:

[[number of channels]]

The number of audio channels for this AudioBuffer , which is an unsigned long.

[[length]]

The length of each channel of this AudioBuffer , which is an unsigned long.

[[sample rate]]

The sample-rate, in Hz, of this AudioBuffer , a float.

[[internal data]]

A data block holding the audio sample data.

[Exposed=Window]
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);
  void copyFromChannel (Float32Array destination,
                        unsigned long channelNumber,
                        optional unsigned long bufferOffset = 0);
  void copyToChannel (Float32Array source,
                      unsigned long channelNumber,
                      optional unsigned long bufferOffset = 0);
};

1.4.1. Constructors

AudioBuffer(options)

1. If any of the values in options lie outside its nominal range, throw a NotSupportedError exception and abort the following steps.

2. Let b be a new AudioBuffer object.

3. 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]] .

4. 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.

5. Return b .

Arguments for the AudioBuffer.constructor(options) method.

Parameter	Type	Nullable	Optional	Description
options	AudioBufferOptions	✘	✘	An AudioBufferOptions that determine the properties for this AudioBuffer .

1.4.2. Attributes

duration , of type double , readonly

Duration of the PCM audio data in seconds.

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

length , of type unsigned long , readonly

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

numberOfChannels , of type unsigned long , readonly

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

sampleRate , of type float , readonly

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

1.4.3. Methods

copyFromChannel(destination, channelNumber, 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: void

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: void

getChannelData(channel)

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

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

Arguments for the AudioBuffer.getChannelData() method.

Parameter	Type	Nullable	Optional	Description
channel	unsigned long	✘	✘	This parameter is an index representing the particular channel to get data for. An index value of 0 represents the first channel. This index value MUST be less than [[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.

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

AudioNode s 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 .

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

1.5.1. AudioNode Creation

AudioNode s 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 AudioNode s 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.

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.

AudioNode s are EventTarget s, as described in [DOM] . This means that it is possible to dispatch events to AudioNode s the same way that other EventTarget s 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.

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

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.

AudioNode s have a non-zero tail-time if they have internal processing state such that input in the past affects the future output. AudioNode s 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 AudioNode s 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 AudioNode s that have a tail-time .

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

1.5.4. Attributes

channelCount , of type unsigned long

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

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

AudioDestinationNode

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

AudioContext

The channel count MUST be between 1 and 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.

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.

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.

context , of type BaseAudioContext , readonly

The BaseAudioContext which owns this AudioNode .

numberOfInputs , of type unsigned long , readonly

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

numberOfOutputs , of type unsigned long , readonly

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

1.5.5. Methods

connect(destinationNode, output, input)

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

For example:

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

will have the same effect as

nodeA.connect(nodeB);

This method returns destination AudioNode object.

Arguments for the AudioNode.connect(destinationNode, output, input) method.

Parameter	Type	Nullable	Optional	Description
destinationNode				The destination parameter is the AudioNode to connect to. If the destination parameter is an AudioNode that has been created using another AudioContext , an InvalidAccessError MUST be thrown . That is, AudioNode s cannot be shared between AudioContext s. 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

connect(destinationParam, output)

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: void

disconnect()

Disconnects all outgoing connections from the AudioNode .

No parameters.

Return type: void

disconnect(output)

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

Arguments for the AudioNode.disconnect(output) method.

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

Return type: void

disconnect(destinationNode)

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

Arguments for the AudioNode.disconnect(destinationNode) method.

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

Return type: void

disconnect(destinationNode, output)

Disconnects a specific output of the AudioNode from 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: void

disconnect(destinationNode, output, input)

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

Arguments for the AudioNode.disconnect(destinationNode, output, input) method.

Parameter	Type	Nullable	Optional	Description
destinationNode				The destinationNode parameter is the AudioNode to disconnect. If there is no connection to the destinationNode from the given 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: void

disconnect(destinationParam)

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

Arguments for the AudioNode.disconnect(destinationParam) method.

Parameter	Type	Nullable	Optional	Description
destinationParam	AudioParam	✘	✘	The destinationParam parameter is the AudioParam to disconnect. If there is no connection to the destinationParam , an InvalidAccessError exception MUST be thrown.

Return type: void

disconnect(destinationParam, output)

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

Arguments for the AudioNode.disconnect(destinationParam, output) method.

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

Return type: void

1.5.6. AudioNodeOptions

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

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

1.5.6.1. 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 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 AudioNode s can be connected to an AudioParam , summing with the intrinsic parameter value.

Some synthesis and processing AudioNode s have AudioParam s as attributes whose values MUST be taken into account on a per-audio-sample basis. For other AudioParam s, 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 AudioParam s 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 AudioParam s 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 AudioParam s have constraints on whether the automation rate can be changed.

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

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 AudioParam s playbackRate and detune MUST be " k-rate ". An InvalidStateError must be thrown if the rate is changed to " a-rate " .

DynamicsCompressorNode

The AudioParam s threshold , knee , ratio , attack , and release MUST be " k-rate ". An InvalidStateError must be thrown if the rate is changed to " a-rate " .

PannerNode

If the panningModel is " HRTF ", the setting of the automationRate for any AudioParam of the PannerNode is ignored. Likewise, the setting of the automationRate for any AudioParam of the AudioListener is ignored. In this case, the AudioParam behaves as if the automationRate were set to " k-rate ".

defaultValue , of type float , readonly

Initial value for the value attribute.

maxValue , of type float , readonly

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

minValue , of type float , readonly

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

value , of type float

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

Getting this attribute returns the contents of the [[current value]] slot. 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

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

1. Let \(E_1\) be the event (if any) at time \(t_1\) where \(t_1\) is the largest number satisfying \(t_1 \le t_c\).

2. Let \(E_2\) be the event (if any) at time \(t_2\) where \(t_2\) is the smallest number satisfying \(t_c \lt t_2\).

3. 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.

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.

5. Remove all events with time greater than \(t_c\).

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

Arguments for the AudioParam.cancelAndHoldAtTime() method.

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

Return type: AudioParam

cancelScheduledValues(cancelTime)

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

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

exponentialRampToValueAtTime(value, endTime)

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

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

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

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

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

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

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

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

Arguments for the AudioParam.exponentialRampToValueAtTime() method.

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

Return type: AudioParam

linearRampToValueAtTime(value, endTime)

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

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

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

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

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

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

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

Arguments for the AudioParam.linearRampToValueAtTime() method.

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

Return type: AudioParam

setTargetAtTime(target, startTime, timeConstant)

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

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

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

where \(V_0\) is the initial value (the [[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

setValueAtTime(value, startTime)

Schedules a parameter value change at the given time.

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

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

$$
  v(t) = V
$$

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

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

Arguments for the AudioParam.setValueAtTime() method.

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

Return type: AudioParam

setValueCurveAtTime(values, startTime, duration)

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

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

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

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

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

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

Arguments for the AudioParam.setValueCurveAtTime() method.

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

Return type: AudioParam

1.6.3. Computation of Value

There are two different kind of AudioParam s, 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 AudioParam s that are used with other AudioParam s 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 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.

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

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.

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

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

1.7.1. Attributes

onended , of type EventHandler

A property used to set the EventHandler (described in HTML [HTML] ) for the ended event that is dispatched 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 AudioScheduledSourceNode s, the onended event is dispatched when the stop time determined by stop() is reached. For an AudioBufferSourceNode , the event is also dispatched because the duration has been reached or if the entire buffer has been played.

1.7.2. Methods

start(when)

Schedules a sound to playback at an exact time.

When this method is called, execute these steps:

1. If this AudioScheduledSourceNode internal slot [[source started]] is true, an InvalidStateError exception MUST be thrown.

2. 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.

3. Set the internal slot [[source started]] on this AudioScheduledSourceNode to true .

4. Queue a control message to start the AudioScheduledSourceNode , including the parameter values in the messsage.

5. Send a control message to the associated AudioContext to run it in the rendering thread only when 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 .

Arguments for the AudioScheduledSourceNode.start(when) method.

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

Return type: void

stop(when)

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

When this method is called, execute these steps:

1. If this AudioScheduledSourceNode internal slot [[source started]] is not true , an InvalidStateError exception MUST be thrown.

2. Check for any errors that must be thrown due to parameter constraints described below.

3. Queue a control message to stop the AudioScheduledSourceNode , including the parameter values in the messsage.

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

Arguments for the AudioScheduledSourceNode.stop(when) method.

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

Return type: void

1.8. The AnalyserNode Interface

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

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

1.8.1. Constructors

AnalyserNode(context, options)

Arguments for the AnalyserNode.constructor(context, options) method.

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

1.8.2. Attributes

fftSize , of type unsigned long

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

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

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

frequencyBinCount , of type unsigned long , readonly

Half the FFT size.

maxDecibels , of type double

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

minDecibels , of type double

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

smoothingTimeConstant , of type double

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

1.8.3. Methods

getByteFrequencyData(array)

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: void

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: void

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: void

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: void

1.8.4. AnalyserOptions

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

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

1.8.4.1. Dictionary AnalyserOptions Members

fftSize , of type unsigned long , defaulting to 2048

The desired initial size of the FFT for frequency-domain analysis.

maxDecibels , of type double , defaulting to -30

The desired initial maximum power in dB for FFT analysis.

minDecibels , of type double , defaulting to -100

The desired initial minimum power in dB for FFT analysis.

smoothingTimeConstant , of type double , defaulting to 0.8

The desired initial smoothing constant for the FFT analysis.

1.8.5. Time-Domain Down-Mixing

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