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 one render quantum. Thus, for a running context, currentTime increases steadily as the system processes audio blocks, and always represents the time of the start of the next audio block to be processed. It is also the earliest possible time when any change scheduled in the current state might take effect.

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

destination of type AudioDestinationNode, readonly

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

listener of type AudioListener, readonly

An AudioListener which is used for 3D spatialization.

onstatechange of type EventHandler

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

sampleRate of type float, readonly

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

state of type AudioContextState, readonly

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

2.1.2 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

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 8000 to 96000.

Return type: AudioBuffer

createBufferSource

Factory method for a AudioBufferSourceNode.

No parameters.

Return type: AudioBufferSourceNode

createChannelMerger

Factory method for 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

Factory method for a 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

createConstantSource

Factory method for a ConstantSourceNode.

No parameters.

Return type: ConstantSourceNode

createConvolver

Factory method for a ConvolverNode.

No parameters.

Return type: ConvolverNode

createDelay

Factory method for a DelayNode. 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

Factory method for a DynamicsCompressorNode.

No parameters.

Return type: DynamicsCompressorNode

createGain

Factory method for GainNode.

No parameters.

Return type: GainNode

createIIRFilter

Factory method for 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 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

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 a new object of type PeriodicWaveOption.
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. Return the result of constructing a new PeriodicWave p, passing the BaseAudioContext this factory method has been called on as a first argument, and o.

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

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.

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 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 = 2	✘	✔	This parameter determines the number of channels for this node's input. Values of up to 32 must be supported. A NotSupportedError must be thrown if the number of channels is not supported.
numberOfOutputChannels	unsigned long = 2	✘	✔	This parameter determines the number of channels for this node's output. Values of up to 32 must be supported. A NotSupportedError must be thrown if the number of channels is not supported.

Return type: ScriptProcessorNode

No parameters.

Return type: Node

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

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

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

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

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

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 queue a task to execute the following step, on the control thread's event loop:
   1. Let error be a DOMException whose name is "EncodingError".
   2. Reject promise with error.
   3. If errorCallback is not missing, invoke errorCallback with error.
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. Queue a task on the control thread's event loop to execute the following steps:
      1. Let buffer be an AudioBuffer containing the final result (after possibly sample-rate conversion).
      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 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 flag on the BaseAudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
3. If the state attribute of the BaseAudioContext is already running, resolve promise, return it, and abort these steps.
4. If the BaseAudioContext is not allowed to start, append promise to pendingResumePromises and abort these steps, returning promise.
5. Set the control thread state flag on the BaseAudioContext to running.
6. Queue a control message to resume the BaseAudioContext.
7. Return promise.

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

1. Attempt to acquire system resources.
2. Set the rendering thread state flag on the BaseAudioContext 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 pendingResumePromises in order, then clear pendingResumePromises.
   2. Reject promise.
5. Queue a task on the control thread's event loop, to execute these steps:
   1. Resolve all promises from pendingResumePromises in order, then clear pendingResumePromises.
   2. Resolve promise.
   3. If the state attribute of the BaseAudioContext is not already running:
      1. Set the state attribute of the BaseAudioContext to running.
      2. Queue a task to fire a simple event named statechange at the BaseAudioContext.

No parameters.

Return type: Node

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 Lifetime

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

2.1.7 System resources associated with BaseAudioContext subclasses

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

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

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

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

2.2.1 Constructors

AudioContext

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 pendingResumePromises be an 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 AudioContext is not allowed to start, abort these steps.
6. Send a control message to start processing.

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

1. Attempt to acquire system resources.
2. In case of failure, abort these steps.
3. Set the rendering thread state to running on the AudioContext.
4. 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.

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

2.2.2 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.

2.2.3 Methods

close

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

When close is called, execute these steps:

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. If the state attribute of the AudioContext is already closed, resolve promise, return it, and abort these steps.
4. Set the control thread state flag on the AudioContext to closed.
5. Queue a control message to the AudioContext.
6. 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.
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 closed:
      1. Set the state attribute of the AudioContext to closed.
      2. Queue a task to fire a simple event named statechange at the AudioContext.

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

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

Creates a MediaStreamAudioSourceNode.

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

Return type: MediaStreamAudioSourceNode

createMediaStreamTrackSource

Creates a MediaStreamTrackAudioSourceNode.

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.

No parameters.

Return type: MediaStreamTrackAudioSourceNode

getOutputTimestamp

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

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

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

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

Example 4

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

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

Note

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

No parameters.

Return type: AudioTimestamp

suspend

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

When suspend is called, execute these steps:

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. If the state attribute of the AudioContext is already suspended, resolve promise, return it, and abort these steps.
4. Set the control thread state flag on the AudioContext to suspended.
5. Queue a control message to suspend the AudioContext.
6. 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, MediaStreams will have their output ignored; that is, data will be lost by the real time nature of media streams. HTMLMediaElements will similarly have their output ignored until the system is resumed. AudioWorkletNodes and ScriptProcessorNodes will cease to have their processing handlers invoked while suspended, but will resume when the context is resumed. For the purpose of AnalyserNode window functions, the data is considered as a continuous stream - i.e. the resume()/suspend() does not cause silence to appear in the AnalyserNode's stream of data. In particular, calling AnalyserNode functions repeatedly when a AudioContext is suspended MUST return the same data.

No parameters.

Return type: Promise<void>

2.2.4 AudioContextOptions

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

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

2.2.5 AudioTimestamp

[Exposed=Window]
dictionary AudioTimestamp {
    double              contextTime;
    DOMHighResTimeStamp performanceTime;
};

2.3.1 Constructors

OfflineAudioContext

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.

Parameter	Type	Nullable	Optional	Description
contextOptions	OfflineAudioContextOptions	✘	✘	The initial parameters needed to construct this context.

OfflineAudioContext

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

The OfflineAudioContext is constructed as if

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

were called instead.

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.

2.3.2 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, readonly

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

2.3.3 Methods

startRendering

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

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

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

1. Set a flag called renderingStarted on the OfflineAudioContext to true.
2. If the renderingStarted flag on the OfflineAudioContext is true, return a rejected promise with InvalidStateError, and abort these steps.
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. 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 the 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 promise with [[rendered buffer]].
   2. Queue a task to fire an event named complete at this instance, using an instance of OfflineAudioCompletionEvent whose renderedBuffer property is set to [[rendered buffer]].

No parameters.

Return type: Promise<AudioBuffer>

suspend

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

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

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>

2.3.4 OfflineAudioContextOptions

This specifies the options to use in constructing an OfflineAudioContext.

[Exposed=Window]
dictionary OfflineAudioContextOptions {
             unsigned long numberOfChannels = 1;
    required unsigned long length;
    required float         sampleRate;
};

2.3.5 The OfflineAudioCompletionEvent Interface

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

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

[Exposed=Window]
dictionary OfflineAudioCompletionEventInit : EventInit {
    required AudioBuffer renderedBuffer;
};

2.4.1 AudioNode Creation

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

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

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

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

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

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

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

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

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

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

2.4.2 Attributes

channelCount of type unsigned long

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

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

AudioDestinationNode

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

AudioContext

The channel count MUST be between 1 and 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.

ChannelSplitterNode

The channel count cannot be changed, and an InvalidStateError exception MUST be thrown for any attempt to change the value.

ChannelMergerNode

The channel count cannot be changed, and an InvalidStateError exception MUST be thrown for any attempt to change the value.

ConvolverNode

The channel count cannot changed from two, and a NotSupportedError exception MUST be thrown for any attempt to change the value..

DynamicsCompressorNode

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

PannerNode

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

ScriptProcessorNode

The channel count cannot be changed, and an InvalidStateError exception MUST be thrown for any attempt to change the value.

StereoPannerNode

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

See the 6. Channel up-mixing and down-mixing section 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. 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.

ChannelSplitterNode

The channel count mode cannot be changed from "explicit" and 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.

ConvolverNode

The channel count mode cannot be changed from "clamped-max", and a NotSupportedError exception MUST be thrown for any attempt to change the value.

DynamicsCompressorNode

The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".

PannerNode

The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".

ScriptProcessorNode

The channel count mode cannot be changed from "explicit" and an InvalidStateError exception MUST be thrown for any attempt to change the value.

StereoPannerNode

The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".

See the 6. 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. This attribute has no effect for nodes with no inputs.

See the 6. Channel up-mixing and down-mixing section for more information on this attribute.

When attempting to this attribute on a ChannelSplitterNode, an InvalidStateError MUST be thrown.

context of type BaseAudioContext, readonly

The BaseAudioContext which owns this AudioNode.

numberOfInputs of type unsigned long, readonly

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

numberOfOutputs of type unsigned long, readonly

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

2.4.3 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 5

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

will have the same effect as

Example 6

nodeA.connect(nodeB);

This method returns destination AudioNode object.

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 parameter is out-of-bound, an IndexSizeError exception MUST be thrown. It is possible to connect an AudioNode output to more than one input with multiple calls to connect(). Thus, "fan-out" is supported.
input	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.

The down-mixing to mono is equivalent to the down-mixing for an AudioNode with channelCount = 1, channelCountMode = "explicit", and channelInterpetation = "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);

Parameter	Type	Nullable	Optional	Description
destination	AudioParam	✘	✘	The destination parameter is the AudioParam to connect to. This method does not return destination AudioParam object. If destination 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 = 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 the destination, an InvalidAccessError exception MUST be thrown.

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, 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, 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.

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.

2.4.4 AudioNodeOptions

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

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

2.4.5 Lifetime

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

There are several types of references:

1. A normal reference obeying normal garbage collection rules.
2. A playing reference for AudioBufferSourceNodes, MediaElementAudioSourceNodes, MediaStreamAudioSourceNodes and OscillatorNodes. These nodes maintain a playing reference to themselves while they are currently playing.
3. An active reference for AudioWorkletNodes whose processorState property is set to running.
4. A connection reference which occurs if another AudioNode is connected to one or more of its inputs. Connections to a node's AudioParams do not imply a connection reference.
5. 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.
6. MediaStreams keep a MediaStreamAudioSourceNode alive as long as the underlying MediaStreamTrack that is playing through the MediaStreamAudioSourceNode has not ended (as per [mediacapture-streams]).
7. HTMLMediaElements keep their associated MediaElementAudioSourceNode alive as long as the HTMLMediaElement is in a state where audio could ever be played in the future.
   Note

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

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

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.

2.6.1 Attributes

defaultValue of type float, readonly

Initial value for the value attribute.

maxValue of type float, readonly

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

minValue of type float, readonly

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

value of type float

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

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

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

2.6.2 Methods

cancelAndHoldAtTime

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 the constant value that the original timeline would have had at time \(t_c\).

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

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.

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

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.

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

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.

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

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

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

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

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

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

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

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 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

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.

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 time 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

2.6.3 Computation of Value

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

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

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

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

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

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

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

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

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

onended of type EventHandler

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

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

2.7.2 Methods

start

Schedules a sound to playback at an exact time.

When this method is called, execute these steps:

1. If stop has been called on this node, or if an earlier call to start has already occurred, 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 start the AudioScheduledSourceNode, including the parameter values in the messsage.

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. 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

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 an earlier call to start has not already occurred, 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.

Parameter	Type	Nullable	Optional	Description
when	double = 0	✘	✔	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

2.8.1 Constructors

GainNode

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

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

2.8.2 Attributes

gain of type AudioParam, readonly

Represents the amount of gain to apply.

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

2.8.3 GainOptions

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

dictionary GainOptions : AudioNodeOptions {
    float gain = 1;
};

2.9.1 Constructors

DelayNode

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

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

2.9.2 Attributes

delayTime of type AudioParam, readonly

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

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

Parameter	Value	Notes
defaultValue	0
minValue	0
maxValue	maxDelayTime
Rate	a-rate

2.9.3 DelayOptions

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

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

2.10.1 Constructors

AudioBuffer

Let b be a new AudioBuffer object. Respectively assign the values of the attributes numberOfChannels, length, sampleRate of the AudioBufferOptions passed in the constructor to the internal slots [[number of channels]], [[length]], [[sample rate]].

Set the internal slot [[internal data]] of this AudioBuffer to the result of calling CreateByteDataBlock([[length]] * [[number of channels]]).

Note

This initializes the underlying storage to zero.

Parameter	Type	Nullable	Optional	Description
options	AudioBufferOptions	✘	✘

2.10.2 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]].

2.10.3 Methods

copyFromChannel

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

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

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

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

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

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.

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

2.10.4 AudioBufferOptions

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

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

2.11.1 Constructors

AudioBufferSourceNode

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

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

2.11.2 Attributes

buffer of type AudioBuffer, nullable

Represents the audio asset to be played. To set the buffer attribute, execute these steps:

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

detune of type AudioParam, readonly

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

Parameter	Value	Notes
defaultValue	0
minValue	most-negative-single-float	Approximately -3.4028235e38
maxValue	most-positive-single-float	Approximately 3.4028235e38
Rate	k-rate

loop of type boolean

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

loopEnd of type double

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

loopStart of type double

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

playbackRate of type AudioParam, readonly

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

Parameter	Value	Notes
defaultValue	1
minValue	most-negative-single-float	Approximately -3.4028235e38
maxValue	most-positive-single-float	Approximately 3.4028235e38
Rate	k-rate

2.11.3 Methods

start

Schedules a sound to playback at an exact time.

When this method is called, execute these steps:

1. If stop has been called on this node, or if an earlier call to start has already occurred, 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 start the AudioBufferSourceNode, including the parameter values in the messsage.

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

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 RangeError exception MUST be thrown if when is negative.
offset	double	✘	✔	The offset parameter supplies a playhead position where playback will begin. If 0 is passed in for this value, then playback will start from the beginning of the buffer. A RangeError exception MUST be thrown if offset is negative. If offset is greater than loopEnd, playback will begin at loopEnd (and immediately loop to loopStart). offset is silently clamped to [0, duration], when startTime is reached, where duration is the value of the duration attribute of the AudioBuffer set to the buffer attribute of this AudioBufferSourceNode.
duration	double	✘	✔	The duration parameter describes the duration of the sound (in seconds) to be played. If this parameter is passed, this method has exactly the same effect as the invocation of start(when, offset) followed by stop(when + duration). A RangeError exception MUST be thrown if duration is negative.

Return type: void

stop

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 an earlier call to start has not already occurred, 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 AudioBufferSourceNode, including the parameter values in the messsage.

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

Parameter	Type	Nullable	Optional	Description
when	double = 0	✘	✔	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

2.11.4 AudioBufferSourceOptions

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

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

2.11.5 Looping

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

Setting the loop attribute to true causes playback of the region of the buffer defined by the endpoints loopStart and loopEnd to continue indefinitely, once any part of the looped region has been played. While loop remains true, looped playback will continue until stop() is called, or the scheduled stop time has been reached.

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

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

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

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

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

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

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

2.11.6 Playback of AudioBuffer Contents

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

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

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

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

The description of the algorithm is as follows:

let buffer;  // AudioBuffer employed by this node
let context; // AudioContext employed by this node

// The following variables capture attribute and AudioParam values for the node.
// They are updated on a k-rate basis, prior to each invocation of process().
let loop;
let detune;
let loopStart;
let loopEnd;
let playbackRate;

// Variables for the node's playback parameters
let start = 0, offset = 0; // Set by start()
let stop = Infinity;  // Set by stop(), or by start() with a supplied duration

// Variables for tracking node's playback state
let bufferTime = 0, started = false, enteredLoop = false;
let dt = 1 / context.sampleRate;

// Handle invocation of start method call
function handleStart(when, pos, duration) {
  if (arguments.length >= 1) {
    start = when;
  }
  offset = pos;
  if (arguments.length >= 3) {
    stop = when + duration;
  }
}

// Handle invocation of stop method call
function handleStop(when) {
  if (arguments.length >= 1) {
    stop = when;
  } else {
    stop = context.currentTime;
  }
}

// Interpolate a multi-channel signal value for some sample frame.
// Returns an array of signal values.
function playbackSignal(position) {
  /*
    This function provides the playback signal function for buffer, which is a
    function that maps from a playhead position to a set of output signal
    values, one for each output channel. If |position| corresponds to the
    location of an exact sample frame in the buffer, this function returns
    that frame. Otherwise, its return value is determined by a UA-supplied
    algorithm that interpolates between sample frames in the neighborhood of
    position.

    If position is greater than or equal to loopEnd and there is no subsequent
    sample frame in buffer, then interpolation should be based on the sequence
    of subsequent frames beginning at loopStart.
   */
   ...
}

// Generate a single render quantum of audio to be placed
// in the channel arrays defined by output. Returns an array
// of |numberOfFrames| sample frames to be output.
function process(numberOfFrames) {
  let currentTime = context.currentTime; // context time of next rendered frame
  let output = [];  // accumulates rendered sample frames

  // Combine the two k-rate parameters affecting playback rate
  let computedPlaybackRate = playbackRate * Math.pow(2, detune / 1200);

  // Determine loop endpoints as applicable
  let actualLoopStart, actualLoopEnd;
  if (loop && buffer != null) {
    if (loopStart >= 0 && loopEnd > 0 && loopStart < loopEnd) {
      actualLoopStart = loopStart;
      actualLoopEnd = Math.min(loopEnd, buffer.duration);
    } else {
      actualLoopStart = 0;
      actualLoopEnd = buffer.duration;
    }
  } else {
    // If the loop flag is false, remove any record of the loop having been entered
    enteredLoop = false;
  }

  // Handle null buffer case
  if (buffer == null) {
    stop = currentTime; // force zero output for all time
  }
}
  // Render each sample frame in the quantum
  for (let index = 0; index < numberOfFrames; index++) {
    // Check that currentTime is within allowable range for playback
    if (currentTime < start || currentTime >= stop) {
      output.push(0); // this sample frame is silent
      currentTime += dt;
      continue;
    }

    if (!started) {
      // Take note that buffer has started playing and get initial playhead position.
      bufferTime = offset + ((currentTime - start) * computedPlaybackRate);
      started = true;
    }

    // Handle loop-related calculations
    if (loop) {
      // Determine if looped portion has been entered for the first time
      if (!enteredLoop) {
        if (offset < actualLoopEnd && bufferTime >= actualLoopStart) {
          // playback began before or within loop, and playhead is now past loop start
          enteredLoop = true;
        }
        if (offset >= actualLoopEnd && bufferTime < actualLoopEnd) {
          // playback began after loop, and playhead is now prior to the loop end
          enteredLoop = true;
        }
      }

      // Wrap loop iterations as needed. Note that enteredLoop
      // may become true inside the preceding conditional.
      if (enteredLoop) {
        while (bufferTime >= actualLoopEnd) {
          bufferTime -= actualLoopEnd - actualLoopStart;
        }
        while (bufferTime < actualLoopStart) {
          bufferTime += actualLoopEnd - actualLoopStart;
        }
      }
    }

    if (bufferTime >= 0 && bufferTime < buffer.duration) {
      output.push(playbackSignal(bufferTime));
    } else {
      output.push(0); // past end of buffer, so output silent frame
    }

    bufferTime += dt * computedPlaybackRate;
    currentTime += dt;
  } // End of render quantum loop

  if (currentTime >= stop) {
    // end playback state of this node.
    // no further invocations of process() will occur.
  }

  return output;
}

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

• context sample rate is 1000 Hz
• AudioBuffer content is shown with the first sample frame at the x origin.
• output signals are shown with the sample frame located at time start at the x origin.
• linear interpolation is depicted throughout, although a UA could employ other interpolation techniques.

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

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

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

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

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

2.12.1 Constructors

ConstantSourceNode

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

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

2.12.2 Attributes

offset of type AudioParam, readonly

The constant value of the source.

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

2.12.3 ConstantSourceOptions

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

dictionary ConstantSourceOptions {
    float offset = 1;
};

2.13.1 Constructors

MediaElementAudioSourceNode

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

Parameter	Type	Nullable	Optional	Description
context	BaseAudioContext	✘	✘	The BaseAudioContext this new MediaElementAudioSourceNode will be associated with.
options	MediaElementAudioSourceOptions	✘	✘	Parameter value for this MediaElementAudioSourceNode.

2.13.2 Attributes

mediaElement of type HTMLMediaElement, readonly

the HTMLMediaElement used when constructing this MediaElementAudioSourceNode.

2.13.3 MediaElementAudioSourceOptions

This specifies the options to use in constructing a MediaElementAudioSourceNode.

dictionary MediaElementAudioSourceOptions {
    required HTMLMediaElement mediaElement;
};