2.1.1 Attributes

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 the audio block size of 128 samples. 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.

destination of type AudioDestinationNode, readonly

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

listener of type AudioListener, readonly

An AudioListener which is used for 3D spatialization.

onstatechange of type EventHandler

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

sampleRate of type float, readonly

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

state of type AudioContextState, readonly

Describes the current state of this BaseAudioContext. The context state MUST begin in "suspended", and transitions to "running" when system resources are acquired and audio has begun processing. For OfflineAudioContexts, the state will remain in "suspended" until startRendering() is called, at which point it will transition to "running", and then to "closed" once audio processing has completed and oncomplete has been fired.

When the state is "suspended", a call to resume() will cause a transition to "running", or a call to close() will cause a transition to "closed".

When the state is "running", a call to suspend() will cause a transition to "suspended", or a call to close() will cause a transition to "closed".

When the state is "closed", no further state transitions are possible.

2.1.2 Methods

close

Closes the audio context, releasing any system audio resources used by the BaseAudioContext. This will not automatically release all BaseAudioContext-created objects, unless other references have been released as well; however, it will forcibly release any system audio resources that might prevent additional AudioContexts from being created and used, suspend the progression of the BaseAudioContext's currentTime, and stop processing audio data. The promise resolves when all AudioContext-creation-blocking resources have been released. If this is called on OfflineAudioContext, then return a promise rejected with a DOMException whose name is InvalidStateError.

No parameters.

Return type: Promise<void>

createAnalyser

Create an AnalyserNode.

No parameters.

Return type: AnalyserNode

createAudioWorker

Creates an AudioWorker object and loads the associated script into an AudioWorkerGlobalScope, then resolves the returned Promise.

Parameter	Type	Nullable	Optional	Description
scriptURL	DOMString	✘	✘	This parameter represents the URL of the script to be loaded as an AudioWorker node factory. See AudioWorker section for more detail.

Return type: Promise<AudioWorker>

createBiquadFilter

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

No parameters.

Return type: BiquadFilterNode

createBuffer

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.

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.
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 8192 to 96000.

Return type: AudioBuffer

createBufferSource

Creates an AudioBufferSourceNode.

No parameters.

Return type: AudioBufferSourceNode

createChannelMerger

Creates a ChannelMergerNode representing a channel merger. An IndexSizeError exception MUST be thrown for invalid parameter values.

Parameter	Type	Nullable	Optional	Description
numberOfInputs	unsigned long = 6	✘	✔	The numberOfInputs parameter 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

Creates an ChannelSplitterNode representing a channel splitter. An IndexSizeError exception MUST be thrown for invalid parameter values.

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

Return type: ChannelSplitterNode

createConvolver

Creates a ConvolverNode.

No parameters.

Return type: ConvolverNode

createDelay

Creates a DelayNode representing a variable delay line. The initial default delay time will be 0 seconds.

Parameter	Type	Nullable	Optional	Description
maxDelayTime	double = 1.0	✘	✔	The maxDelayTime parameter is optional and 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.

Return type: DelayNode

createDynamicsCompressor

Creates a DynamicsCompressorNode

No parameters.

Return type: DynamicsCompressorNode

createGain

Create an GainNode.

No parameters.

Return type: GainNode

createIIRFilter

Creates an IIRFilterNode representing a general IIR Filter.

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 tranfer 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

Creates an OscillatorNode

No parameters.

Return type: OscillatorNode

createPanner

This method is DEPRECATED, as it is intended to be replaced by createSpatialPanner or createStereoPanner, depending on the scenario. Creates a PannerNode.

No parameters.

Return type: PannerNode

createPeriodicWave

Creates a PeriodicWave representing a waveform containing arbitrary harmonic content. The real and imag parameters must be of type Float32Array (described in [TYPED-ARRAYS]) of equal lengths greater than zero or an IndexSizeError exception MUST be thrown. All implementations must support arrays up to at least 8192. These parameters specify the Fourier coefficients of a Fourier series representing the partials of a periodic waveform. The created PeriodicWave will be used with an OscillatorNode and, by default, will represent a normalized time-domain waveform having maximum absolute peak value of 1. Another way of saying this is that the generated waveform of an OscillatorNode will have maximum peak value at 0dBFS. Conveniently, this corresponds to the full-range of the signal values used by the Web Audio API. Because the PeriodicWave is normalized by default on creation, the real and imag parameters represent relative values. If normalization is disabled via the disableNormalization parameter, this normalization is disabled, and the time-domain waveform has the amplitudes as given by the Fourier coefficients.

As PeriodicWave objects maintain their own copies of these arrays, any modification of the arrays uses as the real and imag parameters after the call to createPeriodicWave() will have no effect on the PeriodicWave object.

Parameter	Type	Nullable	Optional	Description
real	Float32Array	✘	✘	The real parameter represents an array of cosine terms (traditionally the A terms). In audio terminology, the first element (index 0) is the DC-offset of the periodic waveform. The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on. The first element is ignored and implementations must set it to zero internally.
imag	Float32Array	✘	✘	The imag parameter represents an array of sine terms (traditionally the B terms). The first element (index 0) should be set to zero (and will be ignored) since this term does not exist in the Fourier series. The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.
constraints	PeriodicWaveConstraints	✘	✔	If not given, the waveform is normalized. Otherwise, the waveform is normalized according the value given by constraints.

Return type: PeriodicWave

createScriptProcessor

This method is DEPRECATED, as it is intended to be replaced by createAudioWorker. Creates a ScriptProcessorNode for direct audio processing using JavaScript. 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.

Parameter	Type	Nullable	Optional	Description
bufferSize	unsigned long = 0	✘	✔	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 audioprocess 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 = 2	✘	✔	This parameter determines the number of channels for this node's input. Values of up to 32 must be supported.
numberOfOutputChannels	unsigned long = 2	✘	✔	This parameter determines the number of channels for this node's output. Values of up to 32 must be supported.

Return type: ScriptProcessorNode

createSpatialPanner

Creates a SpatialPannerNode.

No parameters.

Return type: SpatialPannerNode

createStereoPanner

Creates a StereoPannerNode.

No parameters.

Return type: StereoPannerNode

createWaveShaper

Creates a WaveShaperNode representing a non-linear distortion.

No parameters.

Return type: WaveShaperNode

decodeAudioData

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 or video elements. The buffer passed to decodeAudioData has its content-type determined by sniffing, as described in [mimesniff].

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

The following steps must be performed:

1. Let promise be a new promise.
2. If audioData is null or not a valid ArrayBuffer:
   1. Let error be a DOMException whose name is NotSupportedError.
   2. Reject promise with error.
   3. If errorCallback is not missing, invoke errorCallback with error.
   4. Terminate this algorithm.
3. Neuter the audioData ArrayBuffer in such a way that JavaScript code may not access or modify the data anymore.
4. Queue a decoding operation to be performed on another thread.
5. Return promise.
6. In the decoding thread:
   1. Attempt to decode the encoded audioData into linear PCM.
   2. If a decoding error is encountered due to the audio format not being recognized or supported, or because of corrupted/unexpected/inconsistent data, then, on the main thread's event loop:
      1. Let error be a DOMException whose name is "EncodingError".
      2. Reject promise with error.
      3. If errorCallback is not missing, invoke errorCallback with error.
   3. 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. On the main thread's event loop:
         1. Let buffer be an AudioBuffer containing the final result (after possibly sample-rate converting).
         2. Resolve promise with buffer.
         3. If successCallback is not missing, invoke successCallback with buffer.

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

Return type: Promise<AudioBuffer>

resume

Resumes the progression of the BaseAudioContext's currentTime in an audio context that has been suspended, which may involve re-priming the frame buffer contents. The promise resolves when the system has re-acquired (if necessary) access to audio hardware and has begun streaming to the destination, or immediately (with no other effect) if the context is already running. The promise is rejected if the context has been closed. If the context is not currently suspended, the promise will resolve.

Note that until the first block of audio has been rendered following a call to this method, currentTime remains unchanged.

No parameters.

Return type: Promise<void>

suspend

Suspends the progression of BaseAudioContext'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 BaseAudioContext for some time, and wishes to let the audio hardware power down. 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.

While the system is suspended, MediaStreams will have their output ignored; that is, data will be lost by the real time nature of media streams. HTMLMediaElements will similarly have their output ignored until the system is resumed. Audio Workers and ScriptProcessorNodes will simply not fire their onaudioprocess events while suspended, but will resume when 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.

No parameters.

Return type: Promise<void>

2.1.3 Callback DecodeSuccessCallback Parameters

decodedData of type AudioBuffer

The AudioBuffer containing the decoded audio data.

2.1.4 Callback DecodeErrorCallback Parameters

error of type DOMException

The error that occurred while decoding.

2.1.5 Dictionary AudioContextOptions Members

playbackCategory of type AudioContextPlaybackCategory, defaulting to "interactive"

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

2.1.6 Lifetime

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

2.2.1 Methods

createMediaElementSource

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.

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

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

Return type: MediaStreamAudioSourceNode

2.3.1 Attributes

oncomplete of type EventHandler

An EventHandler of type OfflineAudioCompletionEvent.

2.3.2 Methods

resume

Resumes the progression of time in an audio context that has been suspended. The promise resolves immediately because the OfflineAudioContext does not require the audio hardware. If the context is not currently suspended or the rendering has not started, the promise is rejected with InvalidStateError.

In contrast to a live AudioContext, the value of currentTime always reflects the start time of the next block to be rendered by the audio graph, since the context's audio stream does not advance in time during suspension.

No parameters.

Return type: Promise<void>

startRendering

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

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

The following steps must be performed:

1. If startRendering has already been called previously, then return a promise rejected with InvalidStateError.
2. Let promise be a new promise.
3. Asynchronously perform the following steps:
   1. Let buffer be a new AudioBuffer, with a number of channels, length and sample rate equal respectively to the numberOfChannels, length and sampleRate parameters used when this instance's constructor was called.
   2. Given the current connections and scheduled changes, start rendering length sample-frames of audio into buffer.
   3. For every render quantum, check and suspend the rendering if necessary.
   4. If a suspended context is resumed, continue to render the buffer.
   5. Once the rendering is complete,
      1. Resolve promise with 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 buffer.
4. Return promise.

No parameters.

Return type: Promise<AudioBuffer>

suspend

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

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

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

Return type: Promise<void>

2.3.3 The OfflineAudioCompletionEvent Interface

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

interface OfflineAudioCompletionEvent : Event {
    readonly        attribute AudioBuffer renderedBuffer;
};

2.4.1 Attributes

channelCount of type unsigned long

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.

See the 5. Channel up-mixing and down-mixing section for more information on this attribute.

channelCountMode of type ChannelCountMode

Determines how channels will be counted when up-mixing and down-mixing connections to any inputs to the node. This attribute has no effect for nodes with no inputs.

See the 5. Channel up-mixing and down-mixing section for more information on this attribute.

channelInterpretation of type ChannelInterpretation

Determines how individual channels will be treated when up-mixing and down-mixing connections to any inputs to the node. This attribute has no effect for nodes with no inputs.

See the 5. Channel up-mixing and down-mixing section for more information on this attribute.

context of type AudioContext, readonly

The AudioContext 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, like the ChannelMergerNode and the AudioWorkerNode 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 AudioWorkerNode.

2.4.2 Methods

connect

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:

Example 4

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

will have the same effect as

Example 5

nodeA.connect(nodeB);

Parameter	Type	Nullable	Optional	Description
destination	AudioNode	✘	✘	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, AudioNodes cannot be shared between AudioContexts.
output	unsigned long = 0	✘	✔	The output parameter is an index describing which output of the AudioNode from which to connect. If this paremeter is out-of-bound, an IndexSizeError exception MUST be thrown. It is possible to connect an AudioNode output to more than one input with multiple calls to connect(). Thus, "fan-out" is supported.
input	unsigned long = 0	✘	✔	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 first AudioNode. This is allowed only if there is at least one DelayNode in the cycle or a NotSupportedError exception MUST be thrown.

Return type: AudioNode

connect

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

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

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

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

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);
    

Parameter	Type	Nullable	Optional	Description
destination	AudioParam	✘	✘	The destination parameter is the AudioParam to connect to. This method does not return destination AudioParam object.
output	unsigned long = 0	✘	✔	The output parameter is an index describing which output of the AudioNode from which to connect. If the parameter is out-of-bound, an IndexSizeError exception MUST be thrown.

Return type: void

disconnect

Disconnects all outgoing connections from the AudioNode.

No parameters.

Return type: void

disconnect

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

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

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

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

Return type: void

disconnect

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

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

Return type: void

disconnect

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

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

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.

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

Return type: void

disconnect

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.

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

Return type: void

2.4.3 Lifetime

This section is informative.

An implementation may choose any method to avoid unnecessary resource usage and unbounded memory growth of unused/finished nodes. The following is a description to help guide the general expectation of how node lifetime would be managed.

An AudioNode will live as long as there are any references to it. There are several types of references:

1. A normal JavaScript reference obeying normal garbage collection rules.
2. A playing reference for both AudioBufferSourceNodes and OscillatorNodes. These nodes maintain a playing reference to themselves while they are currently playing.
3. A connection reference which occurs if another AudioNode is connected to it.
4. A tail-time reference which an AudioNode maintains on itself as long as it has any internal processing state which has not yet been emitted. For example, a ConvolverNode has a tail which continues to play even after receiving silent input (think about clapping your hands in a large concert hall and continuing to hear the sound reverberate throughout the hall). Some AudioNodes have this property. Please see details for specific nodes.

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

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

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

2.5.1 Attributes

maxChannelCount of type unsigned long, readonly

The maximum number of channels that the channelCount attribute can be set to. An AudioDestinationNode representing the audio hardware end-point (the normal case) can potentially output more than 2 channels of audio if the audio hardware is multi-channel. maxChannelCount is the maximum number of channels that this hardware is capable of supporting. If this value is 0, then this indicates that channelCount may not be changed. This will be the case for an AudioDestinationNode in an OfflineAudioContext and also for basic implementations with hardware support for stereo output only.

channelCount defaults to 2 for a destination in a normal AudioContext, and may be set to any non-zero value less than or equal to maxChannelCount. An IndexSizeError exception MUST be thrown if this value is not within the valid range. Giving a concrete example, if the audio hardware supports 8-channel output, then we may set channelCount to 8, and render 8-channels of output.

For anAudioDestinationNode in an OfflineAudioContext, the channelCount is determined when the offline context is created and this value may not be changed.

2.6.1 Attributes

defaultValue of type float, readonly

Initial value for the value attribute.

value of type float

The parameter's floating-point value. This attribute is initialized to the defaultValue. If value is set during a time when there are any automation events scheduled then it will be ignored and no exception will be thrown.

The effect of setting this attribute is equivalent to calling setValueAtTime() with the current AudioContext's currentTime and the requested value. Subsequent accesses to this attribute's getter will return the same value.

2.6.2 Methods

cancelScheduledValues

Cancels all scheduled parameter changes with times greater than or equal to startTime. Active setTargetAtTime automations (those with startTime less than the supplied time value) will also be cancelled.

Parameter	Type	Nullable	Optional	Description
startTime	double	✘	✘	The starting time at and 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 TypeError exception MUST be thrown if startTime is negative or is not a finite number.

Return type: AudioParam

exponentialRampToValueAtTime

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. It is an error if either \(V_0\) or \(V_1\) is not strictly positive.

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 \(t \geq T_1\), \(v(t) = V_1\).

Parameter	Type	Nullable	Optional	Description
value	float	✘	✘	The value the parameter will exponentially ramp to at the given time. A NotSupportedError exception MUST be thrown if this value is less than or equal to 0, or if the value at the time of the previous event is less than or 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 TypeError exception MUST be thrown if endTime is negative or is not a finite number.

Return type: AudioParam

linearRampToValueAtTime

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 \(t \geq T_1\), \(v(t) = V_1\).

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 TypeError exception MUST be thrown if endTime is negative or is not a finite number.

Return type: AudioParam

setTargetAtTime

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: \(T_0 \leq t < T_1\), where \(T_0\) is the startTime parameter and \(T_1\) represents the time of the event following this event (or \(\infty\) if there are no following events):

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

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

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 TypeError exception MUST be thrown if start is negative or is not a finite number.
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 strictly positive or a TypeError exception MUST be thrown. 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

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.

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 AudioContext's currentTime attribute at which the parameter changes to the given value. A TypeError exception MUST be thrown if startTime is negative or is not a finite number.

Return type: AudioParam

setValueCurveAtTime

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).

Parameter	Type	Nullable	Optional	Description
values	Float32Array	✘	✘	A Float32Array 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 the AudioParam.
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 TypeError exception MUST be thrown if startTime is negative or is not a finite number.
duration	double	✘	✘	The amount of time in seconds (after the time parameter) where values will be calculated according to the values parameter.

Return type: AudioParam

2.6.3 Computation of Value

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

1. An intrinsic parameter value will be calculated at each time, which is either the value set directly to the value attribute, or, if there are any scheduled parameter changes (automation events) with times before or at this time, the value as calculated from these events. If the value attribute is set after any automation events have been scheduled, then these events will be removed. When read, the value attribute always returns the intrinsic value for the current time. If automation events are removed from a given time range, then the intrinsic value will remain unchanged and stay at its previous value until either the value attribute is directly set, or automation events are added for the time range.
2. 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. If there are no AudioNodes connected to it, then this value is 0, having no effect on the computedValue.
3. The computedValue is the sum of the intrinsic value and the value calculated from (2).

2.6.4 AudioParam Automation Example

var curveLength = 44100;
var curve = new Float32Array(curveLength);
for (var i = 0; i < curveLength; ++i)
    curve[i] = Math.sin(Math.PI * i / curveLength);

var t0 = 0;
var t1 = 0.1;
var t2 = 0.2;
var t3 = 0.3;
var t4 = 0.325;
var t5 = 0.5;
var t6 = 0.6;
var t7 = 0.7;
var t8 = 1.0;
var timeConstant = 0.1;

param.setValueAtTime(0.2, t0);
param.setValueAtTime(0.3, t1);
param.setValueAtTime(0.4, t2);
param.linearRampToValueAtTime(1, t3);
param.linearRampToValueAtTime(0.8, t4);
param.setTargetAtTime(.5, t4, timeConstant);
// Compute where the setTargetAtTime will be at time t5 so we can make
// the following exponential start at the right point so there's no
// jump discontinuity.  From the spec, we have
//   v(t) = 0.5 + (0.8 - 0.5)*exp(-(t-t4)/timeConstant)
// Thus v(t5) = 0.5 + (0.8 - 0.5)*exp(-(t5-t4)/timeConstant)
param.setValueAtTime(0.5 + (0.8 - 0.5)*Math.exp(-(t5 - t4)/timeConstant), t5);
param.exponentialRampToValueAtTime(0.75, t6);
param.exponentialRampToValueAtTime(0.05, t7);
param.setValueCurveAtTime(curve, t7, t8 - t7);

2.7.1 Attributes

gain of type AudioParam, readonly

Represents the amount of gain to apply. Its default value is 1 (no gain change). The nominal minValue is 0, but may be set negative for phase inversion. The nominal maxValue is 1, but higher values are allowed (no exception thrown).This parameter is a-rate

2.8.1 Attributes

delayTime of type AudioParam, readonly

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

If DelayNode is part of a cycle, then the value of the delayTime attribute is clamped to a minimum of 128 frames (one block).

This parameter is a-rate.

2.9.1 Attributes

duration of type double, readonly

Duration of the PCM audio data in seconds.

length of type long, readonly

Length of the PCM audio data in sample-frames.

numberOfChannels of type long, readonly

The number of discrete audio channels.

sampleRate of type float, readonly

The sample-rate for the PCM audio data in samples per second.

2.9.2 Methods

copyFromChannel

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

Parameter	Type	Nullable	Optional	Description
destination	Float32Array	✘	✘	The array the channel data will be copied to.
channelNumber	unsigned long	✘	✘	The index of the channel to copy the data from. If channelNumber is greater or equal than the number of channel of the AudioBuffer, an IndexSizeError MUST be thrown.
startInChannel	unsigned long = 0	✘	✔	An optional offset to copy the data from. If startInChannel is greater than the length of the AudioBuffer, an IndexSizeError MUST be thrown.

Return type: void

copyToChannel

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

Parameter	Type	Nullable	Optional	Description
source	Float32Array	✘	✘	The array the channel data will be copied from.
channelNumber	unsigned long	✘	✘	The index of the channel to copy the data to. If channelNumber is greater or equal than the number of channel of the AudioBuffer, an IndexSizeError MUST be thrown.
startInChannel	unsigned long = 0	✘	✔	An optional offset to copy the data to. If startInChannel is greater than the length of the AudioBuffer, an IndexSizeError MUST be thrown.

Return type: void

getChannelData

Returns the Float32Array representing the PCM audio data for the specific channel.

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

Return type: Float32Array

2.10.1 Attributes

buffer of type AudioBuffer, nullable

Represents the audio asset to be played. This attribute can only be set once, or a InvalidStateError MUST be thrown.

detune of type AudioParam, readonly

An aditional parameter to modulate the speed at which is rendered the audio stream. Its default value is 0. Its nominal range is [-1200; 1200]. This parameter is k-rate.

loop of type boolean

Indicates if the audio data should play in a loop. The default value is false. If loop is dynamically modified during playback, the new value will take effect on the next processing block of audio.

loopEnd of type double

An optional value in seconds where looping should end if the loop attribute is true. Its value is exclusive of the content of the loop: the sample frames comprising the loop run from the values loopStart to loopEnd-(1.0/sampleRate). Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer. If loopEnd is less than 0, looping will end at 0. If loopEnd is greater than the duration of the buffer, looping will end at the end of the buffer. This attribute is converted to an exact sample frame offset within the buffer by multiplying by the buffer's sample rate and rounding to the nearest integer value. Thus its behavior is independent of the value of the playbackRate parameter.

loopStart of type double

An optional value in seconds where looping should begin if the loop attribute is true. Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer. If loopStart is less than 0, looping will begin at 0. If loopStart is greater than the duration of the buffer, looping will begin at the end of the buffer. This attribute is converted to an exact sample frame offset within the buffer by multiplying by the buffer's sample rate and rounding to the nearest integer value. Thus its behavior is independent of the value of the playbackRate parameter.

onended of type EventHandler

A property used to set the EventHandler (described in HTML[HTML]) for the ended event that is dispatched to AudioBufferSourceNode node types. When the playback of the buffer for an AudioBufferSourceNode is finished, an event of type Event (described in HTML [HTML]) will be dispatched to the event handler.

playbackRate of type AudioParam, readonly

The speed at which to render the audio stream. Its default value is 1. This parameter is k-rate.

2.10.2 Methods

start

Schedules a sound to playback at an exact time. start may only be called one time and must be called before stop is called or an InvalidStateError exception MUST be thrown.

Parameter	Type	Nullable	Optional	Description
when	double = 0	✘	✔	The when parameter describes at what time (in seconds) the sound should start playing. It is in the same time coordinate system as the AudioContext's currentTime attribute. If 0 is passed in for this value or if the value is less than currentTime, then the sound will start playing immediately. A TypeError exception MUST be thrown if when is negative.
offset	double = 0	✘	✔	The offset parameter describes the offset time in the buffer (in seconds) where playback will begin. If 0 is passed in for this value, then playback will start from the beginning of the buffer. A TypeError exception MUST be thrown if offset is negative. If offset is greater than loopEnd, playback will begin at loopEnd (and immediately loop to loopStart). This parameter is converted to an exact sample frame offset within the buffer by multiplying by the buffer's sample rate and rounding to the nearest integer value. Thus its behavior is independent of the value of the playbackRate parameter.
duration	double	✘	✔	The duration parameter describes the duration of the portion (in seconds) to be played. If this parameter is not passed, the duration will be equal to the total duration of the AudioBuffer minus the offset parameter. Thus if neither offset nor duration are specified then the implied duration is the total duration of the AudioBuffer. An TypeError exception MUST be thrown if duration is negative.

Return type: void

stop

Schedules a sound to stop playback at an exact time.

Parameter	Type	Nullable	Optional	Description
when	double = 0	✘	✔	The when parameter describes at what time (in seconds) the sound 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 TypeError exception MUST be thrown if when is negative. If stop is called again after already have 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.

Return type: void

2.10.3 Looping

If the loop attribute is true when start() is called, then playback will continue indefinitely until stop() is called and the stop time is reached. We'll call this "loop" mode. Playback always starts at the point in the buffer indicated by the offset argument of start(), and in loop mode will continue playing until it reaches the actualLoopEnd position in the buffer (or the end of the buffer), at which point it will wrap back around to the actualLoopStart position in the buffer, and continue playing according to this pattern.

In loop mode then the actual loop points are calculated as follows from the loopStart and loopEnd attributes:

if ((loopStart || loopEnd) && loopStart >= 0 && loopEnd > 0 && loopStart < loopEnd) {
    actualLoopStart = loopStart;
    actualLoopEnd = min(loopEnd, buffer.duration);
} else {
    actualLoopStart = 0;
    actualLoopEnd = buffer.duration;
}

Note that the default values for loopStart and loopEnd are both 0, which indicates that looping should occur from the very start to the very end of the buffer.

Please note that as a low-level implementation detail, the AudioBuffer is at a specific sample-rate (usually the same as the AudioContext sample-rate), and that the loop times (in seconds) must be converted to the appropriate sample-frame positions in the buffer according to this sample-rate.

When scheduling the beginning and the end of playback using the start() and stop() methods, the resulting start or stop time MUST be rounded to the nearest sample-frame in the sample rate of the AudioContext. That is, no sub-sample scheduling is possible.

2.11.1 Security with MediaElementAudioSourceNode and cross-origin resources

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

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

2.12.1 Attributes

onloaded of type EventHandler

The onloaded handler is called after the script is successfully loaded and its global scope code is run to initialize the AudioWorkerGlobalScope.

onmessage of type EventHandler

The onmessage handler is called whenever the AudioWorkerGlobalScope posts a message back to the main thread.

parameters of type array of AudioWorkerParamDescriptor, readonly

This array contains descriptors for each of the current parameters on nodes created by this AudioWorker. This enables users of the AudioWorker to easily iterate over the AudioParam names and default values.

2.12.2 Methods

addParameter

Causes a correspondingly-named read-only AudioParam to be present on any AudioWorkerNodes created (previously or subsequently) by this AudioWorker, and a correspondingly-named read-only Float32Array to be present on the parameters object exposed on the AudioProcessEvent on subsequent audio processing events for such nodes. The AudioParam may immediately have its scheduling methods called, its .value set, or AudioNodes connected to it.

The name parameter is the name used for the read-only AudioParam added to the AudioWorkerNode, and the name used for the read-only Float32Array that will be present on the parameters object exposed on subsequent AudioProcessEvents.

The defaultValue parameter is the default value for the AudioParam's value attribute, as well as therefore the default value that will appear in the Float32Array in the worker script (if no other parameter changes or connections affect the value).

Parameter	Type	Nullable	Optional	Description
name	DOMString	✘	✘
defaultValue	float	✘	✘

Return type: AudioParam

createNode

Creates a node instance in the audio worker.

Parameter	Type	Nullable	Optional	Description
numberOfInputs	int	✘	✘
numberOfOutputs	int	✘	✘

Return type: AudioWorkerNode

postMessage

postMessage may be called to send a message to the AudioWorkerGlobalScope, similar to the algorithm defined by [Workers].

Parameter	Type	Nullable	Optional	Description
message	any	✘	✘
transfer	sequence<Transferable>	✘	✔

Return type: void

removeParameter

Removes a previously-added parameter named name from all AudioWorkerNodes associated with this AudioWorker and its AudioWorkerGlobalScope. This will also remove the correspondingly-named read-only AudioParam from the AudioWorkerNode, and will remove the correspondingly-named read-only Float32Arrays from the AudioProcessEvent's parameters member on subsequent audio processing events. A NotFoundError exception must be thrown if no parameter with that name exists on this AudioWorker.

The name parameter identifies the parameter to be removed.

Parameter	Type	Nullable	Optional	Description
name	DOMString	✘	✘

Return type: void

terminate

The terminate() method, when invoked, must cause the cessation of any AudioProcessEvents being dispatched inside the AudioWorker's associated AudioWorkerGlobalScope. It will also cause all associated AudioWorkerNodes to cease processing, and will cause the destruction of the worker's context. In practical terms, this means all nodes created from this AudioWorker will disconnect themselves, and will cease performing any useful functions.

No parameters.

Return type: void

2.12.3 The AudioWorkerNode Interface

This interface represents an AudioNode which interacts with a Worker thread to generate, process, or analyse audio directly. The user creates a separate audio processing worker script, which is hosted inside the AudioWorkerGlobalScope and runs inside the audio processing thread, rather than the main UI thread. The AudioWorkerNode represents the processing node in the main processing thread's node graph; the AudioWorkerGlobalScope represents the context in which the user's audio processing script is run.

Nota bene that if the Web Audio implementation normally runs audio process at higher than normal thread priority, utilizing AudioWorkerNodes may cause demotion of the priority of the audio thread (since user scripts cannot be run with higher than normal priority).

    numberOfInputs  : variable
    numberOfOutputs : variable

    channelCount = numberOfInputChannels;
    channelCountMode = "explicit";
    channelInterpretation = "speakers";

The number of input and output channels specified in the createAudioWorkerNode() call determines the initial number of input and output channels (and the number of channels present for each input and output in the AudioBuffers passed to the AudioProcess event handler inside the AudioWorkerGlobalScope). It is invalid for both numberOfInputChannels and numberOfOutputChannels to be zero.

Example usage:

    var bitcrusherFactory = context.createAudioWorker( "bitcrusher.js" );
    var bitcrusherNode = bitcrusherFactory.createNode();

interface AudioWorkerNode : AudioNode {
    void postMessage (any message, optional sequence<Transferable> transfer);
                    attribute EventHandler onmessage;
};

Note that AudioWorkerNode objects will also have read-only AudioParam objects for each named parameter added via the addParameter method on the AudioWorker. As this is dynamic, it cannot be captured here in IDL.

2.12.4 The AudioWorkerParamDescriptor Interface

This interface represents the description of an AudioWorkerNode AudioParam - in short, its name and default value. This enables easy iteration over the AudioParams from an AudioWorkerGlobalScope (which does not have an instance of those AudioParams).

interface AudioWorkerParamDescriptor {
    readonly        attribute DOMString name;
    readonly        attribute float     defaultValue;
};

2.12.5 The AudioWorkerGlobalScope Interface

This interface is a DedicatedWorkerGlobalScope-derived object representing the context in which an audio processing script is run; it is designed to enable the generation, processing, and analysis of audio data directly using JavaScript in a Worker thread, with shared context between multiple instances of audio nodes. This facilitates nodes that may have substantial shared data, e.g. a convolution node.

The AudioWorkerGlobalScope handles - audioprocess events dispatched - synchronously to process audio frame blocks for nodes created by this worker. - audioprocess events are only - dispatched for nodes that have at least one input - or one output connected. TODO: should this be true?

interface AudioWorkerGlobalScope : DedicatedWorkerGlobalScope {
    readonly        attribute float                        sampleRate;
    AudioParam addParameter (DOMString name, float defaultValue);
    void       removeParameter (DOMString name);
                    attribute EventHandler                 onaudioprocess;
                    attribute EventHandler                 onnodecreate;
    readonly        attribute AudioWorkerParamDescriptor[] parameters;
};

2.12.6 The AudioWorkerNodeProcessor Interface

An object supporting this interface represents each individual node instantiated in an AudioWorkerGlobalScope; it is designed to manage the data for an individual node. Shared context between multiple instances of audio nodes is accessible from the AudioWorkerGlobalScope; this object represents the individual node and can be used for data storage or main-thread communication.

interface AudioWorkerNodeProcessor : EventTarget {
    void postMessage (any message, optional sequence<Transferable> transfer);
                    attribute EventHandler onmessage;
};

2.14.1 Attributes

inputs of type Array, readonly

An array of channelCounts for the inputs.

node of type AudioWorkerNodeProcessor, readonly

The new node being created. Any node-local data storage (e.g., the buffer for a delay node) should be created on this object.

outputs of type Array, readonly

An array of channelCounts for the outputs.

2.15.1 Attributes

inputs of type array of array of Float32Array, readonly

A readonly Array of Arrays of Float32Arrays. The top-level Array is organized by input; each input may contain multiple channels; each channel contains a Float32Array of sample data. The initial size of the channel array will be determined by the number of channels specified for that input in the createAudioWorkerNode() method. However, an onprocess handler may alter this number of channels in the input dynamically, either by adding a Float32Array of blocksize length (128) or by reducing the Array (by reducing the Array.length or by using Array.pop() or Array.slice(). The event object, the Array and the Float32Arrays will be reused by the processing system, in order to minimize memory churn.

Any reordering performed on the Array for an input will not reorganize the connections to the channels for subsequent events.

node of type AudioWorkerNodeProcessor, readonly

The node to which this processing event is being dispatched. Any node-local data storage (e.g., the buffer for a delay node) should be maintained on this object.

outputs of type array of array of Float32Array, readonly

A readonly Array of Arrays of Float32Arrays. The top-level Array is organized by output; each output may contain multiple channels; each channel contains a Float32Array of sample data. The initial size of the channel array will be determined by the number of channels specified for that output in the createAudioWorkerNode() method. However, an onprocess handler may alter this number of channels in the output dynamically, either by adding a Float32Array of blocksize length (128) or by reducing the Array (by reducing the Array.length or by using Array.pop() or Array.slice(). The event object, the Array and the Float32Arrays will be reused by the processing system, in order to minimize memory churn.

Any reordering performed on the Array for an output will not reorganize the connections to the channels for subsequent events.

parameters of type object, readonly

This object attribute exposes a correspondingly-named read-only Float32Array for each parameter that has been added via addParameter. As this is dynamic, this cannot be captured in IDL. The length of this Float32Array will correspond to the length of the inputBuffer. The contents of this Float32Array will be the values to be used for the AudioParam at the corresponding points in time. It is expected that this Float32Array will be reused by the audio engine.

playbackTime of type double, readonly

The starting time of the block of audio being processed in response to this event. By definition this will be equal to the value of BaseAudioContext's currentTime attribute that was most recently observable in the control thread.

2.17.1 Attributes

coneInnerAngle of type float

A parameter for directional audio sources, this is an angle, in degrees, inside of which there will be no volume reduction. The default value is 360, and the value is used modulo 360.

coneOuterAngle of type float

A parameter for directional audio sources, this is an angle, in degrees, outside of which the volume will be reduced to a constant value of coneOuterGain. The default value is 360 and the value is used modulo 360.

coneOuterGain of type float

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

distanceModel of type DistanceModelType

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

maxDistance of type float

The maximum distance between source and listener, after which the volume will not be reduced any further. The default value is 10000.

panningModel of type PanningModelType

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

refDistance of type float

A reference distance for reducing volume as source move further from the listener. The default value is 1.

rolloffFactor of type float

Describes how quickly the volume is reduced as source moves away from listener. The default value is 1.

2.17.2 Methods

setOrientation

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

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

The default value is (1,0,0)

Parameter	Type	Nullable	Optional	Description
x	float	✘	✘
y	float	✘	✘
z	float	✘	✘

Return type: void

setPosition

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

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

The default value is (0,0,0)

Parameter	Type	Nullable	Optional	Description
x	float	✘	✘
y	float	✘	✘
z	float	✘	✘

Return type: void

setVelocity

Sets the velocity vector of the audio source. This vector controls both the direction of travel and the speed in 3D space. This velocity relative to the listener's velocity is used to determine how much doppler shift (pitch change) to apply. The units used for this vector is meters / second and is independent of the units used for position and orientation vectors.

The x, y, z parameters describe a direction vector indicating direction of travel and intensity.

The default value is (0,0,0)

Parameter	Type	Nullable	Optional	Description
x	float	✘	✘
y	float	✘	✘
z	float	✘	✘

Return type: void