The BaseAudioContext Interface

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

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

suspended

This context is currently suspended (context time is not proceeding, audio hardware may be powered down/released).

running

Audio is being processed.

closed

This context has been released, and can no longer be used to process audio. All system audio resources have been released. Attempts to create new Nodes on the AudioContext will throw InvalidStateError. (AudioBuffers may still be created, through createBuffer, decodeAudioData, or the AudioBuffer constructor.)

readonly attribute AudioDestinationNode destination

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.

readonly attribute float sampleRate

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.

readonly attribute double currentTime

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.

readonly attribute AudioListener listener

An AudioListener which is used for 3D spatialization.

readonly attribute AudioContextState state

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

Promise<void> 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.

attribute EventHandler onstatechange

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.

AudioBuffer 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.

unsigned long numberOfChannels

Determines how many channels the buffer will have. An implementation must support at least 32 channels.

unsigned long length

Determines the size of the buffer in sample-frames.

float sampleRate

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.

Promise<AudioBuffer> 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]].

ArrayBuffer audioData

An ArrayBuffer containing compressed audio data.

optional DecodeSuccessCallback successCallback

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.

optional DecodeErrorCallback errorCallback

A callback function which will be invoked if there is an error decoding the audio file.

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.

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.

AudioBufferSourceNode createBufferSource()

Factory method for a AudioBufferSourceNode.

ConstantSourceNode createConstantSource()

Factory method for a ConstantSourceNode.

ScriptProcessorNode 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.

optional unsigned long bufferSize = 0

The bufferSize parameter determines the buffer size in units of sample-frames. If it's not passed in, or if the value is 0, then the implementation will choose the best buffer size for the given environment, which will be constant power of 2 throughout the lifetime of the node. Otherwise if the author explicitly specifies the bufferSize, it must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. This value controls how frequently the audioprocess event is dispatched and how many sample-frames need to be processed each call. Lower values for bufferSize will result in a lower (better) latency. Higher values will be necessary to avoid audio breakup and glitches. It is recommended for authors to not specify this buffer size and allow the implementation to pick a good buffer size to balance between latency and audio quality. If the value of this parameter is not one of the allowed power-of-2 values listed above, an IndexSizeError MUST be thrown.

optional unsigned long numberOfInputChannels = 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.

optional unsigned long numberOfOutputChannels = 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.

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

AnalyserNode createAnalyser()

Factory method for an AnalyserNode.

GainNode createGain()

Factory method for GainNode.

DelayNode createDelay()

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

optional double maxDelayTime = 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.

BiquadFilterNode createBiquadFilter()

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

IIRFilterNode createIIRFilter(sequence<double> b, sequence<double> a)

Factory method for an IIRFilterNode representing a general IIR Filter.

sequence<double> feedforward

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.

sequence<double> feedback

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.

WaveShaperNode createWaveShaper()

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

PannerNode createPanner()

Factory method for a PannerNode.

StereoPannerNode createStereoPanner()

Factory method for an a StereoPannerNode.

ConvolverNode createConvolver()

Factory method for a ConvolverNode.

ChannelSplitterNode createChannelSplitter()

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

optional unsigned long numberOfOutputs = 6

The number of outputs. Values of up to 32 must be supported. If not specified, then 6 will be used.

ChannelMergerNode createChannelMerger()

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

optional unsigned long numberOfInputs = 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.

DynamicsCompressorNode createDynamicsCompressor()

Factory method for a DynamicsCompressorNode.

OscillatorNode createOscillator()

Factory method for an OscillatorNode.

PeriodicWave createPeriodicWave()

Factory method to create a PeriodicWave. When calling this method, execute these steps:

• If real and imag are not of the same length, an IndexSizeError MUST be thrown.
• Let o a new object of type PeriodicWaveOption.
• Respectively set the real and imag parameters passed to this factory method to the attributes of the same name on o.
• Set the disableNormalization attribute on o to the value of the disableNormalization attribute of the constraints attribute passed to the factory method.
• 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.

sequence<float> real

A sequence of cosine parameters. See its constructor counterpart for a more detailed description.

sequence<float> imag

A sequence of sine parameters. See its constructor counterpart for a more detailed description.

optional PeriodicWaveConstraints constraints

If not given, the waveform is normalized. Otherwise, the waveform is normalized according the value given by constraints.

AudioBuffer decodedData

The AudioBuffer containing the decoded audio data.

DOMException error

The error that occurred while decoding.

The AudioContext Interface

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

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

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

balanced

Balance audio output latency and power consumption.

interactive

Provide the lowest audio output latency possible without glitching. This is the default.

playback

Prioritize sustained playback without interruption over audio output latency. Lowest power consumption.

Constructor

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 the AudioContext is not allowed to start, abort these steps.
5. 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.

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.

readonly attribute double baseLatency

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.

readonly attribute double outputLatency

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.

AudioTimestamp getOutputTimestamp()

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

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

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

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

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

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

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.

Promise<void> suspend()

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

When suspend is called, execute these steps:

1. Let promise be a new Promise.
2. If the control thread state 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.

Promise<void> 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 has been closed, implementation can choose to aggressively release more resources than when suspending.

MediaElementAudioSourceNode 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.

HTMLMediaElement mediaElement

The media element that will be re-routed.

MediaStreamAudioSourceNode createMediaStreamSource()

Creates a MediaStreamAudioSourceNode.

MediaStream mediaStream

The media stream that will act as source.

MediaStreamTrackAudioSourceNode createMediaStreamTrackSource()

Creates a MediaStreamTrackAudioSourceNode.

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.

MediaStreamAudioDestinationNode createMediaStreamDestination()

Creates a MediaStreamAudioDestinationNode

The OfflineAudioContext Interface

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

The OfflineAudioContext is 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.

unsigned long numberOfChannels

Determines how many channels the buffer will have. See createBuffer for the supported number of channels.

unsigned long length

Determines the size of the buffer in sample-frames.

float sampleRate

Describes the sample-rate of the linear PCM audio data in the buffer in sample-frames per second. See createBuffer for valid sample rates.

Promise<AudioBuffer> 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. Start to render the audio graph on another thread.
5. Return promise.

When rendering an audio graph on another thread, the following steps MUST happen on a rendering thread that is created for the occasion.

1. Let buffer be a new AudioBuffer, with a number of channels, length and sample rate equal respectively to the numberOfChannels, length and sampleRate parameters used when this instance's constructor was called.
2. Given the current connections and scheduled changes, start rendering length sample-frames of audio into buffer.
3. For every render quantum, check and suspend the rendering if necessary.
4. If a suspended context is resumed, continue to render the buffer.
5. Once the rendering is complete, queue a task on the control thread's event loop to perform the following steps:
   1. Resolve promise with buffer.
   2. If a suspended context is resumed, continue to render the buffer.
   3. Once the rendering is complete,
      1. Resolve promise with buffer.
      2. Queue a task to fire an event named complete at this instance, using an instance of OfflineAudioCompletionEvent whose renderedBuffer property is set to buffer.

Promise<void> 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.

double suspendTime

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

1. is negative or
2. is less than or equal to the current time or
3. is greater than or equal to the total render duration or
4. is scheduled by another suspend for the same time,

then the promise is rejected with InvalidStateError.

readonly attribute unsigned long length

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

attribute EventHandler oncomplete

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

The AudioNode Interface

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

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

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

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

For each input, an AudioNode performs a mixing (usually an up-mixing) of all connections to that input. Please see for more informative details, and the section for normative requirements.

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

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

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

max

computedNumberOfChannels is computed as the maximum of the number of channels of all connections. In this mode channelCount is ignored

clamped-max

Same as “max” up to a limit of the channelCount

explicit

computedNumberOfChannels is the exact value as specified in channelCount

speakers

use up-down-mix equations for mono/stereo/quad/5.1. In cases where the number of channels do not match any of these basic speaker layouts, revert to "discrete".

discrete

Up-mix by filling channels until they run out then zero out remaining channels. down-mix by filling as many channels as possible, then dropping remaining channels.

AudioNode connect()

AudioNode destination

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.

optional unsigned long output = 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.

optional unsigned long input = 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.

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

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

will have the same effect as

      nodeA.connect(nodeB);
    

This method returns destination AudioNode object.

void connect()

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

AudioParam destination

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.

optional unsigned long output = 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.

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

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

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

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

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

will have the same effect as

      nodeA.connect(param);
    

void disconnect()

Disconnects all outgoing connections from the AudioNode.

void disconnect()

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

unsigned long output

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.

void disconnect()

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

AudioNode destination

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

void disconnect()

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

AudioNode destination

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.

unsigned long output

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.

void disconnect()

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

AudioNode destination

The destination parameter is the AudioNode to disconnect. If there is no connection to the destination from the given output to the given input, an InvalidAccessError exception MUST be thrown.

unsigned long output

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.

unsigned long input

The input parameter is an index describing which input of the destination AudioNode to disconnect. If this parameter is out-of-bounds, an IndexSizeError exception MUST be thrown.

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.

AudioParam destination

The destination parameter is the AudioParam to disconnect. If there is no connection to the destination, an InvalidAccessError exception MUST be thrown.

void disconnect()

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

AudioParam destination

The destination parameter is the AudioParam to disconnect. If there is no connection to the destination, an InvalidAccessError exception MUST be thrown.

unsigned long output

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.

readonly attribute BaseAudioContext context

The BaseAudioContext which owns this AudioNode.

readonly attribute unsigned long numberOfInputs

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.

readonly attribute unsigned long numberOfOutputs

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.

attribute unsigned long channelCount

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..

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 section for more information on this attribute.

attribute ChannelCountMode 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.

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 section for more information on this attribute.

attribute ChannelInterpretation channelInterpretation

channelInterpetation 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 section for more information on this attribute.

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

The AudioDestinationNode Interface

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

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

      numberOfInputs  : 1
      numberOfOutputs : 1

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

For an AudioContext, the defaults are

      channelCount = 2
      channelCountMode = "explicit"
      channelInterpretation = "speakers"

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

For an OfflineAudioContext, the defaults are

      channelCount = numberOfChannels
      channelCountMode = "explicit"
      channelInterpretation = "speakers"

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

readonly attribute unsigned long maxChannelCount

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.

The AudioParam Interface

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

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

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

For each render quantum, the value of a k-rate parameter must be sampled at the time of the very first sample-frame, and that value must be used for the entire block. a-rate parameters must be sampled for each sample-frame of the block.

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

An AudioParam maintains a time-ordered event list which is initially empty. The times are in the time coordinate system of the AudioContext's currentTime attribute. The events define a mapping from time to value. The following methods can change the event list by adding a new event into the list of a type specific to the method. Each event has a time associated with it, and the events will always be kept in time-order in the list. These methods will be called automation methods:

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

The following rules will apply when calling these methods:

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

attribute float value

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

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

readonly attribute float defaultValue

Initial value for the value attribute.

readonly attribute float minValue

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

readonly attribute float maxValue

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

AudioParam setValueAtTime(float value, double startTime)

Schedules a parameter value change at the given time.

float value

The value the parameter will change to at the given time.

double startTime

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.

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.

AudioParam linearRampToValueAtTime(float value, double endTime)

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

float value

The value the parameter will linearly ramp to at the given time.

double endTime

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.

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.

AudioParam exponentialRampToValueAtTime(float value, double endTime)

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

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

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

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

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

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

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

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

float value

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.

double endTime

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.

AudioParam setTargetAtTime(float target, double startTime, float timeConstant)

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

float target

The value the parameter will start changing to at the given time.

double startTime

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.

float timeConstant

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).

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.

AudioParam setValueCurveAtTime(sequence<float> values, double startTime, double duration)

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

sequence<float> values

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.

double startTime

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.

double duration

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.

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.

AudioParam cancelScheduledValues(double cancelTime)

Cancels all scheduled parameter changes with times greater than or equal to cancelTime. Cancelling a scheduled parameter change means removing the scheduled event from the event list. Any active automations whose 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.

double cancelTime

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.

AudioParam cancelAndHoldAtTime(double cancelTime)

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

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

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

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

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

double cancelTime

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.

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 scheduled parameter changes (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.
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 nominal range for this parameter. Compound parameters have their final value clamped to their nominal range after having been computed from the different AudioParam value they are composed of.

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

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

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

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

AudioParam Automation 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);

The AudioScheduledSourceNode Interface

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

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

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

attribute EventHandler onended

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.

void start()

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

optional double when = 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.

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.

optional double when = 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.

The GainNode Interface

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

  numberOfInputs  : 1
  numberOfOutputs : 1

  channelCountMode = "max";
  channelInterpretation = "speakers";

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

This node has no tail-time reference.

Constructor

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

BaseAudioContext context

The BaseAudioContext this new GainNode will be associated with.

optional GainOptions options

Optional initial parameter values for this GainNode.

readonly attribute AudioParam gain

Represents the amount of gain to apply. This parameter is a-rate. Its nominal range is (-\(\infty\), +\(\infty\)).

The DelayNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

    channelCountMode = "max";
    channelInterpretation = "speakers";

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

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

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

This node has a tail-time reference such that this node continues to output non-silent audio with zero input up to the maxDelayTime of the node.

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

Constructor

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

BaseAudioContext context

The BaseAudioContext this new DelayNode will be associated with.

optional DelayOptions options

Optional initial parameter value for this DelayNode.

readonly attribute AudioParam delayTime

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.

Its nominal range is [0, maxDelayTime], where maxDelayTime is the value passed to the createDelay method on the AudioContext or the maxDelayTime member of the DelayOptions dictionary in the node constructor.

This parameter is a-rate.

The AudioBuffer Interface

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

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

AudioBuffer has four internal slots:

[[number of channels]]

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

[[\length]]

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

[[sample rate]]

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

[[internal data]]

A data block holding the audio sample data.

Constructor

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]]).

This initializes the underlying storage to zero.

AudioBufferOptions options

readonly attribute float sampleRate

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

readonly attribute unsigned long length

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

readonly attribute double duration

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]].

readonly attribute unsigned long numberOfChannels

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

Float32Array 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.

unsigned long channel

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.

void 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.

Float32Array destination

The array the channel data will be copied to.

unsigned long channelNumber

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.

optional unsigned long startInChannel = 0

An optional offset to copy the data from. If startInChannel is greater than the length of the AudioBuffer, an IndexSizeError MUST be thrown.

void copyToChannel()

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

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

Float32Array source

The array the channel data will be copied from.

unsigned long channelNumber

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.

optional unsigned long startInChannel = 0

An optional offset to copy the data to. If startInChannel is greater than the length of the AudioBuffer, an IndexSizeError MUST be thrown.

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

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

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

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

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

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

The AudioBufferSourceNode Interface

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

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

  numberOfInputs  : 0
  numberOfOutputs : 1

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

This node has no tail-time reference.

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

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

Constructor

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

BaseAudioContext context

The BaseAudioContext this new AudioBufferSourceNode will be associated with.

optional AudioBufferSourceOptions options

Optional initial parameter value for this AudioBufferSourceNode.

attribute AudioBuffer? buffer

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

1. Let new buffer be the AudioBuffer 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.

readonly attribute AudioParam playbackRate

The speed at which to render the audio stream. Its default value is 1. This parameter is k-rate. This is a compound parameter with detune. Its nominal range is \([-\infty, \infty]\).

readonly attribute AudioParam detune

An additional parameter, in cents, to modulate the speed at which is rendered the audio stream. Its default value is 0. This parameter is k-rate. This parameter is a compound parameter with playbackRate. Its nominal range is \((-\infty, \infty)\).

attribute boolean loop

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

attribute double loopStart

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.

attribute double loopEnd

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.

void 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.

optional double when = 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.

optional double offset

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.

optional double duration

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.

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 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 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.

optional double when = 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.

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) {
    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;
  }

  // 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:

The ConstantSourceNode Interface

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

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

  numberOfInputs  : 0
  numberOfOutputs : 1

Constructor

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

BaseAudioContext context

The BaseAudioContext this new ConstantSourceNode will be associated with.

optional ConstantSourceOptions options

Optional initial parameter value for this ConstantSourceNode.

readonly attribute AudioParam offset

The constant value of the source. Its default value is 1. This parameter is a-rate. Its nominal range is \((-\infty, \infty)\).

The MediaElementAudioSourceNode Interface

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

  numberOfInputs  : 0
  numberOfOutputs : 1

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

This node has no tail-time reference.

Constructor

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

BaseAudioContext context

The BaseAudioContext this new MediaElementAudioSourceNode will be associated with.

MediaElementAudioSourceOptions options

Parameter value for this MediaElementAudioSourceNode.

[SameObject] readonly attribute HTMLMediaElement mediaElement

the HTMLMediaElement used when constructing this MediaElementAudioSourceNode.

A MediaElementAudioSourceNode is created given an HTMLMediaElement using the AudioContext createMediaElementSource() method.

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

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

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

The AudioWorklet interface

Concepts

The AudioWorklet object allows developers to supply scripts (such as JavaScript or WebAssembly code) to process audio on the rendering thread, supporting custom AudioNodes. This processing mechanism ensures the synchronous execution of the script code with other built-in AudioNodes in the audio graph.

An associated pair of objects must be defined in order to realize this mechanism: AudioWorkletNode and AudioWorkletProcessor. The former represents the interface for the main global scope similar to other AudioNode objects, and the latter implements the internal audio processing within a special scope named AudioWorkletGlobalScope.

Importing a script via the addModule(moduleUrl) method registers class definitions of AudioWorkletProcessor under the AudioWorkletGlobalScope. There are two internal storage areas for the imported class definitions and the active instances created from the definition.

node name to processor definition map

Belongs to AudioWorkletGlobalScope. This map associates a string key to the corresponding AudioWorkletProcessor definition. Initially this map is empty and becomes populated when registerProcessor method is called.

node name to parameter descriptor map

Belongs to BaseAudioContext. This contains a collection of AudioParamDescriptors with an associated string key of node name. This internal storage is populated from node name to processor definition map by extracting a node name and the matching value from parameterDescriptors when a promise from addModule() on the Worklet gets resolevd.

// bypass.js script file, AudioWorkletGlobalScope
registerProcessor("Bypass", class extends AudioWorkletProcessor {
  process (inputs, outputs) {
    // Single input, single channel.
    var input = inputs[0], output = outputs[0];
    output[0].set(input[0]);
  }
});

// The main global scope
window.audioWorklet.addModule("bypass.js").then(function () {
  var context = new AudioContext();
  var bypass = new AudioWorkletNode(context, "Bypass");
});

At the instantiation of AudioWorkletNode in the main global scope, the counterpart AudioWorkletProcessor will also be created in AudioWorkletGlobalScope. These two objects communicate via the asynchronous message passing described in the processing model section.

[SameObject] readonly attribute Worklet audioWorklet

The audioWorklet attributes allows access to the Worklet object that can import a script containing AudioWorkletProcessor class definitions via the algorithm defined by [[!worklets-1]].

class MyProcessor extends AudioWorkletProcessor {
  static get parameterDescriptors() { 
    return [{
      name: 'myParam',
      defaultValue: 0.5,
      minValue: 0,
      maxValue: 1 
    }];
  }

  process(inputs, outputs, parameters) {
    // Get the first input and output.
    var input = inputs[0];
    var output = outputs[0];
    var myParam = parameters.myParam;

    // A simple amplifier for single input and output.
    for (var channel = 0; channel < output.length; ++channel) {
      for (var i = 0; i < output[channel].length; ++i) {
        output[channel][i] = input[channel][i] * myParam[i];
      }
    }
  }
}

AudioWorklet Sequence of Events

The following figure illustrates an idealized sequence of events occurring relative to an AudioWorklet:

The steps depicted in the diagram are one possible sequence of events involving the creation of an AudioContext and an associated AudioWorkletGlobalScope, followed by the creation of an AudioWorkletNode and its associated AudioWorkletProcessor.

1. In the main scope, window.audioWorklet is requested to import a script. No AudioWorkletGlobalScopes exist yet, so the script is fetched and added to the Worklet module responses map.
2. An AudioContext is created.
3. An AudioWorkletGlobalScope is created in association with the context's audio rendering thread. This is the global scope in which AudioWorkletProcessor class definitions will be evaluated.
4. As part of the global scope's initialization, the set of imported scripts is run, including the one that was previously imported.
5. As part of running the imported script, an AudioWorkletProcessor is registered under the key "Custom1" within the AudioWorkletGlobalScope.
6. In the main scope, an AudioWorkletNode is created using the key "Custom1" along with an opts dictionary of options.
7. As part of the node's creation, this key is used to look up the correct AudioWorkletProcessor subclass for instantiation.
8. An instance of the AudioWorkletProcessor subclass is instantiated with a structured clone of the same opts dictionary. This instance is paired with the previously created AudioWorkletNode.

window.audioWorklet.addModule('bitcrusher.js').then(function () {
  let context = new AudioContext();
  let osc = new OscillatorNode(context);
  let amp = new GainNode(context);

  // Create a worklet node. 'BitCrusher' identifies the 
  // AudioWorkletProcessor previously registered when
  // bitcrusher.js was imported. The options automatically
  // initialize the correspondingly named AudioParams.
  let bitcrusher = new AudioWorkletNode(context, 'BitCrusher', { 
    bitDepth: 8, 
    frequencyReduction: 0.5
  });

  osc.connect(bitcrusher).connect(amp).connect(context.destination);
  osc.start();
});

registerProcessor('BitCrusher', class extends AudioWorkletProcessor {

  static get parameterDescriptors () {
    return [{
      name: 'bitDepth',
      defaultValue: 12,
      minValue: 1,
      maxValue: 16 
    }, {
      name: 'frequencyReduction',
      defaultValue: 0.5,
      minValue: 0,
      maxValue: 1
    }];
  }

  constructor (options) {
    // We don't need to look at options: only AudioParams are initialized,
    // which were taken care of by the node.
    super(options);
    this._phase = 0;
    this._lastSampleValue = 0;
  }

  process (inputs, outputs, parameters) {
    let input = inputs[0];
    let output = outputs[0];
    let bitDepth = parameters.bitDepth;
    let frequencyReduction = parameters.frequencyReduction;

    for (let channel = 0; channel < output.length; ++channel) { 
      for (let i = 0; i < output[channel].length; ++i) {
        let step = Math.pow(0.5, bitDepth[i]);
        this._phase += frequencyReduction[i];
        if (this._phase >= 1.0) {
          this._phase -= 1.0;
          this._lastSampleValue = 
            step * Math.floor(input[channel][i] / step + 0.5);
        }
        output[channel][i] = this._lastSampleValue;
      }
    }

    // No need to return a value; this node's lifetime is dependent only on its
    // input connections.
  }

});

class VUMeterNode extends AudioWorkletNode {

  constructor (context, options) {
    // Setting default values for the input, the output and the channel count.
    options.numberOfInputs = 1;
    options.numberOfOutputs = 0;
    options.channelCount = 1;
    options.updatingInterval = options.hasOwnProperty('updatingInterval') 
      ? options.updatingInterval 
      : 100;

    super(context, 'VUMeter', options);

    // States in AudioWorkletNode
    this._updatingInterval = options.updatingInterval;
    this._volume = 0;

    // Handles updated values from AudioWorkletProcessor
    this.port.onmessage = event => {
      if (event.data.volume)
        this._volume = event.data.volume;
    }
    this.port.start();
  }

  get updatingInterval() {
    return this._updatingInterval;
  }

  set updatingInterval (intervalValue) {
    this._updatingInterval = intervalValue;
    this.port.postMessage({ updatingInterval: intervalValue });
  }

  draw () {
    /* Draw the meter based on the volume value. */
  }

}

// The application can use the node when this promise resolves.
let importAudioWorkletNode = window.audioWorklet.addModule('vumeterprocessor.js');

registerProcessor('VUMeter', class extends AudioWorkletProcessor {

  static meterSmoothingFactor = 0.9;
  static meterMinimum = 0.00001;

  constructor (options) {
    super(options);
    this._volume = 0;
    this._updatingInterval = options.updatingInterval;
    this._nextUpdateFrames = this.interval;

    this.port.onmessage = event => {
      if (event.data.updatingInterval)
        this._updatingInterval = event.data.updatingInterval;
    }
    this.port.start();
  }

  get interval () {
    return this._updatingInterval / 1000 * this.contextInfo.sampleRate;
  }

  process (inputs, outputs, parameters) {
    // Note that the input will be down-mixed to mono; however, if no inputs are
    // connected then zero channels will be passed in.
    if (inputs[0].length > 0) {
      let buffer = inputs[0][0];
      let bufferLength = buffer.length;
      let sum = 0, x = 0, rms = 0;

      // Calculated the squared-sum.
      for (let i = 0; i < bufferLength; ++i) {
        x = buffer[i];
        sum += x * x;
      }

      // Calculate the RMS level and update the volume.
      rms =  Math.sqrt(sum / bufferLength);
      this.volume = Math.max(rms, this._volume * meterSmoothingFactor);

      // Update and sync the volume property with the main thread.
      this._nextUpdateFrame -= bufferLength;
      if (this._nextUpdateFrame < 0) {
        this._nextUpdateFrame += this.interval;
        this.port.postMessage({ volume: this._volume });
      }
    }

    // Keep on processing if the volume is above a threshold, so that
    // disconnecting inputs does not immediately cause the meter to stop 
    // computing its smoothed value.
    return this._volume >= meterMinimum;
  }

});

<script src="vumeternode.js"></script>
<script>
  importAudioWorkletNode.then(function () {
    let context = new AudioContext();
    let oscillator = new Oscillator(context);
    let vuMeterNode = new VUMeterNode(context, { updatingInterval: 50 });

    oscillator.connect(vuMeterNode);

    function drawMeter () {
      vuMeterNode.draw();
      requestAnimationFrame(drawMeter);
    }

    drawMeter();
  });
</script>

The PannerNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

    channelCount = 2;
    channelCountMode = "clamped-max";
    channelInterpretation = "speakers";

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

There are channelCount constraints and channelCountMode constraints for this node.

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

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

This node may have a tail-time reference. If the panningModel is set to "HRTF", the node will produce non-silent output for silent input due to the inherent processing for the head responses.

equalpower

A simple and efficient spatialization algorithm using equal-power panning.

When this panning model is used, all the AudioParams used to compute the output of this node are a-rate.

HRTF

A higher quality spatialization algorithm using a convolution with measured impulse responses from human subjects. This panning method renders stereo output.

When this panning model is used, all the AudioParams used to compute the output of this node are k-rate.

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

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

linear

A linear distance model which calculates distanceGain according to:

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

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

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

inverse

An inverse distance model which calculates distanceGain according to:

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

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

exponential

An exponential distance model which calculates distanceGain according to:

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

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

Constructor

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

BaseAudioContext context

The BaseAudioContext this new PannerNode will be associated with.

optional PannerOptions options

Optional initial parameter value for this PannerNode.

attribute PanningModelType panningModel

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

readonly attribute AudioParam positionX

Sets the x coordinate position of the audio source in a 3D Cartesian system. The default value is 0. This parameter is a-rate when panningModel is "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam positionY

Sets the y coordinate position of the audio source in a 3D Cartesian system. The default value is 0. This parameter is a-rate when panningModel is "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam positionZ

Sets the z coordinate position of the audio source in a 3D Cartesian system. The default value is 0. The default value is 0. This parameter is a-rate when panningModel is "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam orientationX

Describes the x component of the vector of the direction the audio source is pointing in 3D Cartesian coordinate space. Depending on how directional the sound is (controlled by the cone attributes), a sound pointing away from the listener can be very quiet or completely silent. The default value is 1. This parameter is a-rate when panningModel is "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam orientationY

Describes the y component of the vector of the direction the audio source is pointing in 3D cartesian coordinate space. The default value is 0. This parameter is a-rate when panningModel is "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam orientationZ

Describes the Z component of the vector of the direction the audio source is pointing in 3D cartesian coordinate space. The default value is 0. This parameter is a-rate when panningModel is "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

attribute DistanceModelType distanceModel

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

attribute double refDistance

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

attribute double maxDistance

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

attribute double rolloffFactor

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

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

linear

The nominal range is \([0, 1]\).

inverse

The nominal range is \([0, \infty)\).

exponential

The nominal range is \([0, \infty)\).

attribute double coneInnerAngle

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

attribute double coneOuterAngle

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

attribute double coneOuterGain

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

void setPosition(float x, float y, float z)

This method is DEPRECATED. It is equivalent to setting positionX, positionY, and positionZ AudioParams directly.

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

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

The default value is (0,0,0)

void setOrientation(float x, float y, float z)

This method is DEPRECATED. It is equivalent to setting orientationX, orientationY, and orientationZ AudioParams directly.

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

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

The default value is (1,0,0)

The AudioListener Interface

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

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

The forwardX, forwardY, forwardZ parameters represent a direction vector in 3D space. Both a forward vector and an up vector are used to determine the orientation of the listener. In simple human terms, the forward vector represents which direction the person's nose is pointing. The up vector represents the direction the top of a person's head is pointing. These values are expected to be linearly independent (at right angles to each other), and unpredictable behavior may result if they are not. For normative requirements of how these values are to be interpreted, see the Spatialization/Panning section.

readonly attribute AudioParam positionX

Sets the x coordinate position of the audio listener in a 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate when used with a PannerNode that has a panningModel set to "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam positionY

Sets the y coordinate position of the audio listener in a 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate when used with a PannerNode that has a panningModel set to "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam positionZ

Sets the z coordinate position of the audio listener in a 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate when used with a PannerNode that has a panningModel set to "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam forwardX

Sets the x coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate when used with a PannerNode that has a panningModel set to "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam forwardY

Sets the y coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate when used with a PannerNode that has a panningModel set to "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam forwardZ

Sets the z coordinate component of the forward direction the listener is pointing in 3D Cartesian coordinate space. The default value is -1. This parameter is a-rate when used with a PannerNode that has a panningModel set to "equalpower", k-rate otherwise. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam upX

Sets the x coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam upY

Sets the y coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space. The default value is 1. This parameter is a-rate. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam upZ

Sets the z coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate. Its nominal range is \((-\infty, \infty)\).

void setPosition(float x, float y, float z)

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

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

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

The default value is (0,0,0)

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

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

Describes which direction the listener is pointing in the 3D cartesian coordinate space. Both a front vector and an up vector are provided. In simple human terms, the front vector represents which direction the person's nose is pointing. The up vector represents the direction the top of a person's head is pointing. These values are expected to be linearly independent (at right angles to each other). For normative requirements of how these values are to be interpreted, see the spatialization section.

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

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

The StereoPannerNode Interface

This interface represents a processing node which positions an incoming audio stream in a stereo image using a low-cost equal-power panning algorithm. This panning effect is common in positioning audio components in a stereo stream.

    numberOfInputs  : 1
    numberOfOutputs : 1

    channelCount = 2;
    channelCountMode = "clamped-max";
    channelInterpretation = "speakers";

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

There are channelCount constraints and channelCountMode constraints for this node.

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

This node has no tail-time reference.

Constructor

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

BaseAudioContext context

The BaseAudioContext this new StereoPannerNode will be associated with.

optional StereoPannerOptions options

Optional initial parameter value for this StereoPannerNode.

readonly attribute AudioParam pan

The position of the input in the output's stereo image. -1 represents full left, +1 represents full right. Its default value is 0, and its nominal range is [-1, 1]. This parameter is a-rate.

The ConvolverNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

    channelCount = 2;
    channelCountMode = "clamped-max";
    channelInterpretation = "speakers";

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

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

This node has a tail-time reference such that this node continues to output non-silent audio with zero input for the length of the buffer.

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

Constructor

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

BaseAudioContext context

The BaseAudioContext this new ConvolverNode will be associated with.

optional ConvolverOptions options

Optional initial parameter value for this ConvolverNode.

attribute AudioBuffer? buffer

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

To set the buffer attribute, execute these steps:

1. Let new buffer be the AudioBuffer 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.

The following text is non-normative. For normative information please see the channel configuration diagrams.

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

attribute boolean normalize

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

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

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

function calculateNormalizationScale(buffer)
{
    var GainCalibration = 0.00125;
    var GainCalibrationSampleRate = 44100;
    var MinPower = 0.000125;

    // Normalize by RMS power.
    var numberOfChannels = buffer.numberOfChannels;
    var length = buffer.length;

    var power = 0;

    for (var i = 0; i < numberOfChannels; i++) {
        var channelPower = 0;
        var channelData = buffer.getChannelData(i);

        for (var j = 0; j < length; j++) {
            var sample = channelData[j];
            channelPower += sample * sample;
        }

        power += channelPower;
    }

    power = Math.sqrt(power / (numberOfChannels * length));

    // Protect against accidental overload.
    if (!isFinite(power) || isNaN(power) || power < MinPower)
        power = MinPower;

    var scale = 1 / power;

    // Calibrate to make perceived volume same as unprocessed.
    scale *= GainCalibration;

    // Scale depends on sample-rate.
    if (buffer.sampleRate)
        scale *= GainCalibrationSampleRate / buffer.sampleRate;

    // True-stereo compensation.
    if (numberOfChannels == 4)
        scale *= 0.5;

    return scale;
}
      

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

Channel Configurations for Input, Impulse Response and Output

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

The first image in the diagram illustrates the general case, where the source has N input channels, the impulse response has K channels, and the playback system has M output channels. Because ConvolverNode is limited to 1 or 2 channels of input, not every case can be handled.

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

The AnalyserNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1    Note that this output may be left unconnected.

    channelCount = 1;
    channelCountMode = "max";
    channelInterpretation = "speakers";

This node has no tail-time reference.

Constructor

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

BaseAudioContext context

The BaseAudioContext this new AnalyserNode will be associated with.

optional AnalyserOptions options

Optional initial parameter value for this AnalyserNode.

void getFloatFrequencyData()

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

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

The frequency data are in dB units.

Float32Array array

This parameter is where the frequency-domain analysis data will be copied.

void getByteFrequencyData()

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

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

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

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

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

Uint8Array array

This parameter is where the frequency-domain analysis data will be copied.

void getFloatTimeDomainData()

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

Float32Array array

This parameter is where the time-domain sample data will be copied.

void getByteTimeDomainData()

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

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

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

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

Uint8Array array

This parameter is where the time-domain sample data will be copied.

attribute unsigned long fftSize

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

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

readonly attribute unsigned long frequencyBinCount

Half the FFT size.

attribute double minDecibels

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

attribute double maxDecibels

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

attribute double smoothingTimeConstant

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

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

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

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

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

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

The ChannelSplitterNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : Variable N (defaults to 6) // number of "active" (non-silent) outputs is determined by number of channels in the input

    channelCountMode = "explicit";
    channelInterpretation = "discrete";

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

The channelCount is set equal to numberOfOutputs. There are channelCount constraints and channelCountMode constraints for this node.

This node has no tail-time reference.

Example:

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

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

Constructor

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

BaseAudioContext context

The BaseAudioContext this new ChannelSplitterNode will be associated with.

optional ChannelSplitterNode options

Optional initial parameter value for this ChannelSplitterNode.

The ChannelMergerNode Interface

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

  numberOfInputs  : Variable N (default to 6)
  numberOfOutputs : 1

  channelCount = 1;
  channelCountMode = "explicit";
  channelInterpretation = "speakers";

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

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

There are channelCount constraints and channelCountMode constraints for this node.

This node has no tail-time reference.

Example:

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

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

Constructor

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

BaseAudioContext context

The BaseAudioContext this new ChannelMergerNode will be associated with.

optional ChannelMergerOptions options

Optional initial parameter value for this ChannelMergerNode.

The DynamicsCompressorNode Interface

DynamicsCompressorNode is an AudioNode processor implementing a dynamics compression effect.

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

    numberOfInputs  : 1
    numberOfOutputs : 1

    channelCount = 2;
    channelCountMode = "explicit";
    channelInterpretation = "speakers";

This node has no tail-time reference.

Constructor

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

BaseAudioContext context

The BaseAudioContext this new DynamicsCompressorNode will be associated with.

optional DynamicsCompressorOptions options

Optional initial parameter value for this DynamicsCompressorNode.

readonly attribute AudioParam threshold

The decibel value above which the compression will start taking effect. Its default value is -24. This parameter is k-rate. Its nominal range is [-100, 0].

readonly attribute AudioParam knee

A decibel value representing the range above the threshold where the curve smoothly transitions to the "ratio" portion. Its default value is 30. This parameter is k-rate. Its nominal range is [0, 40].

readonly attribute AudioParam ratio

The amount of dB change in input for a 1 dB change in output. Its default value is 12. This parameter is k-rate. Its nominal range is [1, 20].

readonly attribute float reduction

A read-only decibel value for metering purposes, representing the current amount of gain reduction that the compressor is applying to the signal. If fed no signal the value will be 0 (no gain reduction).

readonly attribute AudioParam attack

The amount of time (in seconds) to reduce the gain by 10dB. Its default value is 0.003. This parameter is k-rate. Its nominal range is [0, 1].

readonly attribute AudioParam release

The amount of time (in seconds) to increase the gain by 10dB. Its default value is 0.250. This parameter is k-rate. Its nominal range is [0, 1].

The BiquadFilterNode Interface

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

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

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

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

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

    numberOfInputs  : 1
    numberOfOutputs : 1

    channelCountMode = "max";
    channelInterpretation = "speakers";

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

This node has a tail-time reference such that this node continues to output non-silent audio with zero input. Since this is an IIR filter, the filter produces non-zero input forever, but in practice, this can be limited after some finite time where the output is sufficiently close to zero. The actual time depends on the filter coefficients.

lowpass

A lowpass filter allows frequencies below the cutoff frequency to pass through and attenuates frequencies above the cutoff. It implements a standard second-order resonant lowpass filter with 12dB/octave rolloff.

frequency

The cutoff frequency

Q

Controls how peaked the response will be at the cutoff frequency. A large value makes the response more peaked. Please note that for this filter type, this value is not a traditional Q, but is a resonance value in decibels.

gain

Not used in this filter type

highpass

A highpass filter is the opposite of a lowpass filter. Frequencies above the cutoff frequency are passed through, but frequencies below the cutoff are attenuated. It implements a standard second-order resonant highpass filter with 12dB/octave rolloff.

frequency

The cutoff frequency below which the frequencies are attenuated

Q

Controls how peaked the response will be at the cutoff frequency. A large value makes the response more peaked. Please note that for this filter type, this value is not a traditional Q, but is a resonance value in decibels.

gain

Not used in this filter type

bandpass

A bandpass filter allows a range of frequencies to pass through and attenuates the frequencies below and above this frequency range. It implements a second-order bandpass filter.

frequency

The center of the frequency band

Q

Controls the width of the band. The width becomes narrower as the Q value increases.

gain

Not used in this filter type

lowshelf

The lowshelf filter allows all frequencies through, but adds a boost (or attenuation) to the lower frequencies. It implements a second-order lowshelf filter.

frequency

The upper limit of the frequences where the boost (or attenuation) is applied.

Q

Not used in this filter type.

gain

The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.

highshelf

The highshelf filter is the opposite of the lowshelf filter and allows all frequencies through, but adds a boost to the higher frequencies. It implements a second-order highshelf filter

frequency

The lower limit of the frequences where the boost (or attenuation) is applied.

Q

Not used in this filter type.

gain

The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.

peaking

The peaking filter allows all frequencies through, but adds a boost (or attenuation) to a range of frequencies.

frequency

The center frequency of where the boost is applied.

Q

Controls the width of the band of frequencies that are boosted. A large value implies a narrow width.

gain

The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.

notch

The notch filter (also known as a band-stop or band-rejection filter) is the opposite of a bandpass filter. It allows all frequencies through, except for a set of frequencies.

frequency

The center frequency of where the notch is applied.

Q

Controls the width of the band of frequencies that are attenuated. A large value implies a narrow width.

gain

Not used in this filter type.

allpass

An allpass filter allows all frequencies through, but changes the phase relationship between the various frequencies. It implements a second-order allpass filter

frequency

The frequency where the center of the phase transition occurs. Viewed another way, this is the frequency with maximal group delay.

Q

Controls how sharp the phase transition is at the center frequency. A larger value implies a sharper transition and a larger group delay.

gain

Not used in this filter type.

All attributes of the BiquadFilterNode are a-rate AudioParam.

Constructor

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

BaseAudioContext context

The BaseAudioContext this new BiquadFilterNode will be associated with.

optional BiquadFilterOptions options

Optional initial parameter value for this BiquadFilterNode.

attribute BiquadFilterType type

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

readonly attribute AudioParam frequency

The frequency at which the BiquadFilterNode will operate, in Hz. Its default value is 350Hz. It forms a compound parameter with detune. Its nominal range is [0, Nyquist frequency].

readonly attribute AudioParam detune

A detune value, in cents, for the frequency. Its default value is 0. It forms a compound parameter with frequency. Its nominal range is \((-\infty, \infty)\).

readonly attribute AudioParam Q

The Q factor has a default value of 1. Its nominal range is \((-\infty, \infty)\). This is not used for lowshelf or highshelf filters.

readonly attribute AudioParam gain

The gain has a default value of 0. Its nominal range is \((-\infty, \infty)\). Its value is in dB units. The gain is only used for lowshelf, highshelf, and peaking filters.

void getFrequencyResponse()

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

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

Float32Array frequencyHz

This parameter specifies an array of frequencies at which the response values will be calculated.

Float32Array magResponse

This parameter specifies an output array receiving the linear magnitude response values.

If a value in the frequencyHz parameter is not within [0; sampleRate/2], where sampleRate is the value of the sampleRate property of the AudioContext, the corresponding value at the same index of the magResponse array MUST be NaN.

Float32Array phaseResponse

This parameter specifies an output array receiving the phase response values in radians.

If a value in the frequencyHz parameter is not within [0; sampleRate/2], where sampleRate is the value of the sampleRate property of the AudioContext, the corresponding value at the same index of the phaseResponse array MUST be NaN.

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

The IIRFilterNode Interface

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

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

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

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

    numberOfInputs  : 1
    numberOfOutputs : 1

    channelCountMode = "max";
    channelInterpretation = "speakers";

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

This node has a tail-time reference such that this node continues to output non-silent audio with zero input. Since this is an IIR filter, the filter produces non-zero input forever, but in practice, this can be limited after some finite time where the output is sufficiently close to zero. The actual time depends on the filter coefficients.

Constructor

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

BaseAudioContext context

The BaseAudioContext this new IIRFilterNode will be associated with.

IIRFilterOptions options

Optional initial parameter value for this IIRFilterNode.

void getFrequencyResponse()

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

Float32Array frequencyHz

This parameter specifies an array of frequencies at which the response values will be calculated.

Float32Array magResponse

This parameter specifies an output array receiving the linear magnitude response values.

If a value in the frequencyHz parameter is not within [0; sampleRate/2], where sampleRate is the value of the sampleRate property of the AudioContext, the corresponding value at the same index of the magResponse array MUST be NaN.

Float32Array phaseResponse

This parameter specifies an output array receiving the phase response values in radians.

If a value in the frequencyHz parameter is not within [0; sampleRate/2], where sampleRate is the value of the sampleRate property of the AudioContext, the corresponding value at the same index of the phaseResponse array MUST be NaN.

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

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

The WaveShaperNode Interface

WaveShaperNode is an AudioNode processor implementing non-linear distortion effects.

Non-linear waveshaping distortion is commonly used for both subtle non-linear warming, or more obvious distortion effects. Arbitrary non-linear shaping curves may be specified.

    numberOfInputs  : 1
    numberOfOutputs : 1

    channelCountMode = "max";
    channelInterpretation = "speakers";

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

WaveShaperNodes are created with an internal flag curve set, initially set to false.

If the oversample attribute is set to none, the WaveShaperNode has no tail-time. If the oversample attribute is set to 2x or 4x, the WaveShaperNode can have tail-time caused by the resampling technique used. The duration of this tail-time is therefore implementation-dependent.

none

Don't oversample

2x

Oversample two times

4x

Oversample four times

Constructor

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

BaseAudioContext context

The BaseAudioContext this new WaveShaperNode will be associated with.

optional WaveShaperOptions options

Optional initial parameter value for this WaveShaperNode.

attribute Float32Array? curve

The shaping curve used for the waveshaping effect. The input signal is nominally within the range [-1; 1]. Each input sample within this range will index into the shaping curve, with a signal level of zero corresponding to the center value of the curve array if there are an odd number of entries, or interpolated between the two centermost values if there are an even number of entries in the array. Any sample value less than -1 will correspond to the first value in the curve array. Any sample value greater than +1 will correspond to the last value in the curve array.

The implementation must perform linear interpolation between adjacent points in the curve. Initially the curve attribute is null, which means that the WaveShaperNode will pass its input to its output without modification.

Values of the curve are spread with equal spacing in the [-1; 1] range. This means that a curve with a even number of value will not have a value for a signal at zero, and a curve with an odd number of value will have a value for a signal at zero.

A InvalidStateError MUST be thrown if this attribute is set with a Float32Array that has a length less than 2.

When this attribute is set, an internal copy of the curve is created by the WaveShaperNode. Subsequent modifications of the contents of the array used to set the attribute therefore have no effect: the attribute must be set again in order to change the curve.

To set the curve attribute, execute these steps:

1. Let new curve be the Float32Array to be assigned to curve.
2. If new curve is not null and curve set is true, throw an InvalidStateError and abort these steps.
3. If new curve is not null, set curve set to true.
4. Assign new curve to the curve attribute.

attribute OverSampleType oversample

Specifies what type of oversampling (if any) should be used when applying the shaping curve. The default value is "none", meaning the curve will be applied directly to the input samples. A value of "2x" or "4x" can improve the quality of the processing by avoiding some aliasing, with the "4x" value yielding the highest quality. For some applications, it's better to use no oversampling in order to get a very precise shaping curve.

A value of "2x" or "4x" means that the following steps must be performed:

1. Up-sample the input samples to 2x or 4x the sample-rate of the AudioContext. Thus for each render quantum, generate 256 (for 2x) or 512 (for 4x) samples.
2. Apply the shaping curve.
3. Down-sample the result back to the sample-rate of the AudioContext. Thus taking the 256 (or 512) processed samples, generating 128 as the final result.

The exact up-sampling and down-sampling filters are not specified, and can be tuned for sound quality (low aliasing, etc.), low latency, and performance.

Use of oversampling introduces some degree of audio processing latency due to the up-sampling and down-sampling filters. The amount of this latency can vary from one implementation to another.

The OscillatorNode Interface

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

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

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

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

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

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

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

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

  numberOfInputs  : 0
  numberOfOutputs : 1 (mono output)

sine

A sine wave

square

A square wave of duty period 0.5

sawtooth

A sawtooth wave

triangle

A triangle wave

custom

A custom periodic wave

Constructor

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

BaseAudioContext context

The BaseAudioContext this new OscillatorNode will be associated with.

optional OscillatorOptions options

Optional initial parameter value for this OscillatorNode.

attribute OscillatorType type

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

readonly attribute AudioParam frequency

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

readonly attribute AudioParam detune

A detuning value (in cents) which will offset the frequency by the given amount. Its default value is 0. This parameter is a-rate. It forms a compound parameter with frequency. Its nominal range is \((-\infty, \infty)\).

void setPeriodicWave(PeriodicWave periodicWave)

Sets an arbitrary custom periodic waveform given a PeriodicWave.

The PeriodicWave Interface

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

Conforming implementations MUST support PeriodicWave up to at least 8192 elements.

Constructor

1. Let p be a new PeriodicWave object. Let [[associated context]] be a reference to the BaseAudioContext passed as first argument of this constructor.
2. If the real and imag parameters of the PeriodicWaveOptions are not of the same length, an IndexSizeError exception MUST be thrown.
3. If options has not been passed in, let [[\real]] and [[\imag]] be two internal slots of type Float32Array and length 2. Set the second element of the [[\imag]] array be 1.
   When setting this PeriodicWave on an OscillatorNode, this is equivalent to using the built-in type "sine".
4. Otherwise, let [[\real]] and [[\imag]] be two internal slots of type Float32Array, of length both equal to the maximum length of the real and imag of the attributes of the PeriodicWaveOptions passed in. Make a copy of those arrays into their respective internal slots.
5. Let [[\normalize]] be an internal slot of the PeriodicWave the is initialized to the inverse of the disableNormalization attribute of the PeriodicWaveConstraints on the PeriodicWaveOptions.
6. Return p.

BaseAudioContext context

The BaseAudioContext for which to create this PeriodicWave.

Unlike AudioBuffer, PeriodicWaves can't be shared accross AudioContexts or OfflineAudioContexts. It is associated with a particular BaseAudioContext.

optional PeriodicWaveOptions options

Optional initial parameter value for this PeriodicWave.

            $$
              x(t) = \sum_{k=1}^{L-1} \left(a[k]\cos2\pi k t + b[k]\sin2\pi k t\right)
            $$
          

          $$
            \tilde{x}(n) = \sum_{k=1}^{L-1} \left(a[k]\cos\frac{2\pi k n}{N} + b[k]\sin\frac{2\pi k n}{N}\right)
          $$
          

            $$
              f = \max_{n = 0, \ldots, N - 1} |\tilde{x}(n)|
            $$
          

            $$
              \hat{x}(n) = \frac{\tilde{x}(n)}{f}
            $$
          

The MediaStreamAudioSourceNode Interface

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

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

    numberOfInputs  : 0
    numberOfOutputs : 1

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

This node has no tail-time reference.

Constructor

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

BaseAudioContext context

The BaseAudioContext this new MediaStreamAudioSourceNode will be associated with.

MediaStreamAudioSourceOptions options

Initial parameter values for this MediaStreamAudioSourceNode. If the mediaStreamTrack parameter does not reference a MediaStreamTrack whose kind attribute has the value "audio", an InvalidStateError MUST be thrown.

[SameObject] readonly attribute MediaStream mediaStream

The MediaStream used when constructing this MediaStreamAudioSourceNode.

The MediaStreamTrackAudioSourceNode Interface

This interface represents an audio source from a MediaStreamTrack.

    numberOfInputs  : 0
    numberOfOutputs : 1

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

This node has no tail-time reference.

Constructor

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

AudioContext context

The AudioContext this new MediaStreamTrackAudioSourceNode will be associated with.

MediaStreamTrackAudioSourceOptions options

Initial parameter value for this MediaStreamTrackAudioSourceNode.

The MediaStreamAudioDestinationNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 0

    channelCount = 2;
    channelCountMode = "explicit";
    channelInterpretation = "speakers";

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

Constructor

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

BaseAudioContext context

The BaseAudioContext this new MediaStreamAudioDestinationNode will be associated with.

optional AudioNodeOptions options

Optional initial parameter value for this MediaStreamAudioDestinationNode.

readonly attribute MediaStream stream

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

Control thread and rendering thread

The Web Audio API MUST be implemented using a control thread, and a rendering thread.

The control thread is the thread from which the AudioContext is instantiated, and from which authors manipulate the audio graph, that is, from where the operation on a BaseAudioContext are invoked. The rendering thread is the thread on which the actual audio output is computed, in reaction to the calls from the control thread. It can be a real-time, callback-based audio thread, if computing audio for an AudioContext, or a normal thread if rendering and audio graph offline using an OfflineAudioContext.

The control thread uses a traditional event loop, as described in [[HTML]].

The rendering thread uses a specialized rendering loop, described in the section Rendering an audio graph

Communication from the control thread to the rendering thread is done using control message passing. Communication in the other direction is done using regular event loop tasks.

Each AudioContext has a single control message queue, that is a list of control messages that are operations running on the control thread.

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

Control messages in a control message queue are ordered by time of insertion. The oldest message is therefore the one at the front of the control message queue.

Swapping a control message queue Q_A with another control message queue Q_B means executing the following steps:

1. Let Q_C be a new, empty control message queue.
2. Move all the control messages Q_A to Q_C.
3. Move all the control messages Q_B to Q_A.
4. Move all the control messages Q_C to Q_B.

For example, successfuly calling start() on an AudioBufferSourceNode source adds a control message to the control message queue of the AudioContext source.context.

Asynchronous operations

Calling methods on AudioNodes is effectively asynchronous, and MUST to be done in two phases, a synchronous part and an asynchronous part. For each method, some part of the execution happens on the control thread (for example, throwing an exception in case of invalid parameters), and some part happens on the rendering thread (for example, changing the value of an AudioParam).

In the description of each operation on AudioNodes and AudioContexts, the synchronous section is marked with a ⌛. All the other operations are executed in parallel, as described in [[HTML]].

The synchronous section is executed on the control thread, and happens immediately. If it fails, the method execution is aborted, possibly throwing an exception. If it succeeds, a control message, encoding the operation to be executed on the rendering thread is enqueued on the control message queue of this rendering thread.

The synchronous and asynchronous sections order with respect to other events MUST be the same: given two operation A and B with respective synchronous and asynchronous section A_Sync and A_Async, and B_Sync and B_Async, if A happens before B, then A_Sync happens before B_Sync, and A_Async happens before B_Async. In other words, synchronous and asynchronous sections can't be reordered.

Rendering an audio graph

Rendering an audio graph is done in blocks of 128 samples-frames. A block of 128 samples-frames is called a render quantum.

Operations that happen atomically on a given thread can only be executed when no other atomic operation is running on another thread.

The algorithm for rendering a block of audio from an AudioContext G with a control message queue Q is as follows.

If the algorithm returns true then it MUST be executed again in the future, to render the next block of audio. Else, the rendering thread yields and the processing stops. The control thread can restart executing this algoritm if needed.

1. Let Q_rendering be an empty control message queue. Atomically swap Q_rendering and Q
2. While there are messages in Q_rendering, execute the following steps:
   1. Execute the asynchronous section of the oldest message of Q_rendering.
   2. Remove the oldest message of Q_rendering.
3. If the rendering thread state of the AudioContext is not running, return false.
4. Order the AudioNodes of the AudioContext to be processed.
   1. Let ordered be an empty list of AudioNodes. It will contain an ordered list of AudioNodes when this ordering algorithm terminates.
   2. Let nodes be the set of all nodes created by this AudioContext, and still alive.
   3. Let cycle breakers be an empty set of DelayNodes. It will contain all the DelayNodes that are part of a cycle.
   4. For each AudioNode node in nodes
      1. If node is a DelayNode that is part of a cycle, add it to cycle breakers and remove it from nodes.
   5. If nodes contains cycles, mute all the AudioNodes that are part of this cycle, and remove them from nodes.
   6. While there are unmarked AudioNodes in nodes:
      1. Choose an AudioNode node in nodes
      2. Visit node.
      Visiting a node node mean performing the following steps:
      1. If node is marked, abort these steps.
      2. Mark node.
      3. For each AudioNode input connected to node:
         1. Visit input.
   7. Add node to the beginning of ordered.
5. Reverse the order of nodes.
6. For each DelayNode in a cycle, make available for reading a block of audio from the DelayNode buffer, available for reading.
7. Compute the value(s) of the AudioListener's AudioParams for this block.
8. For each AudioNode, in the order determined previously:
   1. For each AudioParam of this AudioNode, execute these steps:
      1. If this AudioParam has any AudioNode connected to it, sum the buffers made available for reading by all AudioNode connected to this AudioParam, down mix the resulting buffer down to Mono, and call this buffer the input AudioParam buffer.
      2. Compute the value(s) of this AudioParam for this block.
   2. If this AudioNode has any AudioNodes connected to its input, sum the buffers made available for reading by all AudioNodes connected to this AudioNode. The resulting buffer is called the input buffer. Up or down-mix it to match if number of input channels of this AudioNode.
   3. If this AudioNode is a source node, compute a block of audio, and make it available for reading.
   4. Else, if this AudioNode is a destination node, record the input of this AudioNode.
   5. Else, process the input buffer, and make available for reading the resulting buffer.
9. Atomically increment currentTime by 128 / sampleRate.
10. Return true.

• Muting an AudioNode means that its output MUST be silence for the rendering of this audio block.

  Making a buffer available for reading from an AudioNode means putting it in a state where other AudioNodes connected to this AudioNode can safely read from it.

  For example, implementations can choose to allocate a new buffer, or have a more elaborate mechanism, reusing an existing buffer that is now unused.

  Recording the input of an AudioNode means copying the input data of this AudioNode for future usage.

  Computing a block of audio means running the algorithm for this AudioNode to produce 128 sample-frames.

  Processing an input buffer means running the algorithm for an AudioNode, using an input buffer and the value(s) of the AudioParam(s) of this AudioNode as the input for this algorithm.

Background

This section is non-normative. Please see AudioContext lifetime and AudioNode lifetime for normative requirements.

In addition to allowing the creation of static routing configurations, it should also be possible to do custom effect routing on dynamically allocated voices which have a limited lifetime. For the purposes of this discussion, let's call these short-lived voices "notes". Many audio applications incorporate the ideas of notes, examples being drum machines, sequencers, and 3D games with many one-shot sounds being triggered according to game play.

In a traditional software synthesizer, notes are dynamically allocated and released from a pool of available resources. The note is allocated when a MIDI note-on message is received. It is released when the note has finished playing either due to it having reached the end of its sample-data (if non-looping), it having reached a sustain phase of its envelope which is zero, or due to a MIDI note-off message putting it into the release phase of its envelope. In the MIDI note-off case, the note is not released immediately, but only when the release envelope phase has finished. At any given time, there can be a large number of notes playing but the set of notes is constantly changing as new notes are added into the routing graph, and old ones are released.

The audio system automatically deals with tearing-down the part of the routing graph for individual "note" events. A "note" is represented by an AudioBufferSourceNode, which can be directly connected to other processing nodes. When the note has finished playing, the context will automatically release the reference to the AudioBufferSourceNode, which in turn will release references to any nodes it is connected to, and so on. The nodes will automatically get disconnected from the graph and will be deleted when they have no more references. Nodes in the graph which are long-lived and shared between dynamic voices can be managed explicitly. Although it sounds complicated, this all happens automatically with no extra handling required.

Example

The low-pass filter, panner, and second gain nodes are directly connected from the one-shot sound. So when it has finished playing the context will automatically release them (everything within the dotted line). If there are no longer any references to the one-shot sound and connected nodes, then they will be immediately removed from the graph and deleted. The streaming source, has a global reference and will remain connected until it is explicitly disconnected. Here's how it might look in JavaScript:

var context = 0;
var compressor = 0;
var gainNode1 = 0;
var streamingAudioSource = 0;

// Initial setup of the "long-lived" part of the routing graph
function setupAudioContext() {
    context = new AudioContext();

    compressor = context.createDynamicsCompressor();
    gainNode1 = context.createGain();

    // Create a streaming audio source.
    var audioElement = document.getElementById('audioTagID');
    streamingAudioSource = context.createMediaElementSource(audioElement);
    streamingAudioSource.connect(gainNode1);

    gainNode1.connect(compressor);
    compressor.connect(context.destination);
}

// Later in response to some user action (typically mouse or key event)
// a one-shot sound can be played.
function playSound() {
    var oneShotSound = context.createBufferSource();
    oneShotSound.buffer = dogBarkingBuffer;

    // Create a filter, panner, and gain node.
    var lowpass = context.createBiquadFilter();
    var panner = context.createPanner();
    var gainNode2 = context.createGain();

    // Make connections
    oneShotSound.connect(lowpass);
    lowpass.connect(panner);
    panner.connect(gainNode2);
    gainNode2.connect(compressor);

    // Play 0.75 seconds from now (to play immediately pass in 0)
    oneShotSound.start(context.currentTime + 0.75);
}

Speaker Channel Layouts

When channelInterpretation is "speakers" then the up-mixing and down-mixing is defined for specific channel layouts.

Mono (one channel), stereo (two channels), quad (four channels), and 5.1 (six channels) MUST be supported. Other channel layout may be supported in future version of this specification.

Channel ordering

    Mono
      0: M: mono

    Stereo
      0: L: left
      1: R: right
    

  Quad
      0: L:  left
      1: R:  right
      2: SL: surround left
      3: SR: surround right

    5.1
      0: L:   left
      1: R:   right
      2: C:   center
      3: LFE: subwoofer
      4: SL:  surround left
      5: SR:  surround right
  

Up Mixing speaker layouts

Mono up-mix:

    1 -> 2 : up-mix from mono to stereo
        output.L = input;
        output.R = input;

    1 -> 4 : up-mix from mono to quad
        output.L = input;
        output.R = input;
        output.SL = 0;
        output.SR = 0;

    1 -> 5.1 : up-mix from mono to 5.1
        output.L = 0;
        output.R = 0;
        output.C = input; // put in center channel
        output.LFE = 0;
        output.SL = 0;
        output.SR = 0;

Stereo up-mix:

    2 -> 4 : up-mix from stereo to quad
        output.L = input.L;
        output.R = input.R;
        output.SL = 0;
        output.SR = 0;

    2 -> 5.1 : up-mix from stereo to 5.1
        output.L = input.L;
        output.R = input.R;
        output.C = 0;
        output.LFE = 0;
        output.SL = 0;
        output.SR = 0;

Quad up-mix:

    4 -> 5.1 : up-mix from quad to 5.1
        output.L = input.L;
        output.R = input.R;
        output.C = 0;
        output.LFE = 0;
        output.SL = input.SL;
        output.SR = input.SR;

Down Mixing speaker layouts

A down-mix will be necessary, for example, if processing 5.1 source material, but playing back stereo.

  Mono down-mix:

      2 -> 1 : stereo to mono
          output = 0.5 * (input.L + input.R);

      4 -> 1 : quad to mono
          output = 0.25 * (input.L + input.R + input.SL + input.SR);

      5.1 -> 1 : 5.1 to mono
          output = sqrt(0.5) * (input.L + input.R) + input.C + 0.5 * (input.SL + input.SR)


  Stereo down-mix:

      4 -> 2 : quad to stereo
          output.L = 0.5 * (input.L + input.SL);
          output.R = 0.5 * (input.R + input.SR);

      5.1 -> 2 : 5.1 to stereo
          output.L = L + sqrt(0.5) * (input.C + input.SL)
          output.R = R + sqrt(0.5) * (input.C + input.SR)

  Quad down-mix:

      5.1 -> 4 : 5.1 to quad
          output.L = L + sqrt(0.5) * input.C
          output.R = R + sqrt(0.5) * input.C
          output.SL = input.SL
          output.SR = input.SR

  

Background

A common feature requirement for modern 3D games is the ability to dynamically spatialize and move multiple audio sources in 3D space. Game audio engines such as OpenAL, FMOD, Creative's EAX, Microsoft's XACT Audio, etc. have this ability.

Using an PannerNode, an audio stream can be spatialized or positioned in space relative to an AudioListener. An AudioContext will contain a single AudioListener. Both panners and listeners have a position in 3D space using a right-handed cartesian coordinate system. The units used in the coordinate system are not defined, and do not need to be because the effects calculated with these coordinates are independent/invariant of any particular units such as meters or feet. PannerNode objects (representing the source stream) have an orientation vector representing in which direction the sound is projecting. Additionally, they have a sound cone representing how directional the sound is. For example, the sound could be omnidirectional, in which case it would be heard anywhere regardless of its orientation, or it can be more directional and heard only if it is facing the listener. AudioListener objects (representing a person's ears) have an orientation and up vector representing in which direction the person is facing.

During rendering, the PannerNode calculates an azimuth and elevation. These values are used internally by the implementation in order to render the spatialization effect. See the Panning Algorithm section for details of how these values are used.

Azimuth and Elevation

The following algorithm must be used to calculate the azimuth and elevation for the PannerNode:

  // Calculate the source-listener vector.
  vec3 sourceListener = source.position - listener.position;

  if (sourceListener.isZero()) {
      // Handle degenerate case if source and listener are at the same point.
      azimuth = 0;
      elevation = 0;
      return;
  }

  sourceListener.normalize();

  // Align axes.
  vec3 listenerFront = listener.orientation;
  vec3 listenerUp = listener.up;
  vec3 listenerRight = listenerFront.cross(listenerUp);
  listenerRight.normalize();

  vec3 listenerFrontNorm = listenerFront;
  listenerFrontNorm.normalize();

  vec3 up = listenerRight.cross(listenerFrontNorm);

  float upProjection = sourceListener.dot(up);

  vec3 projectedSource = sourceListener - upProjection * up;
  projectedSource.normalize();

  azimuth = 180 * acos(projectedSource.dot(listenerRight)) / PI;

  // Source in front or behind the listener.
  float frontBack = projectedSource.dot(listenerFrontNorm);
  if (frontBack < 0)
      azimuth = 360 - azimuth;

  // Make azimuth relative to "front" and not "right" listener vector.
  if ((azimuth >= 0) && (azimuth <= 270))
      azimuth = 90 - azimuth;
  else
      azimuth = 450 - azimuth;

  elevation = 90 - 180 * acos(sourceListener.dot(up)) / PI;

  if (elevation > 90)
      elevation = 180 - elevation;
  else if (elevation < -90)
      elevation = -180 - elevation;
  

Panning Algorithm

Mono-to-stereo and stereo-to-stereo panning must be supported. Mono-to-stereo processing is used when all connections to the input are mono. Otherwise stereo-to-stereo processing is used.

PannerNode "HRTF" Panning (stereo only)

This requires a set of HRTF (Head-related Transfer Function) impulse responses recorded at a variety of azimuths and elevations. The implementation requires a highly optimized convolution function. It is somewhat more costly than "equalpower", but provides more perceptually spatialized sound.

Distance Effects

Sounds which are closer are louder, while sounds further away are quieter. Exactly how a sound's volume changes according to distance from the listener depends on the distanceModel attribute.

During audio rendering, a distance value will be calculated based on the panner and listener positions according to:

  function dotProduct(v1, v2) {
    var d = 0;
    for (var i = 0; i < Math.min(v1.length, v2.length); i++)
      d += v1[i] * v2[i];
    return d;
  }
  var v = panner.position - listener.position;
  var distance = Math.sqrt(dotProduct(v, v));
  

distance will then be used to calculate distanceGain which depends on the distanceModel attribute. See the distanceModel section for details of how this is calculated for each distance model. The value computed by the distanceModel equations are to be clamped to [0, 1].

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

Sound Cones

The listener and each sound source have an orientation vector describing which way they are facing. Each sound source's sound projection characteristics are described by an inner and outer "cone" describing the sound intensity as a function of the source/listener angle from the source's orientation vector. Thus, a sound source pointing directly at the listener will be louder than if it is pointed off-axis. Sound sources can also be omni-directional.

The following algorithm must be used to calculate the gain contribution due to the cone effect, given the source (the PannerNode) and the listener:

function dotProduct(v1, v2) {
  var d = 0;
  for (var i = 0; i < Math.min(v1.length, v2.length); i++)
    d += v1[i] * v2[i];
  return d;
}

function diff(v1, v2) {
  var v = [];
  for (var i = 0; i & lt; Math.min(v1.length, v2.length); i++)
    v[i] = v1[i] - v2[i];
  return v;
}

function coneGain() {
  if (dotProduct(source.orientation, source.orientation) == 0 || ((source.coneInnerAngle ==
      360) && (source.coneOuterAngle == 360)))
    return 1; // no cone specified - unity gain

  // Normalized source-listener vector
  var sourceToListener = diff(listener.position, source.position);
  sourceToListener.normalize();

  var normalizedSourceOrientation = source.orientation;
  normalizedSourceOrientation.normalize();

  // Angle between the source orientation vector and the source-listener vector
  var dotProduct = dotProduct(sourceToListener, normalizedSourceOrientation);
  var angle = 180 * Math.acos(dotProduct) / Math.PI;
  var absAngle = Math.abs(angle);

  // Divide by 2 here since API is entire angle (not half-angle)
  var absInnerAngle = Math.abs(source.coneInnerAngle) / 2;
  var absOuterAngle = Math.abs(source.coneOuterAngle) / 2;
  var gain = 1;

  if (absAngle <= absInnerAngle) {
    // No attenuation
    gain = 1;
  } else if (absAngle >= absOuterAngle) {
    // Max attenuation
    gain = source.coneOuterGain;
  } else {
    // Between inner and outer cones
    // inner -> outer, x goes from 0 -> 1
    var x = (absAngle - absInnerAngle) / (absOuterAngle - absInnerAngle);
    gain = (1 - x) + source.coneOuterGain * x;
  }

  return gain;
}  

Audio Buffer Copying

When an acquire the content operation is performed on an AudioBuffer, the entire operation can usually be implemented without copying channel data. In particular, the last step should be performed lazily at the next getChannelData call. That means a sequence of consecutive acquire the contents operations with no intervening getChannelData (e.g. multiple AudioBufferSourceNodes playing the same AudioBuffer) can be implemented with no allocations or copying.

Implementations can perform an additional optimization: if getChannelData is called on an AudioBuffer, fresh ArrayBuffers have not yet been allocated, but all invokers of previous acquire the content operations on an AudioBuffer have stopped using the AudioBuffer's data, the raw data buffers can be recycled for use with new AudioBuffers, avoiding any reallocation or copying of the channel data.

Audio Glitching

Audio glitches are caused by an interruption of the normal continuous audio stream, resulting in loud clicks and pops. It is considered to be a catastrophic failure of a multi-media system and must be avoided. It can be caused by problems with the threads responsible for delivering the audio stream to the hardware, such as scheduling latencies caused by threads not having the proper priority and time-constraints. It can also be caused by the audio DSP trying to do more work than is possible in real-time given the CPU's speed.