The AudioContext Interface

This interface represents a set of AudioNode objects and their connections. It allows for arbitrary routing of signals to the AudioDestinationNode (what the user ultimately hears). Nodes are created from the context and are then connected together. In most use cases, only a single AudioContext is used per document.

suspended

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

running

Audio is being processed.

closed

The AudioContext 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 or decodeAudioData.)

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 AudioContext 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.

readonly attribute double currentTime

This is a time in seconds which starts at zero when the context is created and increases in real-time. All scheduled times are relative to it. This is not a "transport" time which can be started, paused, and re-positioned. It is always moving forward. A GarageBand-like timeline transport system can be very easily built on top of this (in JavaScript). This time corresponds to an ever-increasing hardware timestamp.

readonly attribute AudioListener listener

An AudioListener which is used for 3D spatialization.

readonly attribute AudioContextState state

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

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

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

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

Promise<void> suspend()

Suspends the progression of time in the audio context, 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 let the audio hardware power down. The promise resolves when the frame buffer is empty (has been handed off to the hardware), or immediately (with no other effect) if the context is already suspended. The promise is rejected if the context has been closed. This method will cause an INVALID_STATE_ERR exception to be thrown if called on an OfflineAudioContext.

While the system is suspended, MediaStreams will have their output ignored; that is, data will be lost by the real time nature of media streams. HTMLMediaElements will similarly have their output ignored until the system is resumed. Audio Workers and ScriptProcessorNodes will simply not fire their onaudioprocess events while suspended, but will resume when resumed. For the purpose of AnalyserNode window functions, the data is considered as a continuous stream - i.e. the resume()/suspend() does not cause silence to appear in the AnalyserNode's stream of data.

Promise<void> resume()

Resumes the progression of time in an audio context that has been suspended, which may involve re-priming the frame buffer contents. The promise resolves when the system has re-acquired (if necessary) access to audio hardware and has begun streaming to the destination, or immediately (with no other effect) if the context is already running. The promise is rejected if the context has been closed. If the context is not currently suspended, the promise will resolve. This method will cause an INVALID_STATE_ERR exception to be thrown if called on an OfflineAudioContext.

Promise<void> close()

Closes the audio context, releasing any system audio resources used by the AudioContext. This will not automatically release all AudioContext-created objects, unless other references have been released as well; however, it will forcibly release any system audio resources that might prevent additional AudioContexts from being created and used, suspend the progression of audio time in the audio context, and stop processing audio data. The promise resolves when all AudioContext-creation-blocking resources have been released. This method will cause an INVALID_STATE_ERR exception to be thrown if called on an OfflineAudioContext.

attribute EventHandler onstatechange

A property used to set the EventHandler for an event that is dispatched to AudioContext 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.

AudioBuffer createBuffer()

Creates an AudioBuffer of the given size. The audio data in the buffer will be zero-initialized (silent). An 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 22050 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.

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 following steps must be performed:

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

AudioBufferSourceNode createBufferSource()

Creates an AudioBufferSourceNode.

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()

MediaStream mediaStream

The media stream that will act as source.

MediaStreamAudioDestinationNode createMediaStreamDestination()

Creates a MediaStreamAudioDestinationNode

Promise<AudioWorker> createAudioWorker()

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

DOMString scriptURL

This parameter represents the URL of the script to be loaded as an AudioWorker node factory. See AudioWorker section for more detail.

ScriptProcessorNode createScriptProcessor()

This method is DEPRECATED, as it is intended to be replaced by createAudioWorker. Creates a ScriptProcessorNode for direct audio processing using JavaScript. An IndexSizeError exception MUST be thrown if bufferSize or numberOfInputChannels or numberOfOutputChannels are outside the valid range.

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.

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.

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.

It is invalid for both numberOfInputChannels and numberOfOutputChannels to be zero.

AnalyserNode createAnalyser()

Create an AnalyserNode.

GainNode createGain()

Create an GainNode.

DelayNode createDelay()

Creates a DelayNode representing a variable delay line. 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()

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

WaveShaperNode createWaveShaper()

Creates a WaveShaperNode representing a non-linear distortion.

PannerNode createPanner()

Creates a PannerNode.

StereoPannerNode createStereoPanner()

Creates a StereoPannerNode.

ConvolverNode createConvolver()

Creates a ConvolverNode.

ChannelSplitterNode createChannelSplitter()

Creates an 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()

Creates 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()

Creates a DynamicsCompressorNode

OscillatorNode createOscillator()

Creates an OscillatorNode

PeriodicWave createPeriodicWave()

Creates a PeriodicWave representing a waveform containing arbitrary harmonic content. The real and imag parameters must be of type Float32Array (described in [[!TYPED-ARRAYS]]) of equal lengths greater than zero and less than or equal to 2048 or an IndexSizeError exception MUST be thrown. These parameters specify the Fourier coefficients of a Fourier series representing the partials of a periodic waveform. The created PeriodicWave will be used with an OscillatorNode and will represent a normalized time-domain waveform having maximum absolute peak value of 1. Another way of saying this is that the generated waveform of an OscillatorNode will have maximum peak value at 0dBFS. Conveniently, this corresponds to the full-range of the signal values used by the Web Audio API. Because the PeriodicWave will be normalized on creation, the real and imag parameters represent relative values.

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

Float32Array real

The real parameter represents an array of cosine terms (traditionally the A terms). In audio terminology, the first element (index 0) is the DC-offset of the periodic waveform. The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on. The first element is ignored and implementations must set it to zero internally.

Float32Array imag

The imag parameter represents an array of sine terms (traditionally the B terms). The first element (index 0) should be set to zero (and will be ignored) since this term does not exist in the Fourier series. The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.

AudioBuffer decodedData

The AudioBuffer containing the decoded audio data.

DOMException error

The error that occurred while decoding.

The OfflineAudioContext Interface

OfflineAudioContext is a particular type of AudioContext 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.

Each OfflineAudioContext instance has an associated rendering started flag that is initially false.

Promise<AudioBuffer> startRendering()

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

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

The following steps must be performed:

1. If this instance's rendering started flag is true, return a promise rejected with a DOMException whose name is "InvalidStateError".
2. Set this instance's rendering started flag to true.
3. Let promise be a new promise.
4. Asynchronously perform the following steps:
   1. Let buffer be a new AudioBuffer, with a number of channels, length and sample rate equal respectively to the numberOfChannels, length and sampleRate parameters used when this instance's constructor was called.
   2. Given the current connections and scheduled changes, start rendering length sample-frames of audio into buffer.
   3. Once the rendering is complete,
      1. Resolve promise with buffer.
      2. Fire a simple event named complete at this instance, using an instance of OfflineAudioCompletionEvent whose renderedBuffer property is set to buffer.
5. Return promise.

attribute EventHandler oncomplete

An EventHandler of type OfflineAudioCompletionEvent.

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. An AudioDestinationNode has one input and no outputs and represents the final destination to the audio hardware. 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.

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 44.1KHz.

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.

void 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 AudioContext.

optional unsigned long output = 0

The output parameter is an index describing which output of the AudioNode from which to connect. If this paremeter is out-of-bound, an IndexSizeError exception MUST be thrown. It is possible to connect an AudioNode output to more than one input with multiple calls to connect(). Thus, "fan-out" is supported.

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);
    

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.

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 AudioNode.

void disconnect()

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()

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()

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()

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()

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()

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 AudioContext context

The AudioContext 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 AudioNode, like the ChannelMergerNode and the AudioWorkerNode 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 AudioWorkerNode.

attribute unsigned long channelCount

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, the implementation MUST throw a NotSupportedError exception.

See the section for more information on this attribute.

attribute 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.

See the section for more information on this attribute.

attribute ChannelInterpretation channelInterpretation

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

See the section for more information on this attribute.

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.

      numberOfInputs  : 1
      numberOfOutputs : 0

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

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. If this value is 0, then this indicates that channelCount may not be changed. This will be the case for an AudioDestinationNode in an OfflineAudioContext and also for basic implementations with hardware support for stereo output only.

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

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

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 128 sample-frames in each block.

For each 128 sample-frame block, 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.

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:

• If one of these events is added at a time where there is already an event of the exact same type, then the new event will replace the old one.
• If one of these events is added at a time where there is already one or more events of a different type, 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. If value is set during a time when there are any automation events scheduled then it will be ignored and no exception will be thrown.

readonly attribute float defaultValue

Initial value for the value attribute.

void setValueAtTime(float value, double startTime)

Schedules a parameter value change at the given time.

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

The startTime parameter is the time in the same time coordinate system as the AudioContext's currentTime attribute. An InvalidAccessError exception MUST be thrown if startTime is negative or is not a finite number.

If there are no more events after this SetValue event, then for t >= startTime, v(t) = value. In other words, the value will remain constant.

If the next event (having time T1) after this SetValue event is not of type LinearRampToValue or ExponentialRampToValue, then, for t:

      startTime <= t < T1,  v(t) = value
    

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 details below.

void linearRampToValueAtTime(float value, double endTime)

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

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

The endTime parameter is the time in the same time coordinate system as the AudioContext's currentTime attribute. An InvalidAccessError exception MUST be thrown if endTime is negative or is not a finite number.

The value during the time interval T0 <= t < T1 (where T0 is the time of the previous event and T1 is the endTime parameter passed into this method) will be calculated as:

      v(t) = V0 + (V1 - V0) * ((t - T0) / (T1 - T0))
    

Where V0 is the value at the time T0 and V1 is the value parameter passed into this method.

If there are no more events after this LinearRampToValue event then for t >= T1, v(t) = V1.

void 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 parameter is the value the parameter will exponentially ramp to at the given time. A NotSupportedError exception MUST be thrown if this value is less than or equal to 0, or if the value at the time of the previous event is less than or equal to 0.

The endTime parameter is the time in the same time coordinate system as the AudioContext's currentTime attribute. An InvalidAccessError exception MUST be thrown if endTime is negative or is not a finite number.

The value during the time interval T0 <= t < T1 (where T0 is the time of the previous event and T1 is the endTime parameter passed into this method) will be calculated as:

        v(t) = V0 * (V1 / V0) ^ ((t - T0) / (T1 - T0))
    

Where V0 is the value at the time T0 and V1 is the value parameter passed into this method.

If there are no more events after this ExponentialRampToValue event then for t >= T1, v(t) = V1

void 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.

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

The startTime parameter is the time in the same time coordinate system as the AudioContext's currentTime attribute. An InvalidAccessError exception MUST be thrown if start is negative or is not a finite number.

The timeConstant parameter is 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.

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: T0 <= t < T1, where T0 is the startTime parameter and T1 represents the time of the event following this event (or infinity if there are no following events):

    v(t) = V1 + (V0 - V1) * exp(-(t - T0) / timeConstant)
    

Where V0 is the initial value (the .value attribute) at T0 (the startTime parameter) and V1 is equal to the target parameter.

void setValueCurveAtTime(Float32Array 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.

The values parameter is a Float32Array representing a parameter value curve. These values will apply starting at the given time and lasting for the given duration. Any modification to the the array used as values argument after the call won't have any effect on the AudioParam.

The startTime parameter is the time in the same time coordinate system as the AudioContext's currentTime attribute. An InvalidAccessError exception MUST be thrown if startTime is negative or is not a finite number.

The duration parameter is the amount of time in seconds (after the time parameter) where values will be calculated according to the values parameter.

During the time interval: startTime <= t < startTime + duration, values will be calculated:

        v(t) = values[N * (t - startTime) / duration]
      

where N is the length of the values array.

After the end of the curve time interval (t >= startTime + duration), the value will remain constant at the final curve value, until there is another automation event (if any).

void cancelScheduledValues(double startTime)

Cancels all scheduled parameter changes with times greater than or equal to startTime.

The startTime parameter is the starting time at and after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the AudioContext's currentTime attribute. An InvalidAccessError exception MUST be thrown if startTime is negative or is not a finite number.

AudioParam Automation Example

var t0 = 0;
var t1 = 0.1;
var t2 = 0.2;
var t3 = 0.3;
var t4 = 0.4;
var t5 = 0.6;
var t6 = 0.7;
var t7 = 1.0;

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

param.setValueAtTime(0.2, t0);
param.setValueAtTime(0.3, t1);
param.setValueAtTime(0.4, t2);
param.linearRampToValueAtTime(1, t3);
param.linearRampToValueAtTime(0.15, t4);
param.exponentialRampToValueAtTime(0.75, t5);
param.exponentialRampToValueAtTime(0.05, t6);
param.setValueCurveAtTime(curve, t6, t7 - t6);

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.

The implementation must make gain changes to the audio stream smoothly, without introducing noticeable clicks or glitches. This process is called "de-zippering".

readonly attribute AudioParam gain

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

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 delay time is changed, the implementation must make the transition smoothly, without introducing noticeable clicks or glitches to the audio stream.

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 128 frames (one block).

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 IEEE 32-bit linear PCM with a nominal range of -1 -> +1. 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.

readonly attribute float sampleRate

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

readonly attribute long length

Length of the PCM audio data in sample-frames.

readonly attribute double duration

Duration of the PCM audio data in seconds.

readonly attribute long numberOfChannels

The number of discrete audio channels.

Float32Array getChannelData()

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

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.

Float32Array destination

The array the channel data will be copied to.

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.

Float32Array source

The array the channel data will be copied from.

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 any of the AudioBuffer's ArrayBuffer have been neutered, abort these steps, and return a zero-length channel data buffers to the invoker.
2. Neuter all ArrayBuffers for arrays previously returned by getChannelData on this AudioBuffer.
3. Retain the underlying data buffers 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.

The AudioBufferSourceNode Interface

This interface represents an audio source from an in-memory audio asset in an AudioBuffer. It is useful for playing short audio assets which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways). The start() method is used to schedule when sound playback will happen. 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. start() and stop() may not be issued multiple times for a given AudioBufferSourceNode.

  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.

attribute AudioBuffer? buffer

Represents the audio asset to be played.

readonly attribute AudioParam playbackRate

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

readonly attribute AudioParam detune

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

attribute boolean loop

Indicates if the audio data should play in a loop. The default value is false.

attribute double loopStart

An optional value in seconds where looping should begin if the loop attribute is true. Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer.

attribute double loopEnd

An optional value in seconds where looping should end 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.

void start()

Schedules a sound to playback at an exact time.

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. start may only be called one time and must be called before stop is called or an InvalidStateError exception MUST be thrown. An InvalidAccessError exception MUST be thrown if when is negative or is not a finite number.

optional double offset = 0

The offset parameter describes the offset time in the buffer (in seconds) where playback will begin. If 0 is passed in for this value, then playback will start from the beginning of the buffer. An InvalidAccessError exception MUST be thrown if offset is negative or is not a finite number.

optional double duration

The duration parameter describes the duration of the portion (in seconds) to be played. If this parameter is not passed, the duration will be equal to the total duration of the AudioBuffer minus the offset parameter. Thus if neither offset nor duration are specified then the implied duration is the total duration of the AudioBuffer. An InvalidAccessError exception MUST be thrown if duration is negative or is not a finite number.

void stop()

Schedules a sound to stop playback at an exact time.

optional double when = 0

The when parameter describes at what time (in seconds) the sound should stop playing. It is in the same time coordinate system as the AudioContext's currentTime attribute. If 0 is passed in for this value or if the value is less than currentTime, then the sound will stop playing immediately. An InvalidAccessError exception MUST be thrown if when is negative or is not a finite number. If stop is called again after already have been called, the last invocation will be the only one applied; stop times set by previous calls will not be applied, unless the buffer has already stopped prior to any subsequent calls. If the buffer has already stopped, further calls to stop will have no effect. If a stop time is reached prior to the scheduled start time, the sound will not play.

attribute EventHandler onended

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

Both playbackRate and detune are k-rate parameters and are used together to determine a computedPlaybackRate value:

  computedPlaybackRate(t) = playbackRate(t) * pow(2, detune(t) / 1200)

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. If the .src attribute is not set, then the number of channels output will be one silent channel.

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 AudioWorker interface

An AudioWorker object is the main-thread representation of a worker "thread" that supports processing of audio in Javascript. This AudioWorker object is a factory that is used to create multiple audio nodes of the same type; this enables easy sharing of code, program data and global state across nodes. An AudioWorker can then be used to create instances of AudioWorkerNode, which is the main-thread representation of an individual node processed by that AudioWorker.

These main thread objects cause the instantiation of a Web Worker-like thread context, which executes in the audio thread. All audio processing by AudioWorkerNodes run in the audio processing thread. This has a few side effects that bear mentioning: blocking the audio worker's thread can cause glitches in the audio, and if the audio thread is normally elevated in thread priority (to reduce glitching possibility), it must be demoted to normal thread priority (in order to avoid escalating thread priority of user-supplied script code).

From inside an audio worker script, the Audio Worker factory is represented by an AudioWorkerGlobalScope object representing the node's contextual information, and individual audio nodes created by the factory are represented by AudioWorkerNodeScope objects.

In addition, all AudioWorkerNodes that are created by the same AudioWorker share an AudioWorkerGlobalScope; this can allow them to share context and data across nodes (for example, loading a single instance of a shared database used by the individual nodes, or sharing context in order to implement oscillator synchronization).

void terminate()

The terminate() method, when invoked, must cause the "terminate a worker" algorithm to be run on the worker with which the object is associated. This will cease any AudioProcessEvents being dispatched inside the AudioWorker's associated AudioWorkerGlobalScope, will cause all associated AudioWorkerNodes to cease processing, and will cause the destruction of the worker thread. In practical terms, this means all nodes created from this AudioWorker will disconnect themselves, and will cease performing any useful functions.

void postMessage(any message, optional sequence<Transferable> transfer)

postMessage may be called to send a message to the AudioWorkerGlobalScope, via the algorithm defined by the Worker specification.

readonly attribute AudioWorkerParamDescriptor[] parameters

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

attribute EventHandler onmessage

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

attribute EventHandler onloaded

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

AudioWorkerNode createNode(int numberOfInputs, int numberOfOutputs)

Creates a node instance in the audio worker.

AudioParam addParameter(DOMString name, float defaultValue)

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

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

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

void removeParameter(DOMString name)

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

The name parameter identifies the parameter to be removed.

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

As the AudioWorker interface inherits from Worker, AudioWorkers must implement the Worker interface for communication with the audio worker script.

    numberOfInputs  : variable
    numberOfOutputs : variable

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

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

var bitcrusherFactory = null;
audioContext.createAudioWorker("bitcrusher_worker.js").then( function(factory) 
    {  // cache 'factory' in case you want to create more nodes!
      bitcrusherFactory = factory;
      var bitcrusherNode = factory.createNode();
      bitcrusherNode.bits.setValueAtTime(8,0);
      bitcrusherNode.connect(output); 
      input.connect(bitcrusherNode);
    }
  );

// Custom parameter - number of bits to crush down to - default 8
this.addParameter( "bits", 8 );

// Custom parameter - frequency reduction, 0-1, default 0.5
this.addParameter( "frequencyReduction", 0.5 );

onnodecreate=function(e) {
  e.node.phaser = 0;
  e.node.lastDataValue = 0;
}

onaudioprocess= function (e) {
  for (var channel=0; channel<e.inputs[0].length; channel++) {
    var inputBuffer = e.inputs[0][channel];
    var outputBuffer = e.outputs[0][channel];
    var bufferLength = inputBuffer.length;
    var bitsArray = e.parameters.bits;
    var frequencyReductionArray = e.parameters.frequencyReduction;

    for (var i=0; i<bufferLength; i++) {
      var bits = bitsArray ? bitsArray[i] : 8;
      var frequencyReduction = frequencyReductionArray ? frequencyReductionArray[i] : 0.5;

      var step = Math.pow(1/2, bits);
      e.node.phaser += frequencyReduction;
      if (e.node.phaser >= 1.0) {
          e.node.phaser -= 1.0;
          e.node.lastDataValue = step * Math.floor(inputBuffer[i] / step + 0.5);
      }
      outputBuffer[i] = e.node.lastDataValue;
    }
  }
};

function setupNodeMessaging(node) {
  // This handles communication back from the volume meter
  node.onmessage = function (event) {
    if (event.data instanceof Object ) {
      if (event.data.hasOwnProperty("clip")
        this.clip = event.data.clip;
      if (event.data.hasOwnProperty("volume")
        this.volume = event.data.volume;
    }
  }

  // Set up some default configuration parameters
  node.postMessage(
    { "smoothing": 0.9,   // Smoothing parameter
      "clipLevel": 0.9,   // Level to consider "clipping"
      "clipLag": 750,     // How long to keep "clipping" lit up after clip (ms)
      "updating": 100      // How frequently to update volume and clip param (ms)
    });

  // Set up volume and clip attributes.  These will be updated by our onmessage.
  node.volume = 0;
  node.clip = false;
}

var vuNode = null;

audioContext.createAudioWorker("vu_meter_worker.js").then( function(factory) 
    {  // cache 'factory' in case you want to create more nodes!
      vuFactory = factory;
      vuNode = factory.createNode([1], []); // we don't need an output, and let's force to mono
      setupNodeMessaging(vuNode);
    }
  );

window.requestAnimationFrame( function(timestamp) {
  if (vuNode) {
  // Draw a bar based on vuNode.volume and vuNode.clip
  }
});

// Custom parameter - number of bits to crush down to - default 8
this.addParameter( "bits", 8 );

// Custom parameter - frequency reduction, 0-1, default 0.5
this.addParameter( "frequencyReduction", 0.5 );

onnodecreate=function(e) {
  e.node.timeToNextUpdate = 0.1 * sampleRate;
  e.node.smoothing = 0.5;
  e.node.clipLevel = 0.95;
  e.node.clipLag = 1;
  e.node.updatingInterval = 150;
  // This just handles setting attribute values
  e.node.onmessage = function ( event ) {
    if (event.data instanceof Object ) {
      if (event.data.hasOwnProperty("smoothing")
        this.smoothing = event.data.smoothing;
      if (event.data.hasOwnProperty("clipLevel")
        this.clipLevel = event.data.clipLevel;
      if (event.data.hasOwnProperty("clipLag")
        this.clipLag = event.data.clipLag / 1000;  // convert to seconds
      if (event.data.hasOwnProperty("updating")    // convert to samples
        this.updatingInterval = event.data.updating * sampleRate / 1000 ;
    }
  };
}

onaudioprocess = function ( event ) {
  var buf = event.inputs[0][0];  // Node forces mono
  var bufLength = buf.length;
  var sum = 0;
  var x;

  // Do a root-mean-square on the samples: sum up the squares...
  for (var i=0; i<bufLength; i++) {
    x = buf[i];
    if (Math.abs(x)>=event.node.clipLevel) {
      event.node.clipping = true;
      event.node.unsentClip = true;  // Make sure, for every clip, we send a message.
      event.node.lastClip = event.playbackTime + (i/sampleRate);
    }
    sum += x * x;
  }

  // ... then take the square root of the sum.
  var rms =  Math.sqrt(sum / bufLength);

  // Now smooth this out with the smoothing factor applied
  // to the previous sample - take the max here because we
  // want "fast attack, slow release."
  event.node.volume = Math.max(rms, event.node.volume*event.node.smoothing);
  if (event.node.clipping && (!event.node.unsentClip) && (event.playbackTime > (this.lastClip + clipLag)))
    event.node.clipping = false;

  // How long has it been since our last update?
  event.node.timeToNextUpdate -= event.node.last;
  if (event.node.timeToNextUpdate<0) {
    event.node.timeToNextUpdate = event.node.updatingInterval;
    event.node.postMessage(
      { "volume": event.node.volume,
        "clip": event.node.clipping });
    event.node.unsentClip = false;
  }
};

var mergerNode = audioContext.createAudioWorker("merger_worker.js", [1,1,1,1,1,1], [6] );

var mergerFactory = null;

audioContext.createAudioWorker("merger_worker.js").then( function(factory) 
    {  // cache 'factory' in case you want to create more nodes!
      mergerFactory = factory;
      var merger6channelNode = factory.createNode( [1,1,1,1,1,1], [6] );
      // connect inputs and outputs here
    }
  );

onaudioprocess= function (e) {
  for (var input=0; input<e,node.inputs.length; input++)
    e.node.outputs[0][input].set(e.node.inputs[input][0]);
};

The AudioWorkerNodeCreationEvent Interface

This is an Event object which is dispatched to AudioWorkerGlobalScope objects when a new node instance is created. This allows AudioWorkers to initialize any node-local data (e.g. allocating a delay or initializing local variables).

readonly attribute AudioWorkerNodeProcessor node

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

readonly attribute Array inputs

An array of channelCounts for the inputs.

readonly attribute Array outputs

An array of channelCounts for the outputs.

The AudioProcessEvent Interface

This is an Event object which is dispatched to AudioWorkerGlobalScope objects to perform processing.

The event handler processes audio from the input (if any) by accessing the audio data from the inputBuffers attribute. The audio data which is the result of the processing (or the synthesized data if there are no inputs) is then placed into the outputBuffers.

readonly attribute double playbackTime

The time when the audio will be played in the same time coordinate system as the AudioContext's currentTime. playbackTime allows for very tight synchronization between processing directly in JavaScript with the other events in the context's rendering graph.

readonly attribute AudioWorkerNodeProcessor

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

readonly attribute Float32Array[][] inputs

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

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

readonly attribute Float32Array[][] outputs

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

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

readonly attribute object parameters

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

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, but a NotSupportedError MUST be thrown if an attempt is made to set channelCount to a value greater than 2 or if channelCountMode is set to "max".

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

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

equalpower

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

HRTF

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

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".

linear

A linear distance model which calculates distanceGain according to:

    1 - rolloffFactor * (distance - refDistance) / (maxDistance - refDistance)
  

inverse

An inverse distance model which calculates distanceGain according to:

    refDistance / (refDistance + rolloffFactor * (distance - refDistance))
  

exponential

An exponential distance model which calculates distanceGain according to:

    pow(distance / refDistance, -rolloffFactor)
  

attribute PanningModelType panningModel

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

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

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)

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)

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

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

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

The default value is (0,0,0)

attribute DistanceModelType distanceModel

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

attribute float refDistance

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

attribute float maxDistance

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

attribute float rolloffFactor

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

attribute float 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, and the value is used modulo 360.

attribute float 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 and the value is used modulo 360.

attribute float coneOuterGain

A parameter for directional audio sources, this is the amount of volume reduction outside of the coneOuterAngle. The default value is 0, and the value is used modulo 360.

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 AudioContext's listener. See the Spatialization/Panning section for more details about spatialization.

attribute float dopplerFactor

A constant used to determine the amount of pitch shift to use when rendering a doppler effect. The default value is 1.

attribute float speedOfSound

The speed of sound used for calculating doppler shift. The default value is 343.3.

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

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)

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).

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

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

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

The default value is (0,0,0)

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 , but a NotSupportedError will be thrown if an attempt is made to set channelCount to a value great than 2 or if channelCountMode is set to "max".

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

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 from -1 to 1. This parameter is a a-rate.

The ConvolverNode Interface

This interface represents a processing node which applies a linear convolution effect given an impulse response. Normative requirements for multi-channel convolution matrixing are described here.

    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, but a NotSupportedError MUST be thrown if an attempt is made to set channelCount to a value great than 2 or if channelCountMode is set to "max".

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 an 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.

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:

float calculateNormalizationScale(buffer)
{
    const float GainCalibration = 0.00125;
    const float GainCalibrationSampleRate = 44100;
    const float MinPower = 0.000125;

    // Normalize by RMS power.
    size_t numberOfChannels = buffer->numberOfChannels();
    size_t length = buffer->length();

    float power = 0;

    for (size_t i = 0; i < numberOfChannels; ++i) {
        float* sourceP = buffer->channel(i)->data();
        float channelPower = 0;

        int n = length;
        while (n--) {
            float sample = *sourceP++;
            channelPower += sample * sample;
        }

        power += channelPower;
    }

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

    // Protect against accidental overload.
    if (isinf(power) || isnan(power) || power < MinPower)
        power = MinPower;

    float 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 (buffer->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.

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";

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.

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.

Uint8Array array

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

void getFloatTimeDomainData()

Copies the current 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.

Float32Array array

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

void getByteTimeDomainData()

Copies the current 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.

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.

readonly attribute unsigned long frequencyBinCount

Half the FFT size.

attribute float minDecibels

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 float maxDecibels

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 float 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.

function blackmanWindow(buffer) {
  var alpha = 0.16;
  var a0 = 0.5 * (1.0 - alpha);
  var a1 = 0.5;
  var a2 = 0.5 * alpha;

  for (var i = 0; i < buffer.length; i++) {
    var x = i / buffer.length;
    buffer[i] *= a0 - a1 * Math.cos(2 * Math.PI * x) + a2 * Math.cos(4 * Math.PI * x);
  }
}
    

  function smoothingOverTime(outputBuffer, lastOutputBuffer,
                             smoothingConstant,
                             real, imag, fftSize) {
    for (var i = 0; i < outputBuffer.length; i++) {
      var magnitude = Math.sqrt(real[i] * real[i] + imag[i] * imag[i]) / fftSize;
      outputBuffer[i] = smoothingTimeConstant * lastOutputBuffer[i] + (1.0 - smoothingTimeConstant) * magnitude;
    }
  }

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 = "max";
    channelInterpretation = "speakers";

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.

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.

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.

For ChannelMergerNode, channelCount and channelCountMode properties cannot be changed. InvalidState error MUST be thrown when they changed.

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.

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";

readonly attribute AudioParam threshold

The decibel value above which the compression will start taking effect. Its default value is -24, with a nominal range of -100 to 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, with a nominal range of 0 to 40.

readonly attribute AudioParam ratio

The amount of dB change in input for a 1 dB change in output. Its default value is 12, with a nominal range of 1 to 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, with a nominal range of 0 to 1.

readonly attribute AudioParam release

The amount of time (in seconds) to increase the gain by 10dB. Its default value is 0.250, with a nominal range of 0 to 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 are a-rate parameters and are used together to determine a computedFrequency value:

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

    numberOfInputs  : 1
    numberOfOutputs : 1

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

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

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 k-rate AudioParam.

attribute BiquadFilterType type

The type of this BiquadFilterNode. 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, and its nominal range is from 10Hz to half the Nyquist frequency.

readonly attribute AudioParam detune

A detune value, in cents, for the frequency. Its default value is 0.

readonly attribute AudioParam Q

The Q factor has a default value of 1, with a nominal range of 0.0001 to 1000.

readonly attribute AudioParam gain

The gain has a default value of 0, with a nominal range of -40 to 40.

void getFrequencyResponse()

Given the current filter parameter settings, calculates the frequency response for the specified frequencies.

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.

Float32Array phaseResponse

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

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.

none

Don't oversample

2x

Oversample two times

4x

Oversample four times

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

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 processing block of 128 samples, 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.

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 (half the sample-rate) 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 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 time integral of computedFrequency.

  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

attribute OscillatorType type

The shape of the periodic waveform. It may directly be set to any of the type constant values except for "custom". 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.

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.

void start(optional double when = 0)

Defined the same as the when parameter of the AudioBufferSourceNode

void stop(optional double when = 0)

Defined as in AudioBufferSourceNode.

void setPeriodicWave(PeriodicWave periodicWave)

Sets an arbitrary custom periodic waveform given a PeriodicWave.

attribute EventHandler onended

A property used to set the EventHandler (described in HTML[[HTML]]) for the ended event that is dispatched to OscillatorNode node types. When the OscillatorNode has finished playing (i.e. its stop time has been reached), an event of type Event (described in HTML[[HTML]]) will be dispatched to the event handler.

The PeriodicWave Interface

PeriodicWave represents an arbitrary periodic waveform to be used with an OscillatorNode. Please see createPeriodicWave() and setPeriodicWave() and for more details.

The MediaStreamAudioSourceNode Interface

This interface represents an audio source from a MediaStream. The first AudioMediaStreamTrack from the MediaStream will be used as a source of audio. Those interfaces are described in [[!mediacapture-streams]].

    numberOfInputs  : 0
    numberOfOutputs : 1

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

The MediaStreamAudioDestinationNode Interface

This interface is an audio destination representing a MediaStream with a single AudioMediaStreamTrack. 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). Any connections to the input are up-mixed/down-mixed to the number of channels of the input.

readonly attribute MediaStream stream

A MediaStream containing a single AudioMediaStreamTrack with the same number of channels as the node itself.

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 JavaScript 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 JavaScript 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 = 0.7071 * (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 + 0.7071 * (input.C + input.SL)
          output.R = R + 0.7071 * (input.C + input.SR)

  Quad down-mix:

      5.1 -> 4 : 5.1 to quad
          output.L = L + 0.7071 * input.C
          output.R = R + 0.7071 * 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. Because both the source stream and the listener can be moving, they both have a velocity vector representing both the speed and direction of movement. Taken together, these two velocities can be used to generate a doppler shift effect which changes the pitch.

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.

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 "equal-power", 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:

  v = panner.position - listener.position
  

  distance = sqrt(dot(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:

  if (source.orientation.isZero() || ((source.coneInnerAngle == 0) && (source.coneOuterAngle == 0)))
      return 1; // no cone specified - unity gain

  // Normalized source-listener vector
  vec3 sourceToListener = listener.position - source.position;
  sourceToListener.normalize();

  vec3 normalizedSourceOrientation = source.orientation;
  normalizedSourceOrientation.normalize();

  // Angle between the source orientation vector and the source-listener vector
  float dotProduct = sourceToListener.dot(normalizedSourceOrientation);
  float angle = 180 * acos(dotProduct) / PI;
  float absAngle = fabs(angle);

  // Divide by 2 here since API is entire angle (not half-angle)
  float absInnerAngle = fabs(source.coneInnerAngle) / 2;
  float absOuterAngle = fabs(source.coneOuterAngle) / 2;
  float 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
      float x = (absAngle - absInnerAngle) / (absOuterAngle - absInnerAngle);
      gain = (1 - x) + source.coneOuterGain * x;
  }

  return gain;
  

Doppler Shift

• Introduces a pitch shift which can realistically simulate moving sources.
• Depends on: source / listener velocity vectors, speed of sound, doppler factor.

The following algorithm must be used to calculate the doppler shift value which is used as an additional playback rate scalar for all AudioBufferSourceNodes connecting directly or indirectly to the PannerNode:

  float dopplerShift = 1; // Initialize to default value
  float dopplerFactor = listener.dopplerFactor;

  if (dopplerFactor > 0) {
      float speedOfSound = listener.speedOfSound;

      // Don't bother if both source and listener have no velocity.
      if (!source.velocity.isZero() || !listener.velocity.isZero()) {
          // Calculate the source to listener vector.
          vec3 sourceToListener = source.position - listener.position;

          float sourceListenerMagnitude = sourceToListener.length();

          float listenerProjection = sourceToListener.dot(listener.velocity) / sourceListenerMagnitude;
          float sourceProjection = sourceToListener.dot(source.velocity) / sourceListenerMagnitude;

          listenerProjection = -listenerProjection;
          sourceProjection = -sourceProjection;

          float scaledSpeedOfSound = speedOfSound / dopplerFactor;
          listenerProjection = min(listenerProjection, scaledSpeedOfSound);
          sourceProjection = min(sourceProjection, scaledSpeedOfSound);

          dopplerShift = ((speedOfSound - dopplerFactor * listenerProjection) / (speedOfSound - dopplerFactor * sourceProjection));
          fixNANs(dopplerShift); // Avoid illegal values

          // Limit the pitch shifting to 4 octaves up and 3 octaves down.
          dopplerShift = min(dopplerShift, 16);
          dopplerShift = max(dopplerShift, 0.125);
      }
  }
  

Background

Convolution is a mathematical process which can be applied to an audio signal to achieve many interesting high-quality linear effects. Very often, the effect is used to simulate an acoustic space such as a concert hall, cathedral, or outdoor amphitheater. It can also be used for complex filter effects, like a muffled sound coming from inside a closet, sound underwater, sound coming through a telephone, or playing through a vintage speaker cabinet. This technique is very commonly used in major motion picture and music production and is considered to be extremely versatile and of high quality.

Each unique effect is defined by an impulse response. An impulse response can be represented as an audio file and can be recorded from a real acoustic space such as a cave, or can be synthetically generated through a great variety of techniques.

Motivation for use as a Standard

A key feature of many game audio engines (OpenAL, FMOD, Creative's EAX, Microsoft's XACT Audio, etc.) is a reverberation effect for simulating the sound of being in an acoustic space. The code used to generate this effect has generally been custom and algorithmic (generally using a hand-tweaked set of delay lines and allpass filters which feedback into each other). In nearly all cases, not only is the implementation custom, but the code is proprietary and closed-source, each company adding its own "black magic" to achieve its unique quality. Each implementation being custom with a different set of parameters makes it impossible to achieve a uniform desired effect. And the code being proprietary makes it impossible to adopt a single one of the implementations as a standard. Additionally, algorithmic reverberation effects are limited to a relatively narrow range of different effects, regardless of how the parameters are tweaked.

A convolution effect solves these problems by using a very precisely defined mathematical algorithm as the basis of its processing. An impulse response represents an exact sound effect to be applied to an audio stream and is easily represented by an audio file which can be referenced by URL. The range of possible effects is enormous.

Implementation Guide

Linear convolution can be implemented efficiently. Here are some notes describing how it can be practically implemented.

Reverb Effect (with matrixing)

This section is normative.

In the general case the source has N input channels, the impulse response has K channels, and the playback system has M output channels. Thus it's a matter of how to matrix these channels to achieve the final result.

The subset of N, M, K below must be implemented (note that the first image in the diagram is just illustrating the general case and is not normative, while the following images are normative). Without loss of generality, developers desiring more complex and arbitrary matrixing can use multiple ConvolverNode objects in conjunction with an ChannelMergerNode.

Single channel convolution operates on a mono audio input, using a mono impulse response, and generating a mono output. To achieve a more spacious sound, 2 channel audio inputs and 1, 2, or 4 channel impulse responses will be considered. The following diagram, illustrates the common cases for stereo playback where N and M are 1 or 2 and K is 1, 2, or 4.

Performance Considerations

Latency

For web applications, the time delay between mouse and keyboard events (keydown, mousedown, etc.) and a sound being heard is important.

This time delay is called latency and is caused by several factors (input device latency, internal buffering latency, DSP processing latency, output device latency, distance of user's ears from speakers, etc.), and is cummulative. The larger this latency is, the less satisfying the user's experience is going to be. In the extreme, it can make musical production or game-play impossible. At moderate levels it can affect timing and give the impression of sounds lagging behind or the game being non-responsive. For musical applications the timing problems affect rhythm. For gaming, the timing problems affect precision of gameplay. For interactive applications, it generally cheapens the users experience much in the same way that very low animation frame-rates do. Depending on the application, a reasonable latency can be from as low as 3-6 milliseconds to 25-50 milliseconds.

Requirements and Use Cases

Please see Example Applications

Acknowledgements

This specification is the collective work of the W3C Audio Working Group.

Members of the Working Group are (at the time of writing, and by alphabetical order):
Adenot, Paul (Mozilla Foundation) - Specification Co-editor; Akhgari, Ehsan (Mozilla Foundation); Berkovitz, Joe (Hal Leonard/Noteflight) – WG Chair; Bossart, Pierre (Intel Corporation); Carlson, Eric (Apple, Inc.); Choi, Hongchan (Google, Inc.); Geelnard, Marcus (Opera Software); Goode, Adam (Google, Inc.); Gregan, Matthew (Mozilla Foundation); Jägenstedt, Philip (Opera Software); Kalliokoski, Jussi (Invited Expert); Lilley, Chris (W3C Staff); Lowis, Chris (Invited Expert. WG co-chair from December 2012 to September 2013, affiliated with British Broadcasting Corporation); Mandyam, Giridhar (Qualcomm Innovation Center, Inc); Noble, Jer (Apple, Inc.); O'Callahan, Robert(Mozilla Foundation); Onumonu, Anthony (British Broadcasting Corporation); Paradis, Matthew (British Broadcasting Corporation); Raman, T.V. (Google, Inc.); Schepers, Doug (W3C/MIT); Shires, Glen (Google, Inc.); Smith, Michael (W3C/Keio); Thereaux, Olivier (British Broadcasting Corporation); Toy, Raymond (Google, Inc.); Verdie, Jean-Charles (MStar Semiconductor, Inc.); Wilson, Chris (Google,Inc.) - Specification Co-editor; ZERGAOUI, Mohamed (INNOVIMAX)

Former members of the Working Group and contributors to the specification include:
Caceres, Marcos (Invited Expert); Cardoso, Gabriel (INRIA); Chen, Bin (Baidu, Inc.); MacDonald, Alistair (W3C Invited Experts) — WG co-chair from March 2011 to July 2012; Michel, Thierry (W3C/ERCIM); Rogers, Chris (Google, Inc.) – Specification Editor until August 2013; Wei, James (Intel Corporation);

Web Audio API Change Log

See changelog.html.