This specification describes a high-level JavaScript API for processing and synthesizing audio in web applications. The primary paradigm is of an audio routing graph, where a number of AudioNode objects are connected together to define the overall audio rendering. The actual processing will primarily take place in the underlying implementation (typically optimized Assembly / C / C++ code), but direct JavaScript processing and synthesis is also supported.

The introductory section covers the motivation behind this specification.

This API is designed to be used in conjunction with other APIs and elements on the web platform, notably: XMLHttpRequest [[XHR]] (using the responseType and response attributes). For games and interactive applications, it is anticipated to be used with the canvas 2D [[2dcontext]] and WebGL [[WEBGL]] 3D graphics APIs.

Introduction

Audio on the web has been fairly primitive up to this point and until very recently has had to be delivered through plugins such as Flash and QuickTime. The introduction of the audio element in HTML5 is very important, allowing for basic streaming audio playback. But, it is not powerful enough to handle more complex audio applications. For sophisticated web-based games or interactive applications, another solution is required. It is a goal of this specification to include the capabilities found in modern game audio engines as well as some of the mixing, processing, and filtering tasks that are found in modern desktop audio production applications.

The APIs have been designed with a wide variety of use cases [[webaudio-usecases]] in mind. Ideally, it should be able to support any use case which could reasonably be implemented with an optimized C++ engine controlled via JavaScript and run in a browser. That said, modern desktop audio software can have very advanced capabilities, some of which would be difficult or impossible to build with this system. Apple's Logic Audio is one such application which has support for external MIDI controllers, arbitrary plugin audio effects and synthesizers, highly optimized direct-to-disk audio file reading/writing, tightly integrated time-stretching, and so on. Nevertheless, the proposed system will be quite capable of supporting a large range of reasonably complex games and interactive applications, including musical ones. And it can be a very good complement to the more advanced graphics features offered by WebGL. The API has been designed so that more advanced capabilities can be added at a later time.

Features

The API supports these primary features:

Modular Routing

Modular routing allows arbitrary connections between different AudioNode objects. Each node can have inputs and/or outputs. A source node has no inputs and a single output. A destination node has one input and no outputs. Other nodes such as filters can be placed between the source and destination nodes. The developer doesn't have to worry about low-level stream format details when two objects are connected together; the right thing just happens. For example, if a mono audio stream is connected to a stereo input it should just mix to left and right channels appropriately.

In the simplest case, a single source can be routed directly to the output. All routing occurs within an AudioContext containing a single AudioDestinationNode:

modular routing
A simple example of modular routing.

Illustrating this simple routing, here's a simple example playing a single sound:


var context = new AudioContext();

function playSound() {
    var source = context.createBufferSource();
    source.buffer = dogBarkingBuffer;
    source.connect(context.destination);
    source.start(0);
}

Here's a more complex example with three sources and a convolution reverb send with a dynamics compressor at the final output stage:

modular routing2
A more complex example of modular routing.

var context = 0;
var compressor = 0;
var reverb = 0;

var source1 = 0;
var source2 = 0;
var source3 = 0;

var lowpassFilter = 0;
var waveShaper = 0;
var panner = 0;

var dry1 = 0;
var dry2 = 0;
var dry3 = 0;

var wet1 = 0;
var wet2 = 0;
var wet3 = 0;

var masterDry = 0;
var masterWet = 0;

function setupRoutingGraph () {
    context = new AudioContext();

    // Create the effects nodes.
    lowpassFilter = context.createBiquadFilter();
    waveShaper = context.createWaveShaper();
    panner = context.createPanner();
    compressor = context.createDynamicsCompressor();
    reverb = context.createConvolver();

    // Create master wet and dry.
    masterDry = context.createGain();
    masterWet = context.createGain();

    // Connect final compressor to final destination.
    compressor.connect(context.destination);

    // Connect master dry and wet to compressor.
    masterDry.connect(compressor);
    masterWet.connect(compressor);

    // Connect reverb to master wet.
    reverb.connect(masterWet);

    // Create a few sources.
    source1 = context.createBufferSource();
    source2 = context.createBufferSource();
    source3 = context.createOscillator();

    source1.buffer = manTalkingBuffer;
    source2.buffer = footstepsBuffer;
    source3.frequency.value = 440;

    // Connect source1
    dry1 = context.createGain();
    wet1 = context.createGain();
    source1.connect(lowpassFilter);
    lowpassFilter.connect(dry1);
    lowpassFilter.connect(wet1);
    dry1.connect(masterDry);
    wet1.connect(reverb);

    // Connect source2
    dry2 = context.createGain();
    wet2 = context.createGain();
    source2.connect(waveShaper);
    waveShaper.connect(dry2);
    waveShaper.connect(wet2);
    dry2.connect(masterDry);
    wet2.connect(reverb);

    // Connect source3
    dry3 = context.createGain();
    wet3 = context.createGain();
    source3.connect(panner);
    panner.connect(dry3);
    panner.connect(wet3);
    dry3.connect(masterDry);
    wet3.connect(reverb);

    // Start the sources now.
    source1.start(0);
    source2.start(0);
    source3.start(0);
}

Modular routing also permits the output of AudioNodes to be routed to an AudioParam parameter that controls the behavior of a different AudioNode. In this scenario, the output of a node can act as a modulation signal rather than an input signal.

modular routing3
Modular routing illustrating one Oscillator modulating the frequency of another.
function setupRoutingGraph() {
  var context = new AudioContext();

  // Create the low frequency oscillator that supplies the modulation signal
  var lfo = context.createOscillator();
  lfo.frequency.value = 1.0;

  // Create the high frequency oscillator to be modulated
  var hfo = context.createOscillator();
  hfo.frequency.value = 440.0;

  // Create a gain node whose gain determines the amplitude of the modulation signal
  var modulationGain = context.createGain();
  modulationGain.gain.value = 50;

  // Configure the graph and start the oscillators
  lfo.connect(modulationGain);
  modulationGain.connect(hfo.detune);
  hfo.connect(context.destination);
  hfo.start(0);
  lfo.start(0);
}

API Overview

The interfaces defined are:

There are also several features that have been deprecated from the Web Audio API but not yet removed, pending implementation experience of their replacements:

The following conformance classes are defined by this specification:

conforming implementation

A user agent is considered to be a conforming implementation if it satisfies all of the MUST-, REQUIRED- and SHALL-level criteria in this specification that apply to implementations.

User agents that use ECMAScript to implement the APIs defined in this specification must implement them in a manner consistent with the ECMAScript Bindings defined in the Web IDL specification [[!WEBIDL]] as this specification uses that specification and terminology.

The Audio API

The BaseAudioContext Interface

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

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

suspended
This context is currently suspended (context time is not proceeding, audio hardware may be powered down/released).
running
Audio is being processed.
closed
This context has been released, and can no longer be used to process audio. All system audio resources have been released. Attempts to create new Nodes on the AudioContext will throw InvalidStateError. (AudioBuffers may still be created, through createBuffer or decodeAudioData.)
balanced
Balance audio output latency and power consumption.
interactive
Provide the lowest audio output latency possible without glitching. This is the default.
playback
Prioritize sustained playback without interruption over audio output latency. Lowest power consumption.
readonly attribute AudioDestinationNode destination

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

readonly attribute float sampleRate

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

readonly attribute double currentTime

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

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

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

When the BaseAudioContext is in the running state, the value of this attribute is monotonically increasing and is updated by the rendering thread in uniform increments, corresponding to the audio block size of 128 samples. Thus, for a running context, currentTime increases steadily as the system processes audio blocks, and always represents the time of the start of the next audio block to be processed. It is also the earliest possible time when any change scheduled in the current state might take effect.

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

readonly attribute AudioListener listener

An AudioListener which is used for 3D spatialization.

readonly attribute AudioContextState state

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

readonly attribute double baseLatency

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

For example, if the audio context is running at 44.1 kHz and the AudioDestinationNode implements double buffering internally and can process and output audio every 128 sample frames, then the processing latency is \((2\cdot128)/44100 = 5.805 \mathrm{ ms}\), approximately.

Promise<void> suspend()

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

When suspend is called, execute these steps:

  1. Let promise be a new Promise.
  2. If the control thread state flag on the AudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
  3. If the state attribute of the AudioContext is already suspended, resolve promise, return it, and abort these steps.
  4. Set the control thread state flag on the AudioContext to suspended.
  5. Queue a control message to suspend the AudioContext.
  6. Return promise.

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

  1. Attempt to release system resources.
  2. Set the rendering thread state on the AudioContext to suspended.
  3. Queue a task on the control thread's event loop, to execute these steps:
    1. Resolve promise.
    2. If the state attribute of the AudioContext is not already suspended:
      1. Set the state attribute of the AudioContext to suspended.
      2. Queue a task to fire a simple event named statechange at the AudioContext

While a BaseAudioContext 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. AudioWorkletProcessors and ScriptProcessorNodes will cease to have their processing handlers invoked 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. In particular, calling AnalyserNode functions repeatedly when a BaseAudioContext is suspended MUST return the same data.

Promise<void> resume()

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

When resume is called, execute these steps:

  1. Let promise be a new Promise.
  2. If the control thread state flag on the AudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
  3. If the state attribute of the AudioContext is already running, resolve promise, return it, and abort these steps.
  4. If the AudioContext is not allowed to start, append promise to pendingResumePromises and abort these steps, returning promise.
  5. Set the control thread state flag on the AudioContext to running.
  6. Queue a control message to resume the AudioContext.
  7. Return promise.

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

  1. Attempt to acquire system resources.
  2. Set the rendering thread state flag on the AudioContext to running.
  3. Start rendering the audio graph.
  4. In case of failure, queue a task on the control thread to execute the following, and abort these steps
    1. Reject all promises from pendingResumePromises in order, then clear pendingResumePromises.
    2. Reject promise
  5. Queue a task on the control thread's event loop, to execute these steps:
    1. Resolve all promises from pendingResumePromises in order, then clear pendingResumePromises.
    2. Resolve promise.
    3. If the state attribute of the AudioContext is not already running:
      1. Set the state attribute of the AudioContext to running.
      2. Queue a task to fire a simple event named statechange at the AudioContext
attribute EventHandler onstatechange

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

AudioBuffer createBuffer()

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

unsigned long numberOfChannels
Determines how many channels the buffer will have. An implementation must support at least 32 channels.
unsigned long length
Determines the size of the buffer in sample-frames.
float sampleRate
Describes the sample-rate of the linear PCM audio data in the buffer in sample-frames per second. An implementation must support sample rates in at least the range 8000 to 96000.
Promise<AudioBuffer> decodeAudioData()

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

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

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

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

  1. Let promise be a new promise.
  2. If the operation IsDetachedBuffer (described in [[!ECMASCRIPT]]) on audioData is false, execute the following steps:
    1. Detach the audioData ArrayBuffer. This operation is described in [[!ECMASCRIPT]].
    2. Queue a decoding operation to be performed on another thread.
  3. Else, execute the following steps:
    1. Let error be a DOMException whose name is "TypeError".
    2. Reject promise with error.
    3. Queue a task to invoke errorCallback with error.
  4. Return promise.

When queuing a decoding operation to be performed on another thread, the following steps MUST happen on a thread that is not the control thread nor the rendering thread, called the decoding thread.

Multiple decoding threads can run in parallel to service multiple calls to decodeAudioData.
  1. Attempt to decode the encoded audioData into linear PCM.
  2. If a decoding error is encountered due to the audio format not being recognized or supported, or because of corrupted/unexpected/inconsistent data, then queue a task to execute the following step, on the control thread's event loop:
    1. Let error be a DOMException whose name is "EncodingError".
    2. Reject promise with error.
    3. If errorCallback is not missing, invoke errorCallback with error.
  3. Otherwise:
    1. Take the result, representing the decoded linear PCM audio data, and resample it to the sample-rate of the AudioContext if it is different from the sample-rate of audioData.
    2. Queue a task on the control thread's event loop to execute the following steps:
      1. Let buffer be an AudioBuffer containing the final result (after possibly sample-rate conversion).
      2. Resolve promise with buffer.
      3. If successCallback is not missing, invoke successCallback with buffer.
AudioBufferSourceNode createBufferSource()

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

ConstantSourceNode createConstantSource()

Creates a ConstantSourceNode.

ScriptProcessorNode createScriptProcessor()

This method is DEPRECATED, as it is intended to be replaced by AudioWorkletNode. 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. If the value of this parameter is not one of the allowed power-of-2 values listed above, an IndexSizeError MUST be thrown.
optional unsigned long numberOfInputChannels = 2
This parameter determines the number of channels for this node's input. Values of up to 32 must be supported.
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. In this case an IndexSizeError MUST be thrown.

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.

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

Creates an IIRFilterNode representing a general IIR Filter.

sequence<double> feedforward
An array of the feedforward (numerator) coefficients for the transfer function of the IIR filter. The maximum length of this array is 20. If all of the values are zero, an InvalidStateError MUST be thrown. A NotSupportedError MUST be thrown if the array length is 0 or greater than 20.
sequence<double> feedback
An array of the feedback (denominator) coefficients for the tranfer function of the IIR filter. The maximum length of this array is 20. If the first element of the array is 0, an InvalidStateError MUST be thrown. A NotSupportedError MUST be thrown if the array length is 0 or greater than 20.
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 or an IndexSizeError exception MUST be thrown. All implementations must support arrays up to at least 8192. These parameters specify the Fourier coefficients of a Fourier series representing the partials of a periodic waveform. The created PeriodicWave will be used with an OscillatorNode and, by default, will represent a normalized time-domain waveform having maximum absolute peak value of 1. Another way of saying this is that the generated waveform of an OscillatorNode will have maximum peak value at 0dBFS. Conveniently, this corresponds to the full-range of the signal values used by the Web Audio API. Because the PeriodicWave is normalized by default on creation, the real and imag parameters represent relative values. If normalization is disabled via the disableNormalization parameter, this normalization is disabled, and the time-domain waveform has the amplitudes as given by the Fourier coefficients.

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

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.
optional PeriodicWaveConstraints constraints
If not given, the waveform is normalized. Otherwise, the waveform is normalized according the value given by constraints.
AudioBuffer decodedData
The AudioBuffer containing the decoded audio data.
DOMException error
The error that occurred while decoding.

Lifetime

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

Lack of introspection or serialization primitives

The Web Audio API takes a fire-and-forget approach to audio source scheduling. That is, source nodes are created for each note during the lifetime of the AudioContext, and never explicitly removed from the graph. This is incompatible with a serialization API, since there is no stable set of nodes that could be serialized.

Moreover, having an introspection API would allow content script to be able to observe garbage collections.

System resources associated with an AudioContext

AudioContexts should be considered expensive objects. Creating an AudioContext often involves creating a high-priority thread, and using a low-latency system audio stream, both having an impact on energy consumption. Creating more than one AudioContext in a document is most of the time unnecessary.

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

suspend and close allow authors to release system resources. Releasing system resources means releasing the system resources such as threads, processes, audio streams, but conserving the state of the AudioContext such that it can continue to operate later after resuming if needed.

Constructing or resuming an AudioContext involves acquiring system resources. This means opening a system audio stream. This operation returns when the audio stream is ready.

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

The AudioContext Interface

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

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

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

Constructor

When creating an AudioContext, execute these steps:

  1. Set a control thread state to suspended on the AudioContext.
  2. Set a rendering thread state to suspended on the AudioContext.
  3. Let pendingResumePromises be an empty ordered list of promises.
  4. If the AudioContext is not allowed to start, abort these steps.
  5. Send a control message to start processing.

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

  1. Attempt to acquire system resources.
  2. In case of failure, abort these steps.
  3. Set the rendering thread state to running on the AudioContext.
  4. Queue a task on the control thread event loop, to execute these steps:
    1. Set the state attribute of the AudioContext to running.
    2. Queue a task to fire a simple event named statechange at the AudioContext.

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

readonly attribute double outputLatency

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

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

AudioTimestamp getOutputTimestamp()

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

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

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

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

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

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

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

Promise<void> close()

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

When close is called, execute these steps:

  1. Let promise be a new Promise.
  2. If the control thread state flag on the AudioContext is closed reject the promise with InvalidStateError, abort these steps, returning promise.
  3. If the state attribute of the AudioContext is already closed, resolve promise, return it, and abort these steps.
  4. Set the control thread state flag on the AudioContext to closed.
  5. Queue a control message to the AudioContext.
  6. Return promise.

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

  1. Attempt to release system resources.
  2. Set the rendering thread state to suspended.
  3. Queue a task on the control thread's event loop, to execute these steps:
    1. Resolve promise.
    2. If the state attribute of the AudioContext is not already closed:
      1. Set the state attribute of the AudioContext to closed.
      2. Queue a task to fire a simple event named statechange at the AudioContext

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

MediaElementAudioSourceNode createMediaElementSource()

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

HTMLMediaElement mediaElement
The media element that will be re-routed.
MediaStreamAudioSourceNode createMediaStreamSource()

Creates a MediaStreamAudioSourceNode.

MediaStream mediaStream
The media stream that will act as source.
MediaStreamAudioDestinationNode createMediaStreamDestination()

Creates a MediaStreamAudioDestinationNode

AudioContextOptions

The AudioContextOptions dictionary is used to specify a requested latency for an AudioContext.

(AudioContextLatencyCategory or double) latencyHint = "interactive"

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

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

AudioTimestamp

double contextTime
Represents a point in the time coordinate system of BaseAudioContext's currentTime.
DOMHighResTimeStamp performanceTime
Represents a point in the time coordinate system of a Performance interface implementation (described in [[!hr-time-2]]).

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.

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

unsigned long numberOfChannels
Determines how many channels the buffer will have. See createBuffer for the supported number of channels.
unsigned long length
Determines the size of the buffer in sample-frames.
float sampleRate
Describes the sample-rate of the linear PCM audio data in the buffer in sample-frames per second. See createBuffer for valid sample rates.
Promise<AudioBuffer> startRendering()

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

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

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

  1. Set a flag called renderingStarted on the OfflineAudioContext to true.
  2. If the renderingStarted flag on the OfflineAudioContext is true, return a rejected promise with InvalidStateError, and abort these steps.
  3. Let promise be a new promise.
  4. Start to render the audio graph on another thread.
  5. Return promise

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

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

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

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

Promise<void> suspend()

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

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

double suspendTime
Schedules a suspension of the rendering at the specified time, which is quantized and rounded down to the render quantum size. If the quantized frame number
  1. is negative or
  2. is less than or equal to the current time or
  3. is greater than or equal to the total render duration or
  4. is scheduled by another suspend for the same time,
then the promise is rejected with InvalidStateError.
readonly attribute unsigned long length

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

attribute EventHandler oncomplete

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

The OfflineAudioCompletionEvent Interface

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

readonly attribute AudioBuffer renderedBuffer

An AudioBuffer containing the rendered audio data.

The AudioNode Interface

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

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

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

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

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

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

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

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.
AudioNode connect()
AudioNode destination
The destination parameter is the AudioNode to connect to. If the destination parameter is an AudioNode that has been created using another AudioContext, an InvalidAccessError MUST be thrown. That is, AudioNodes cannot be shared between AudioContexts.
optional unsigned long output = 0
The output parameter is an index describing which output of the AudioNode from which to connect. If this parameter is out-of-bound, an IndexSizeError exception MUST be thrown. It is possible to connect an AudioNode output to more than one input with multiple calls to connect(). Thus, "fan-out" is supported.
optional unsigned long input = 0
The input parameter is an index describing which input of the destination AudioNode to connect to. If this parameter is out-of-bounds, an IndexSizeError exception MUST be thrown. It is possible to connect an AudioNode to another AudioNode which creates a cycle: an AudioNode may connect to another AudioNode, which in turn connects back to the first AudioNode. This is allowed only if there is at least one DelayNode in the cycle or a NotSupportedError exception MUST be thrown.

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

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

will have the same effect as

      nodeA.connect(nodeB);
    

This method returns destination AudioNode object.

void connect()

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

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

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

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

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

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

      nodeA.connect(param);
      nodeA.connect(param);
    
will have the same effect as
      nodeA.connect(param);
    
void disconnect()

Disconnects all outgoing connections from the AudioNode.

void disconnect()

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

unsigned long output
This parameter is an index describing which output of the AudioNode to disconnect. It disconnects all outgoing connections from the given output. If this parameter is out-of-bounds, an IndexSizeError exception MUST be thrown.
void disconnect()

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

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

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

AudioNode destination
The destination parameter is the AudioNode to disconnect. If there is no connection to the destination from the given output, an InvalidAccessError exception MUST be thrown.
unsigned long output
The output parameter is an index describing which output of the AudioNode from which to disconnect. If this parameter is out-of-bound, an IndexSizeError exception MUST be thrown.
void disconnect()

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

AudioNode destination
The destination parameter is the AudioNode to disconnect. If there is no connection to the destination from the given output to the given input, an InvalidAccessError exception MUST be thrown.
unsigned long output
The output parameter is an index describing which output of the AudioNode from which to disconnect. If this parameter is out-of-bound, an IndexSizeError exception MUST be thrown.
unsigned long input
The input parameter is an index describing which input of the destination AudioNode to disconnect. If this parameter is out-of-bounds, an IndexSizeError exception MUST be thrown.
void disconnect()

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

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

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

AudioParam destination
The destination parameter is the AudioParam to disconnect. If there is no connection to the destination, an InvalidAccessError exception MUST be thrown.
unsigned long output
The output parameter is an index describing which output of the AudioNode from which to disconnect. If the parameter is out-of-bound, an IndexSizeError exception MUST be thrown.
readonly attribute 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 AudioWorkletNode have variable number of inputs.

readonly attribute unsigned long numberOfOutputs

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

attribute unsigned long channelCount

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

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

AudioDestinationNode

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

AudioContext
The channel count must be between 1 and maxChannelCount. An IndexSizeError exception must be thrown for any attempt to set the count outside this range.
OfflineAudioContext
The channel count cannot be changed. An InvalidStateError exception MUST be thrown for any attempt to change the value.
ChannelMergerNode
The channel count cannot be changed, and an InvalidStateError exception MUST be thrown for any attempt to change the value.
ConvolverNode
The channel count cannot be greater than two, and a NotSupportedError exception MUST be thrown if count is set to a value greater than two.
PannerNode
The channel count cannot be greater than two, and a NotSupportedError exception MUST be thrown for any attempt to change the to a value greater than two.
ScriptProcessorNode
The channel count cannot be changed, and an InvalidStateError exception MUST be thrown for any attempt to change the value.
StereoPannerNode
The channel count cannot be greater than two, and a NotSupportedError exception MUST be thrown for any attempt to change the to a value greater than two.

See the section for more information on this attribute.

attribute ChannelCountMode channelCountMode

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

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

AudioDestinationNode
If the AudioDestinationNode is the destination node of an OfflineAudioContext, then the channel count mode cannot be changed. An InvalidStateError exception MUST be thrown for any attempt to change the value.
ChannelMergerNode
The channel count mode cannot be changed from "explicit" and an InvalidStateError exception must be thrown for any attempt to change the value.
ConvolverNode
The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".
PannerNode
The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".
ScriptProcessorNode
The channel count mode cannot be changed from "explicit" and an InvalidStateError exception MUST be thrown for any attempt to change the value.
StereoPannerNode
The channel count mode cannot be set to "max", and a NotSupportedError exception MUST be thrown for any attempt to set it to "max".

See the section for more information on this attribute.

attribute ChannelInterpretation channelInterpretation

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

See the section for more information on this attribute.

Dictionaries

unsigned long channelCount
Desired number of channels for the channelCount attribute.
ChannelCountMode channelCountMode
Desired mode for the channelCountMode attribute.
ChannelInterpretation channelInterpretation
Desired mode for the channelCountMode attribute.

Lifetime

This section is informative.

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

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

  1. A normal JavaScript reference obeying normal garbage collection rules.
  2. A playing reference for both AudioBufferSourceNodes and OscillatorNodes. These nodes maintain a playing reference to themselves while they are currently playing.
  3. A connection reference which occurs if another AudioNode is connected to it.
  4. A tail-time reference which an AudioNode maintains on itself as long as it has any internal processing state which has not yet been emitted. For example, a ConvolverNode has a tail which continues to play even after receiving silent input (think about clapping your hands in a large concert hall and continuing to hear the sound reverberate throughout the hall). Some AudioNodes have this property. Please see details for specific nodes.
  5. MediaStreams keep a MediaStreamAudioSourceNode alive as long as the underlying MediaStreamTrack that is playing through the MediaStreamAudioSourceNode have not ended (as per [[!mediacapture-streams]]).
  6. HTMLMediaElements keep their associated MediaElementAudioSourceNode alive as long as the HTMLMediaElement is in a state where audio could ever be played in the future.

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

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

The uninterrupted operation of AudioNodes implies that as long as live references exist to a node, the node will continue processing its inputs and evolving its internal state even if it is disconnected from the audio graph. Since this processing will consume CPU and power, developers should carefully consider the resource usage of disconnected nodes. In particular, it is a good idea to minimize resource consumption by explicitly putting disconnected nodes into a stopped state when possible.

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

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

The AudioDestinationNode Interface

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

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

      numberOfInputs  : 1
      numberOfOutputs : 1

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

For an AudioContext, the defaults are

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

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

For an OfflineAudioContext, the defaults are

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

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

readonly attribute unsigned long maxChannelCount

The maximum number of channels that the channelCount attribute can be set to. An AudioDestinationNode representing the audio hardware end-point (the normal case) can potentially output more than 2 channels of audio if the audio hardware is multi-channel. maxChannelCount is the maximum number of channels that this hardware is capable of supporting. 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.

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.

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

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

The following rules will apply when calling these methods:

attribute float value

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

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

readonly attribute float defaultValue

Initial value for the value attribute.

readonly attribute float minValue

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

readonly attribute float maxValue

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

AudioParam setValueAtTime(float value, double startTime)

Schedules a parameter value change at the given time.

float value
The value the parameter will change to at the given time.
double startTime
The time in the same time coordinate system as the BaseAudioContext's currentTime attribute at which the parameter changes to the given value. A TypeError 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 \(t \geq T_0\), \(v(t) = V\), where \(T_0\) is the startTime parameter and \(V\) is the value parameter. In other words, the value will remain constant.

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

              $$
                v(t) = V
              $$
            

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

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

AudioParam linearRampToValueAtTime(float value, double endTime)

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

float value
The value the parameter will linearly ramp to at the given time.
double endTime
The time in the same time coordinate system as the AudioContext's currentTime attribute at which the automation ends. A TypeError exception MUST be thrown if endTime is negative or is not a finite number.

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

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

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

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

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

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

AudioParam exponentialRampToValueAtTime(float value, double endTime)

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

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

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

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

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

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

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

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

float value
The value the parameter will exponentially ramp to at the given time. A NotSupportedError exception MUST be thrown if this value is equal to 0.
double endTime
The time in the same time coordinate system as the AudioContext's currentTime attribute where the exponential ramp ends. A TypeError exception MUST be thrown if endTime is negative or is not a finite number.
AudioParam setTargetAtTime(float target, double startTime, float timeConstant)

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

float target
The value the parameter will start changing to at the given time.
double startTime
The time at which the exponential approach will begin, in the same time coordinate system as the AudioContext's currentTime attribute. A TypeError exception MUST be thrown if start is negative or is not a finite number. If startTime is less than currentTime, it is clamped to currentTime.
float timeConstant
The time-constant value of first-order filter (exponential) approach to the target value. The larger this value is, the slower the transition will be. The value must be non-negative or a TypeError exception MUST be thrown. If timeConstant is zero, the output value jumps immediately to the final value.

More precisely, timeConstant is the time it takes a first-order linear continuous time-invariant system to reach the value \(1 - 1/e\) (around 63.2%) given a step input response (transition from 0 to 1 value).

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

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

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

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

AudioParam setValueCurveAtTime(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.

Float32Array values

A Float32Array representing a parameter value curve. These values will apply starting at the given time and lasting for the given duration. When this method is called, an internal copy of the curve is created for automation purposes. Subsequent modifications of the contents of the passed-in array therefore have no effect on the AudioParam.

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

double startTime
The start time in the same time coordinate system as the AudioContext's currentTime attribute at which the value curve will be applied. A TypeError exception MUST be thrown if startTime is negative or is not a finite number.. If startTime is less than currentTime, it is clamped to currentTime.
double duration
The amount of time in seconds (after the time parameter) where values will be calculated according to the values parameter. A TypeError exception MUST be thrown if duration is not strictly positive or is not a finite number.

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

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

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

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

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

AudioParam cancelScheduledValues(double startTime)

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

double startTime
The starting time at and after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the AudioContext's currentTime attribute. A TypeError exception MUST be thrown if startTime is negative or is not a finite number. If startTime is less than currentTime, it is clamped to currentTime.

Computation of Value

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

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

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

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

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

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

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

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

AudioParam automation clipping to nominal
An example of clipping of an AudioParam automation from the nominal range.

AudioParam Automation Example

AudioParam automation
An example of parameter automation.
var curveLength = 44100;
var curve = new Float32Array(curveLength);
for (var i = 0; i < curveLength; ++i)
    curve[i] = Math.sin(Math.PI * i / curveLength);

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

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

The GainNode Interface

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

  numberOfInputs  : 1
  numberOfOutputs : 1

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

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

This node has no tail-time reference.

readonly attribute AudioParam gain

Represents the amount of gain to apply. Its default value is 1 (no gain change). This parameter is a-rate. Its nominal range is (-\(\infty\), +\(\infty\)).

GainOptions

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

float gain
The initial gain value for the gain AudioParam.

The DelayNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

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

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

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

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

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

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).

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

This parameter is a-rate.

DelayOptions

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

double maxDelayTime = 1.0
The maximum delay time for the node.
double delayTime
The initial delay time for the node

The AudioBuffer Interface

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

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

readonly attribute float sampleRate

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

readonly attribute unsigned 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 unsigned 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.
unsigned long channelNumber
The index of the channel to copy the data from. If channelNumber is greater or equal than the number of channel of the AudioBuffer, an IndexSizeError MUST be thrown.
optional unsigned long startInChannel = 0
An optional offset to copy the data from. If startInChannel is greater than the length of the AudioBuffer, an IndexSizeError MUST be thrown.
void copyToChannel()

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

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

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

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

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

  1. If the operation IsDetachedBuffer on any of the AudioBuffer's ArrayBuffer return true, abort these steps, and return a zero-length channel data buffers to the invoker.
  2. Detach 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.
The acquire the contents of an AudioBuffer operation is invoked in the following cases:

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

AudioBufferOptions

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

unsigned long numberOfChannels = 1
The number of channels for the buffer.
required unsigned long length
The length in sample frames of the buffer.
float sampleRate
The sample rate in Hz for the buffer. The default is the sample rate of the context used in constructing this object.

The AudioBufferSourceNode Interface

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

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

  numberOfInputs  : 0
  numberOfOutputs : 1

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

This node has no tail-time reference.

attribute AudioBuffer? buffer

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

  1. Let new buffer be the AudioBuffer to be assigned to buffer.
  2. If new buffer is not null and buffer set is true, throw an InvalidStateError and abort these steps.
  3. If new buffer is not null, set buffer set to true.
  4. Assign new buffer to the buffer attribute.
readonly attribute AudioParam playbackRate

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

readonly attribute AudioParam detune

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

attribute boolean loop

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

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. If loopStart is less than 0, looping will begin at 0. If loopStart is greater than the duration of the buffer, looping will begin at the end of the buffer. This attribute is converted to an exact sample frame offset within the buffer by multiplying by the buffer's sample rate and rounding to the nearest integer value. Thus its behavior is independent of the value of the playbackRate parameter.

attribute double loopEnd

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

void start()

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

optional double when = 0
The when parameter describes at what time (in seconds) the sound should start playing. It is in the same time coordinate system as the AudioContext's currentTime attribute. If 0 is passed in for this value or if the value is less than currentTime, then the sound will start playing immediately. A TypeError exception MUST be thrown if when is negative.
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. A TypeError exception MUST be thrown if offset is negative. If offset is greater than loopEnd, playback will begin at loopEnd (and immediately loop to loopStart). This parameter is converted to an exact sample frame offset within the buffer by multiplying by the buffer's sample rate and rounding to the nearest integer value. Thus its behavior is independent of the value of the playbackRate parameter.
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 TypeError exception MUST be thrown if duration is negative.
void stop()

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

optional double when = 0
The when parameter describes at what time (in seconds) the source should stop playing. It is in the same time coordinate system as the AudioContext's currentTime attribute. If 0 is passed in for this value or if the value is less than currentTime, then the sound will stop playing immediately. A TypeError exception MUST be thrown if when is negative.
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 that form a compound parameter and are used together to determine a computedPlaybackRate value:

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

The computedPlaybackRate is the effective speed at which the AudioBuffer of this AudioBufferSourceNode MUST be played. Its nominal range is [-100, 100].

This MUST be implemented by resampling the input data using a resampling ratio of 1 / computedPlaybackRate, hence changing both the pitch and speed of the audio.

AudioBufferSourceOptions

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

AudioBuffer? buffer
The audio asset to be played. This is equivalent to assigning buffer to the buffer attribute of the AudioBufferSourceNode.
float detune
The initial value for the detune AudioParam.
boolean loop
The initial value for the loop attribute.
double loopEnd
The initial value for the loopEnd attribute.
double loopStart
The initial value for the loopStart attribute.
float playbackRate
The initial value for the playbackRate AudioParam.

Looping

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

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

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

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

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

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

The ConstantSourceNode Interface

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

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

  numberOfInputs  : 0
  numberOfOutputs : 1
        
readonly attribute AudioParam offset

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

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 constant value is finished, an event of type Event (described in HTML [[!HTML]]) will be dispatched to the event handler.

void start(optional double when = 0)

Schedules the constant source to playback at an exact time. This is defined exactly the same as for the start() method for an AudioBufferSourceNode, except that the the optional offset and duration parameters are not allowed.

optional double when = 0
Defined the same as the when parameter of the start() method for an AudioBufferSourceNode.
void stop(optional double when = 0)

Schedules the source to stop playback at an exact time. This is defined exactly the same as for the stop() method for an AudioBufferSourceNode

optional double when = 0
Defined the same as the when parameter of the stop() method for an AudioBufferSourceNode

ConstantSourceOptions

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

float offset = 1
The initial value for the offset AudioParam of this node.

The MediaElementAudioSourceNode Interface

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

  numberOfInputs  : 0
  numberOfOutputs : 1

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

This node has no tail-time reference.

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);

MediaElementAudioSourceOptions

This specifies the options to use in constructing a MediaElementAudioSourceNode.

required HTMLMediaElement mediaElement
The media element that will be re-routed. This MUST be specified.

Security with MediaElementAudioSourceNode and cross-origin resources

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

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

The AudioWorklet interface

Concepts

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

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

AudioWorklet concept
AudioWorkletNode and AudioWorkletProcessor

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

node name to processor definition map
Belongs to AudioWorkletGlobalScope. This map associates a string key to the corresponding AudioWorkletProcessor definition. Initially this map is empty and becomes populated when registerProcessor method is called.
node instance set
Belongs to an instance of BaseAudioContext. This set stores the reference to all the active instances of AudioWorkletNode that are created by the associated BaseAudioContext. The context keeps the track of instances until they are destroyed by the user agent.
// bypass.js script file, AudioWorkletGlobalScope
registerProcessor("Bypass", class extends AudioWorkletProcessor {
  process (inputs, outputs) {
    // Single input, single channel.
    var input = inputs[0], output = outputs[0];
    output[0].set(input[0]);
  }
});
          
// The main global scope
window.audioWorklet.import("bypass.js").then(function () {
  var context = new AudioContext();
  var bypass = new AudioWorkletNode(context, "Bypass");
});
          

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

[SameObject] readonly attribute Worklet audioWorklet

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

The AudioWorkletGlobalScope interface

This special execution context is designed to enable the generation, processing, and analysis of audio data directly using JavaScript in the audio rendering thread. The user-supplied script code is evaluated in this scope to define an AudioWorkletProcessor.

AudioWorkletGlobalScope has a node name to processor definition map. This map stores definitions of AudioWorkletProcessor with the associated string key. Initially it is empty and populated when registerProcessor method is called, but this storage is internal and is not directly exposed to the user.

void registerProcessor ()

Registers a class definition derived from AudioWorkletProcessor.

When the registerProcessor(name, processorConstructor) method is called, the user agent must run the following steps:

  1. If the name exists as a key in the node name to processor definition map, throw a NotSupportedError exception and abort these steps because registering a definition with a duplicated key is not allowed.
  2. If the result of IsConstructor(argument=processorConstructor) is false, throw a TypeError and abort these steps.
  3. Let prototype be the result of Get(O=processorConstructor, P="prototype").
  4. If the result of Type(argument=prototype) is not Object, throw a TypeError and abort all these steps.
  5. If the result of IsCallable(argument=Get(O=prototype, P="process")) is false, throw a TypeError and abort these steps.
  6. If the result of Get(O=processorConstructor, P="parameterDescriptors") is not an array or undefined, throw a TypeError and abort these steps.
  7. Let definition be a new AudioWorkletProcessor definition with:
    • node name being name
    • processor class constructor being processorConstructor
  8. Add the key-value pair (name - definition) to the node name to processor definition map of the associated AudioWorkletGlobalScope.

The class constructor should only be looked up once, thus it does not have the opportunity to dynamically change its definition.

DOMString name
A string key that represents a class definition to be registered. This key is used to look up the constructor of AudioWorkletProcessor during construction of an AudioWorkletNode.
VoidFunction processorCtor
A class definition extended from AudioWorkletProcessor.

The AudioWorkletNode interface

This interface represents a user-defined AudioNode which lives on the control thread. The user can create an AudioWorkletNode from an BaseAudioContext, and such a node can be connected with other built-in AudioNodes to form an audio graph. Every AudioWorkletNode has an associated processor reference, initially null.

Constructor(BaseAudioContext context, DOMString name, optional AudioWorkletNodeOptions options)

Performs the construction procedure of an AudioWorkletNode and the corresponding AudioWorkletProcessor object.

attribute EventHandler onmessage

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

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

postMessage may be called to send a message to the corresponding AudioWorkletProcessor.

AudioWorkletNodeOptions

The AudioWorkletNodeOptions dictionary can be used for the custom initialization of AudioNode attributes in the AudioWorkletNode constructor. Entries in this dictionary whose names correspond to AudioParams in the class definition of an AudioWorkletProcessor are used to initialize the parameter values upon the creation of a node.

unsigned long numberOfInputs
Same to numberOfInputs in AudioNode.
unsigned long numberOfOutputs
Same to numberOfOutputs in AudioNode.
unsigned long channelCount
Same to channelCount in AudioNode.
ChannelCountMode channelCountMode
Value used to initialize the channelCountMode attribute of an AudioWorkletNode
ChannelInterpretation channelInterpretation
Same to channelInterpretation in AudioNode.

The AudioWorkletProcessor interface

This interface represents an audio processing code that runs on the audio rendering thread. It lives in an AudioWorkletGlobalScope and the definition of the class manifests the actual audio processing mechanism of a custom audio node. AudioWorkletProcessor can only be instantiated by the construction of an AudioWorkletNode instance. Every AudioWorkletProcessor has an associated node reference, initially null.

readonly attribute AudioContextInfo contextInfo

This getter returns an AudioContextInfo object that describes various states of the associated BaseAudioContext.

attribute EventHandler onmessage

The onmessage handler is called whenever the AudioWorkletNode posts a message back to the corresponding processor object.

void postMessage()

It may be called to send a message to the associated AudioWorkletNode.

any message

Defining A Valid AudioWorkletProcessor

User can define a custom audio processor by extending AudioWorkletProcessor. The subclass must define a method named process() that implements the audio processing algorithm and have a valid static property named parameterDescriptors which is an iterable of AudioParamDescriptor that is looked up by the AudioWorkletProcessor constructor to create instances of AudioParam in the node. The step 5 and 6 of registerProcessor() ensure the validity of a given AudioWorkletProcessor subclass.

An example of subclass is as follows:

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

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

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

process() method is called synchronously by the audio rendering thread at every render quantum. It is invoked with the following arguments:

  1. inputs of type sequence<sequence<Float32Array>>
    The input audio buffer from the incoming connections provided by the user agent. inputs[n][m] is a Float32Array of audio samples for the mth channel of nth input. While the number of inputs is fixed at the construction, the number of channels can be changed dynamically.

    If no connections exist to the nth input of the node during the current render quantum, then the content of inputs[n] is an empty array, indicating that zero channels of input are available. This is the only circumstance under which the number of elements of inputs[n] can be zero.

    By checking this condition, an AudioWorkletProcessor can determine its associated node's lifetime in a way that depends on its connection status. For example, many built-in node types exhibit a tail-time, and this attribute permits AudioWorkletNodes to provide the same thing.
  2. outputs of type sequence<sequence<Float32Array>>
    The output audio buffer that is to be consumed by the user agent. outputs[n][m] is a Float32Array object containing the audio samples for mth channel of nth output. While the number of outputs is fixed at the construction, the number of channels can be changed dynamically.
  3. parameters of type Object
    A map of string keys and associated Float32Arrays. parameter["name"] corresponds to the automation values of the AudioParam named "name".

AudioParamDescriptor

The AudioParamDescriptor dictionary is used to specify properties for an AudioParam object that is used in an AudioWorkletNode.

DOMString name
Represents the name of a parameter. An NotSupportedError exception MUST be thrown when a duplicated name is found when registering the class definition.
float defaultValue = 0
Represents the default value of the parameter. If this value is out of the range of float data type or the range defined by minValue and maxValue, an NotSupportedError exception MUST be thrown.
float minValue
Represents the minimum value. An NotSupportedError exception MUST be thrown if this value is out of range of float data type or it is greater than maxValue.
float maxValue
Represents the maximum value. An NotSupportedError exception MUST be thrown if this value is out of range of float data type or it is smaller than minValue.

AudioContextInfo

The AudioContextInfo dictionary provides an AudioWorkletGlobalScope with a view of an AudioContext.

double playbackTime
The context time of the block of audio being processed. By definition this will be equal to the value of BaseAudioContext's currentTime attribute that was most recently observable in the control thread.
float sampleRate
Represents the sample rate of the associated BaseAudioContext.

The instantiation of AudioWorkletNode and AudioWorkletProcessor

When the constructor of AudioWorkletNode is invoked in the main global scope, the corresponding AudioWorkletProcessor instance is automatically created in AudioWorkletGlobalScope. After the construction, they maintain the internal reference to each other until the AudioWorkletNode instance is destroyed.

When AudioWorkletNode(context, name, options) constructor is called, the user agent must run the following steps atomically.

  1. Let this be the instance being created by constructor of AudioWorkletNode or its subclass.
  2. If name does not exists as a key in the AudioWorkletGlobalScope’s node name to processor definition map, throw a NotSupportedError exception and abort these steps.
  3. Let processorConstructor be the result of looking up name on the AudioWorkletGlobalScope’s node name to processor definition map.
  4. Let processorInstance be the result of Construct(processorConstructor, « options »). The argument list options is serialized using the structured clone algorithm.
  5. Let parameterDescriptors be the result of Get(O=processorConstructor, P="parameterDescriptors"). If parameterDescriptors is undefined, skip to the step 8.
  6. For each descriptor in parameterDescriptors perform the following substeps:
    1. Let audioParam be a new AudioParam instance with:
      • paramName being the result Get(O=descriptor, P="name").
      • defaultValue being the result Get(O=descriptor, P="defaultValue").
      • minValue being the result Get(O=descriptor, P="minValue").
      • maxValue being the result Get(O=descriptor, P="maxValue").
    2. If options contains a dictionary member with name paramName whose value is of type Number, perform Set(audioParam, "value", Get(O=options, P=paramName), false).
    3. Perform CreateDataProperty(O=this, P=paramName, V=audioParam).
  7. Set node reference in processorInstance to this.
  8. Set processor reference in this to processorInstance.
  9. Add nodeInstance to the node instance set in the BaseAudioContext.
  10. Return this as a result of the construction.

AudioWorklet Examples

The BitCrusher Node

Bitcrushing is a mechanism by which the quality of an audio stream is reduced both by quantizing the sample value (simulating a lower bit-depth), and by quantizing in time resolution (simulating a lower sample rate). This example shows how to use AudioParams (in this case, treated as a-rate) inside an AudioWorkletProcessor.

Global Scope

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

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

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

AudioWorkletGlobalScope: bitcrusher.js

registerAudioWorkletProcessor('BitCrusher', class extends AudioWorkletProcessor {

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

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

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

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

});
            

VU Meter Node

This example of a simple sound level meter further illustrates how to create an AudioWorkletNode subclass that acts like a native AudioNode, accepting constructor options and encapsulating the inter-thread communication (asynchronous) between AudioWorkletNode and AudioWorkletProcessor in clean method calls and attribute accesses. This node does not use any output.

Global Scope: vumeternode.js

class VUMeterNode extends AudioWorkletNode {

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

    options.updatingInterval = options.hasOwnProperty('updatingInterval') 
      ? options.updatingInterval 
      : 100;

    super(context, 'VUMeter', options);

    // Mirrored states of AudioWorkletProcessor.
    this._updatingInterval = options.updatingInterval;
    this._volume = 0;
  }

  get updatingInterval() {
    return this._updatingInterval;
  }

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

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

  // handle updated values from audio side
  onmessage (event) {
    if (event.data.hasOwnProperty('volume'))
      this._volume = event.data.volume;
  }
}

var importAudioWorkletNode = window.audioWorklet.import('vumeterprocessor.js');
            

AudioWorkletGlobalScope: vumeterprocessor.js

var VU_METER_SMOOTHING = 0.9;

registerAudioWorkletProcessor('VUMeter', class extends AudioWorkletProcessor {

  // Note: AudioParam definition can be omitted.

  constructor (options) {
    super(options);

    this.volume = 0;
    this.updatingInterval = options.updatingInterval;
    this.nextUpdateFrames = this.interval;
  }

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

  process (inputs, outputs, parameters) {
    // Note that the input will be downmixed to mono.
    var buffer = inputs[0][0];
    var bufferLength = buffer.length;
    var sum = 0, x = 0, rms = 0;

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

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

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

  onmessage (event) {
    if (event.data.hasOwnProperty('updatingInterval'))
      this.updatingInterval = event.data.updatingInterval;
  }
});
            

Main HTML file

<script src="vumeternode.js"></script>
<script>
  importAudioWorkletNode.then(function () {
    var context = new AudioContext();
    var microphone = GET_MICROPHONE(); // e.g) a live input from getUserMedia().
    var vuMeterNode = new VUMeterNode(context, { updatingInterval: 50 });

    microphone.connnect(vuMeterNode);
    requestAnimationFrame(function () {
       vuMeterNode.draw();
    });
  });
</script>
            

The ScriptProcessorNode Interface - DEPRECATED

This interface is an AudioNode which can generate, process, or analyse audio directly using JavaScript. This node type is deprecated, to be replaced by the AudioWorkletNode; this text is only here for informative purposes until implementations remove this node type.

    numberOfInputs  : 1
    numberOfOutputs : 1

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

There are channelCount constraints and channelCountMode constraints for this node.

The ScriptProcessorNode is constructed with a bufferSize which 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. audioprocess events are only dispatched if the ScriptProcessorNode has at least one input or one output connected. Lower numbers for bufferSize will result in a lower (better) latency. Higher numbers will be necessary to avoid audio breakup and glitches. This value will be picked by the implementation if the bufferSize argument to createScriptProcessor is not passed in, or is set to 0.

numberOfInputChannels and numberOfOutputChannels determine the number of input and output channels. It is invalid for both numberOfInputChannels and numberOfOutputChannels to be zero.

var node = context.createScriptProcessor(bufferSize, numberOfInputChannels,
  numberOfOutputChannels);
attribute EventHandler onaudioprocess

A property used to set the EventHandler (described in HTML[[!HTML]]) for the audioprocess event that is dispatched to ScriptProcessorNode node types. An event of type AudioProcessingEvent will be dispatched to the event handler.

readonly attribute long bufferSize

The size of the buffer (in sample-frames) which needs to be processed each time onaudioprocess is called. Legal values are (256, 512, 1024, 2048, 4096, 8192, 16384).

The AudioProcessingEvent Interface - DEPRECATED

This is an Event object which is dispatched to ScriptProcessorNode nodes. It will be removed when the ScriptProcessorNode is removed, as the replacement AudioWorkletNode uses a different approach.

The event handler processes audio from the input (if any) by accessing the audio data from the inputBuffer 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 outputBuffer.

readonly attribute double playbackTime

The time when the audio will be played in the same time coordinate system as the AudioContext's currentTime.

readonly attribute AudioBuffer inputBuffer

An AudioBuffer containing the input audio data. It will have a number of channels equal to the numberOfInputChannels parameter of the createScriptProcessor() method. This AudioBuffer is only valid while in the scope of the onaudioprocess function. Its values will be meaningless outside of this scope.

readonly attribute AudioBuffer outputBuffer

An AudioBuffer where the output audio data should be written. It will have a number of channels equal to the numberOfOutputChannels parameter of the createScriptProcessor() method. Script code within the scope of the onaudioprocess function is expected to modify the Float32Array arrays representing channel data in this AudioBuffer. Any script modifications to this AudioBuffer outside of this scope will not produce any audible effects.

The PannerNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

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

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

There are channelCount constraints and channelCountMode constraints for this node.

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

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

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

equalpower
A simple and efficient spatialization algorithm using equal-power panning.
When this panning model is used, all the AudioParams used to compute the output of this node are a-rate.
HRTF
A higher quality spatialization algorithm using a convolution with measured impulse responses from human subjects. This panning method renders stereo output.
When this panning model is used, all the AudioParams used to compute the output of this node are k-rate.

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

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

linear

A linear distance model which calculates distanceGain according to:

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

That is, \(d\) is clamped to the interval \([d_{ref},\, d_{max}]\).

inverse

An inverse distance model which calculates distanceGain according to:

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

That is, \(d\) is clamped to the interval \([d_{ref},\, \infty)\).

exponential

An exponential distance model which calculates distanceGain according to:

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

That is, \(d\) is clamped to the interval \([d_{ref},\, \infty)\).

attribute PanningModelType panningModel

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

readonly attribute AudioParam positionX

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

readonly attribute AudioParam positionY

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

readonly attribute AudioParam positionZ

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

readonly attribute AudioParam orientationX

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

readonly attribute AudioParam orientationY

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

readonly attribute AudioParam orientationZ

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

attribute DistanceModelType distanceModel

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

attribute double refDistance

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

attribute double maxDistance

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

attribute double rolloffFactor

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

attribute double coneInnerAngle

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

attribute double coneOuterAngle

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

attribute double coneOuterGain

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

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

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

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

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

The default value is (0,0,0)

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

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

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

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

The default value is (1,0,0)

PannerOptions

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

PanningModelType panningModel
The panning model to use for the node.
DistanceModelType distanceModel
The distance model to use for the node.
float positionX
The initial X value for the positionX AudioParam.
float positionY
The initial Y value for the positionY AudioParam.
float positionZ
The initial Z value for the positionZ AudioParam.
float orientationX
The initial X value for the orientationX AudioParam.
float orientationY
The initial Y value for the orientationY AudioParam.
float orientationZ
The initial Z value for the orientationZ AudioParam.
double refDistance
The initial value for the refDistance attribute of the node.
double maxDistance
The initial value for the maxDistance attribute of the node.
double rolloffFactor
The initial value for the rolloffFactor attribute of the node.
double coneInnerAngle
The initial value for the coneInnerAngle attribute of the node.
double coneOuterAngle
The initial value for the coneOuterAngle attribute of the node.
double coneOuterGain
The initial value for the coneOuterGain attribute of the node.

Channel Limitations

The set of channel limitations for StereoPannerNode also apply to PannerNode.

The AudioListener Interface

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

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

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

readonly attribute AudioParam positionX

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

readonly attribute AudioParam positionY

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

readonly attribute AudioParam positionZ

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

readonly attribute AudioParam forwardX

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

readonly attribute AudioParam forwardY

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

readonly attribute AudioParam forwardZ

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

readonly attribute AudioParam upX

Sets the x coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate.

readonly attribute AudioParam upY

Sets the y coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space. The default value is 1. This parameter is a-rate.

readonly attribute AudioParam upZ

Sets the z coordinate component of the up direction the listener is pointing in 3D Cartesian coordinate space. The default value is 0. This parameter is a-rate.

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

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

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

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

The default value is (0,0,0)

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

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

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

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

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

The StereoPannerNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

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

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

There are channelCount constraints and channelCountMode constraints for this node.

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

This node has no tail-time reference.

readonly attribute AudioParam pan

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

StereoPannerOptions

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

float pan
The initial value for the pan AudioParam.

Channel Limitations

Because its processing is constrained by the above definitions, StereoPannerNode is limited to mixing no more than 2 channels of audio, and producing exactly 2 channels. It is possible to use a ChannelSplitterNode, intermediate processing by a subgraph of GainNodes and/or other nodes, and recombination via a ChannelMergerNode to realize arbitrary approaches to panning and mixing.

The ConvolverNode Interface

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

    numberOfInputs  : 1
    numberOfOutputs : 1

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

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

There are channelCount constraints and channelCountMode constraints for this node.

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

attribute AudioBuffer? buffer

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

attribute boolean normalize

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

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

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


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

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

    var power = 0;

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

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

        power += channelPower;
    }

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

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

    var scale = 1 / power;

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

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

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

    return scale;
}
      

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

ConvolverOptions

The specifies options for constructing a ConvolverNode. All members are optional; if not specified, the node is contructing using the normal defaults.

AudioBuffer? buffer
The desired buffer for the ConvolverNode. This buffer will be normalized according to the value of disableNormalization.
boolean disableNormalization = false
The desired initial value for the normalize attribute of the ConvolverNode.

Channel Configurations for Input, Impulse Response and Output

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

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

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

reverb matrixing
A graphical representation of supported input and output channel count possibilities when using a ConvolverNode.

The AnalyserNode Interface

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

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

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

This node has no tail-time reference.

void getFloatFrequencyData()

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

The frequency data are in dB units.

Float32Array array
This parameter is where the frequency-domain analysis data will be copied.
void getByteFrequencyData()

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

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

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

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

Uint8Array array
This parameter is where the frequency-domain analysis data will be copied.
void getFloatTimeDomainData()

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

Float32Array array
This parameter is where the time-domain sample data will be copied.
void getByteTimeDomainData()

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

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

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

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

Uint8Array array
This parameter is where the time-domain sample data will be copied.
attribute unsigned long fftSize

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

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

readonly attribute unsigned long frequencyBinCount

Half the FFT size.

attribute float minDecibels

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

attribute float maxDecibels

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

attribute 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.

AnalyserOptions

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

unsigned long fftSize = 2048
The desired initial size of the FFT for frequency-domain analysis.
float maxDecibels = -30
The desired initial maximum power in dB for FFT analysis.
float minDecibels = -100
The desired initial minimum power in dB for FFT analysis.
float smoothingTimeConstant = 0.8
The desired initial smoothing constant for the FFT analysis.

FFT Windowing and smoothing over time

When the current frequency data are computed, the following operations are to be performed:
  1. Down-mix all channels of the time domain input data to mono assuming a channelCount of 1, channelCountMode of "max" and channelInterpetation of "speakers". This is independent of the settings for the AnalyserNode itself. The most recent fftSizeframes are used for the down-mixing operation.
  2. Apply a Blackman window to the time domain input data
  3. Apply a Fourier tranform to the windowed time domain input data to get imaginary and real frequency data
  4. Smooth over time the frequency domain data
  5. Conversion to dB.

In the following, let \(N\) be the value of the .fftSize attribute of this AnalyserNode.

Applying a Blackman window consists in the following operation on the input time domain data. Let \(x[n]\) for \(n = 0, \ldots, N - 1\) be the time domain data. The Blackman window is defined by

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

The windowed signal \(\hat{x}[n]\) is

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

Applying a Fourier tranform consists of computing the Fourier transform in the following way. Let \(X[k]\) be the complex frequency domain data and \(\hat{x}[n]\) be the windowed time domain data computed above. Then

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

for \(k = 0, \dots, N/2-1\).

Smoothing over time frequency data consists in the following operation:

Then the smoothed value, \(\hat{X}[k]\), is computed by

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

for \(k = 0, \ldots, N - 1\).

Conversion to dB consists of the following operation, where \(\hat{X}[k]\) is computed in smoothing over time:

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

for \(k = 0, \ldots, N-1\).

This array, \(Y[k]\), is copied to the output array for getFloatFrequencyData. For getByteFrequencyData, the \(Y[k]\) is clipped to lie between minDecibels and maxDecibels and then scaled to fit in an unsigned byte such that minDecibels is represented by the value 0 and maxDecibels is represented by the value 255.

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.

This node has no tail-time reference.

Example:

channel splitter
A diagram of a ChannelSplitter

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.

ChannelSplitterOptions

unsigned long numberOfOutputs = 6
The number outputs for the ChannelSplitterNode.

The ChannelMergerNode Interface

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

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

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

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

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

There are channelCount constraints and channelCountMode constraints for this node.

This node has no tail-time reference.

Example:

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

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

channel merger
A diagram of ChannelMerger

ChannelMergerOptions

unsigned long numberOfInputs = 6
The number inputs for the ChannelSplitterNode.

The DynamicsCompressorNode Interface

DynamicsCompressorNode is an AudioNode processor implementing a dynamics compression effect.

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

    numberOfInputs  : 1
    numberOfOutputs : 1

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

This node has no tail-time reference.

readonly attribute AudioParam threshold

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

readonly attribute AudioParam knee

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

readonly attribute AudioParam ratio

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

readonly attribute float reduction

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

readonly attribute AudioParam attack

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

readonly attribute AudioParam release

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

DynamicsCompressorOptions

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

float attack
The initial value for the attack AudioParam.
float knee
The initial value for the knee AudioParam.
float ratio
The initial value for the ratio AudioParam.
float release
The initial value for the release AudioParam.
float threshold
The initial value for the threshold AudioParam.

The BiquadFilterNode Interface

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

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

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

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

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

    numberOfInputs  : 1
    numberOfOutputs : 1

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

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

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

lowpass

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

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

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

frequency
The cutoff frequency below which the frequencies are attenuated
Q
Controls how peaked the response will be at the cutoff frequency. A large value makes the response more peaked. Please note that for this filter type, this value is not a traditional Q, but is a resonance value in decibels.
gain
Not used in this filter type
bandpass

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

frequency
The center of the frequency band
Q
Controls the width of the band. The width becomes narrower as the Q value increases.
gain
Not used in this filter type
lowshelf

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

frequency
The upper limit of the frequences where the boost (or attenuation) is applied.
Q
Not used in this filter type.
gain
The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.
highshelf

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

frequency
The lower limit of the frequences where the boost (or attenuation) is applied.
Q
Not used in this filter type.
gain
The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.
peaking

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

frequency
The center frequency of where the boost is applied.
Q
Controls the width of the band of frequencies that are boosted. A large value implies a narrow width.
gain
The boost, in dB, to be applied. If the value is negative, the frequencies are attenuated.
notch

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

frequency
The center frequency of where the notch is applied.
Q
Controls the width of the band of frequencies that are attenuated. A large value implies a narrow width.
gain
Not used in this filter type.
allpass

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

frequency
The frequency where the center of the phase transition occurs. Viewed another way, this is the frequency with maximal group delay.
Q
Controls how sharp the phase transition is at the center frequency. A larger value implies a sharper transition and a larger group delay.
gain
Not used in this filter type.

All attributes of the BiquadFilterNode are a-rate AudioParam.

attribute BiquadFilterType type

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

readonly attribute AudioParam frequency

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

readonly attribute AudioParam detune

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

readonly attribute AudioParam Q

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

readonly attribute AudioParam gain

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

void getFrequencyResponse()

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

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

Float32Array frequencyHz

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

Float32Array magResponse

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

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

Float32Array phaseResponse

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

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

BiquadFilterOptions

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

BiquadFilterType type
The desired initial type of the filter.
float Q
The desired initial value for Q.
float detune
The desired initial value for detune.
float frequency
The desired initial value for frequency.
float gain
The desired initial value for gain.

Filters characteristics

There are multiple ways of implementing the type of filters available through the BiquadFilterNode each having very different characteristics. The formulas in this section describe the filters that a conforming implementation MUST implement, as they determine the characteristics of the different filter types. They are inspired by formulas found in the Audio EQ Cookbook.

The transfer function for the filters implemented by the BiquadFilterNode is:

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

The initial filter state is 0.

The coefficients in the transfer function above are different for each node type. The following intermediate variable are necessary for their computation, based on the computedValue of the AudioParams of the BiquadFilterNode.
  • Let \(F_s\) be the value of the sampleRate attribute for this AudioContext.
  • Let \(f_0\) be the value of the computedFrequency.
  • Let \(G\) be the value of the gain AudioParam.
  • Let \(Q\) be the value of the Q AudioParam.
  • Finally let
    $$
    \begin{align*}
      A        &= 10^{\frac{G}{40}} \\
      \omega_0 &= 2\pi\frac{f_0}{F_s} \\
      \alpha_Q &= \frac{\sin\omega_0}{2Q} \\
      \alpha_{Q_{dB}} &= \frac{\sin\omega_0}{2 \cdot 10^{Q/20}} \\
      S        &= 1 \\
      \alpha_S &= \frac{\sin\omega_0}{2}\sqrt{\left(A+\frac{1}{A}\right)\left(\frac{1}{S}-1\right)+2}
    \end{align*}
    $$
                
    
The six coefficients (\(b_0, b_1, b_2, a_0, a_1, a_2\)) for each filter type, are:
lowpass
                $$
                  \begin{align*}
                    b_0 &= \frac{1 - \cos\omega_0}{2} \\
                    b_1 &= 1 - \cos\omega_0 \\
                    b_2 &= \frac{1 - \cos\omega_0}{2} \\
                    a_0 &= 1 + \alpha_{Q_{dB}} \\
                    a_1 &= -2 \cos\omega_0 \\
                    a_2 &= 1 - \alpha_{Q_{dB}}
                  \end{align*}
                $$
              
highpass
                  $$
                    \begin{align*}
                      b_0 &= \frac{1 + \cos\omega_0}{2} \\
                      b_1 &= -(1 + \cos\omega_0) \\
                      b_2 &= \frac{1 + \cos\omega_0}{2} \\
                      a_0 &= 1 + \alpha_{Q_{dB}} \\
                      a_1 &= -2 \cos\omega_0 \\
                      a_2 &= 1 - \alpha_{Q_{dB}}
                    \end{align*}
                  $$
              
bandpass
              $$
                \begin{align*}
                  b_0 &= \alpha_Q \\
                  b_1 &= 0 \\
                  b_2 &= -\alpha_Q \\
                  a_0 &= 1 + \alpha_Q \\
                  a_1 &= -2 \cos\omega_0 \\
                  a_2 &= 1 - \alpha_Q
                \end{align*}
              $$
            
notch
                $$
                  \begin{align*}
                    b_0 &= 1 \\
                    b_1 &= -2\cos\omega_0 \\
                    b_2 &= 1 \\
                    a_0 &= 1 + \alpha_Q \\
                    a_1 &= -2 \cos\omega_0 \\
                    a_2 &= 1 - \alpha_Q
                  \end{align*}
                $$
              
allpass
                $$
                  \begin{align*}
                    b_0 &= 1 - \alpha_Q \\
                    b_1 &= -2\cos\omega_0 \\
                    b_2 &= 1 + \alpha_Q \\
                    a_0 &= 1 + \alpha_Q \\
                    a_1 &= -2 \cos\omega_0 \\
                    a_2 &= 1 - \alpha_Q
                  \end{align*}
                $$
              
peaking
                $$
                  \begin{align*}
                    b_0 &= 1 + \alpha_Q\, A \\
                    b_1 &= -2\cos\omega_0 \\
                    b_2 &= 1 - \alpha_Q\,A \\
                    a_0 &= 1 + \frac{\alpha_Q}{A} \\
                    a_1 &= -2 \cos\omega_0 \\
                    a_2 &= 1 - \frac{\alpha_Q}{A}
                  \end{align*}
                $$
              
lowshelf
                $$
                  \begin{align*}
                    b_0 &= A \left[ (A+1) - (A-1) \cos\omega_0 + 2 \alpha_S \sqrt{A})\right] \\
                    b_1 &= 2 A \left[ (A-1) - (A+1) \cos\omega_0 )\right] \\
                    b_2 &= A \left[ (A+1) - (A-1) \cos\omega_0 - 2 \alpha_S \sqrt{A}) \right] \\
                    a_0 &= (A+1) + (A-1) \cos\omega_0 + 2 \alpha_S \sqrt{A} \\
                    a_1 &= -2 \left[ (A-1) + (A+1) \cos\omega_0\right] \\
                    a_2 &= (A+1) + (A-1) \cos\omega_0 - 2 \alpha_S \sqrt{A})
                  \end{align*}
                $$
              
highshelf
                $$
                  \begin{align*}
                    b_0 &= A\left[ (A+1) + (A-1)\cos\omega_0 + 2\alpha_S\sqrt{A} )\right] \\
                    b_1 &= -2A\left[ (A-1) + (A+1)\cos\omega_0 )\right] \\
                    b_2 &= A\left[ (A+1) + (A-1)\cos\omega_0 - 2\alpha_S\sqrt{A} )\right] \\
                    a_0 &= (A+1) - (A-1)\cos\omega_0 + 2\alpha_S\sqrt{A} \\
                    a_1 &= 2\left[ (A-1) - (A+1)\cos\omega_0\right] \\
                    a_2 &= (A+1) - (A-1)\cos\omega_0 - 2\alpha_S\sqrt{A}
                  \end{align*}
                $$
              

The IIRFilterNode Interface

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

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

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

    numberOfInputs  : 1
    numberOfOutputs : 1

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

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

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

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. If this array is shorter than frequencyHz a NotSupportedError MUST be signaled.
Float32Array phaseResponse
This parameter specifies an output array receiving the phase response values in radians. If this array is shorter than frequencyHz a NotSupportedError MUST be signaled.

IIRFilterOptions

The IIRFilterOptions dictionary is used to specify the filter coefficients of the IIRFilterNode.

required sequence<double> feedforward
The feedforward coefficients for the IIRFilterNode. This member is required. If not specifed, a NotFoundError MUST be thrown.
required sequence<double> feedback
The feedback coefficients for the IIRFilterNode. This member is required. If not specifed, a NotFoundError MUST be thrown.

Filter Definition

Let \(b_m\) be the feedforward coefficients and \(a_n\) be the feedback coefficients specified by createIIRFilter. Then the transfer function of the general IIR filter is given by

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

where \(M + 1\) is the length of the \(b\) array and \(N + 1\) is the length of the \(a\) array. The coefficient \(a_0\) cannot be 0. At least one of \(b_m\) must be non-zero.

Equivalently, the time-domain equation is:

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

The initial filter state is the all-zeroes state.

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.

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

none
Don't oversample
2x
Oversample two times
4x
Oversample four times
attribute Float32Array? curve

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

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

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

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

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

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.

WaveShaperOptions

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

sequence<float> curve
The shaping curve for the waveshaping effect.
OverSampleType oversample
The type of oversampling to use for the shaping curve.

The OscillatorNode Interface

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

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

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

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

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

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

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

The OscillatorNode's instantaneous phase at each time is the time integral of computedFrequency. Its nominal range is [-Nyquist frequency, Nyquist frequency].

  numberOfInputs  : 0
  numberOfOutputs : 1 (mono output)
sine
A sine wave
square
A square wave of duty period 0.5
sawtooth
A sawtooth wave
triangle
A triangle wave
custom
A custom periodic wave
attribute OscillatorType type

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

readonly attribute AudioParam frequency

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

readonly attribute AudioParam detune

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

void start()

Schedules a sound to playback at an exact time. This behaves exactly the same as the start() method for an AudioBufferSourceNode except the optional offset and duration parameters are not allowed.

optional double when = 0
Defined the same as the when parameter of the start() method of the AudioBufferSourceNode.
void stop()

Schedules a sound to stop playback at an exact time. This behaves exactly the same as the stop() method for an AudioBufferSourceNode

optional double when = 0
Defined the same as the when parameter of the stop() method of the 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.

OscillatorOptions

This specifies the options to be used when constructing an OscillatorNode. All of the members are optional; if not specified, the normal default values are used for constructing the oscillator.

OscillatorType type
The type of oscillator to be constructed. If this is set to "custom" without also specifying a periodicWave, then an InvalidStateError exception MUST be thrown.
float frequency
The initial frequency for the OscillatorNode.
float detune
The initial detune value for the OscillatorNode.
PeriodicWave periodicWave
The PeriodicWave for the OscillatorNode. If this is specified, then the type member must either be unspecified or set to "custom". If this is not true, an InvalidStateError exception MUST be thrown.

Basic Waveform Phase

The idealized mathematical waveforms for the various oscillator types are defined here. In summary, all waveforms are defined mathematically to be an odd function with a positive slope at time 0. The actual waveforms produced by the oscillator may differ to prevent aliasing affects.

The oscillator must produce the same result as if a PeriodicWave with the appropriate Fourier series and with normalization enabled were used to create these basic waveforms.

"sine"
The waveform for sine oscillator is:
                $$
                  x(t) = \sin t
                $$.
              
"square"
The waveform for the square wave oscillator is:
                $$
                  x(t) = \begin{cases}
                         1 & \mbox{for } 0≤ t < \pi \\
                         -1 & \mbox{for } -\pi < t < 0.
                         \end{cases}
                $$
              
"sawtooth"
The waveform for the sawtooth oscillator is the ramp:
                $$
                  x(t) = \frac{t}{\pi} \mbox{ for } -\pi < t ≤ \pi;
                $$
              
"triangle"
The waveform for the triangle oscillator is:
                $$
                  x(t) = \begin{cases}
                           \frac{2}{\pi} t & \mbox{for } 0 ≤ t ≤ \frac{\pi}{2} \\
                           1-\frac{2}{\pi} (t-\frac{\pi}{2}) & \mbox{for }
                           \frac{\pi}{2} < t ≤ \pi.
                         \end{cases}
                $$
              
This is extended to all \(t\) by using the fact that the waveform is an odd function with period \(2\pi\).

The PeriodicWave Interface

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

PeriodicWaveConstraints

The PeriodicWaveConstraints dictionary is used to specify how the waveform is normalized.
boolean disableNormalization = false
Controls whether the periodic wave is normalized or not. If true, the waveform is not normalized; otherwise, the waveform is normalized.

PeriodicWaveOptions

The PeriodicWaveOptions dictionary is used to specify how the waveform is constructed. At least one of real or imag must be specified. If not given, it is the array of all zeroes of the same length as the other.

sequence<float> real
The array of cosine terms equivalent to the real parameter to createPeriodicWave. This defaults to a sequence of all zeroes if imag is given.
sequence<float> imag
The array of sine terms terms equivalent to the imag parameter to createPeriodicWave. This defaults to a sequence of all zeroes if real is given.

Waveform Generation

The createPeriodicWave() method takes two arrays to specify the Fourier coefficients of the PeriodicWave. Let \(a\) and \(b\) represent the real and imaginary arrays of length \(L\). Then the basic time-domain waveform, \(x(t)\), can be computed using:

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

This is the basic (unnormalized) waveform.

Waveform Normalization

By default, the waveform defined in the previous section is normalized so that the maximum value is 1. The normalization is done as follows.

Let

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

where \(N\) is a power of two. (Note: \(\tilde{x}(n)\) can conveniently be computed using an inverse FFT.) The fixed normalization factor \(f\) is computed as follows.

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

Thus, the actual normalized waveform \(\hat{x}(n)\) is

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

This fixed normalization factor must be applied to all generated waveforms.

Oscillator Coefficients

The builtin oscillator types are created using PeriodicWave objects. For completeness the coefficients for the PeriodicWave for each of the builtin oscillator types is given here. This is useful if a builtin type is desired but without the default normalization.

In the following descriptions, let \(a\) be the array of real coefficients and \(b\) be the array of imaginary coefficients for createPeriodicWave(). In all cases \(a[n] = 0\) for all \(n\) because the waveforms are odd functions. Also, \(b[0] = 0\) in all cases. Hence, only \(b[n]\) for \(n \ge 1\) is specified below.

"sine"
                  $$
                    b[n] = \begin{cases}
                             1 & \mbox{for } n = 1 \\
                             0 & \mbox{otherwise}
                           \end{cases}
                  $$
              
"square"
                  $$
                    b[n] = \frac{2}{n\pi}\left[1 - (-1)^n\right]
                  $$
              
"sawtooth"
                $$
                  b[n] = (-1)^{n+1} \dfrac{2}{n\pi}
                $$
            
"triangle"
                  $$
                    b[n] = \frac{8\sin\dfrac{n\pi}{2}}{(\pi n)^2}
                  $$
              

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.

This node has no tail-time reference.

MediaStreamAudioSourceOptions

This specifies the options for constructing a MediaStreamAudioSourceNode.

required MediaStream mediaStream
The media stream that will act as a source. This MUST be specified.

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).

readonly attribute MediaStream stream

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

Processing model

Background

Real-time audio systems that require low latency are often implemented using callback functions, where the operating system calls the program back when more audio has to be computed in order for the playback to stay uninterrupted. Such callback is called on a high priority thread (often the highest priority on the system). This means that a program that deals with audio only executes code from this callback, as any buffering between a rendering thread and the callback would naturally add latency or make the system less resilient to glitches.

For this reason, the traditional way of executing asynchronous operations on the Web Platform, the event loop, does not work here, as the thread is not continuously executing. Additionally, a lot of unnecessary and potentially blocking operations are available from traditional execution contexts (Windows and Workers), which is not something that is desirable to reach an acceptable level of performance.

Additionally, the Worker model makes creating a dedicated thread necessary for a script execution context, while all AudioNodes usually share the same execution context.

This section specifies how the end result should look like, not how it should be implemented. In particular, instead of using message queue, implementors can use memory that is shared between threads, as long as the memory operations are not reordered.

Control thread and rendering thread

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

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

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

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

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

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

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

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

Swapping a control message queue QA with another control message queue QB means executing the following steps:

  1. Let QC be a new, empty control message queue.
  2. Move all the control messages QA to QC.
  3. Move all the control messages QB to QA.
  4. Move all the control messages QC to QB.

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

Asynchronous operations

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

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

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

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

Rendering an audio graph

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

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

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

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

In practice, the AudioContext rendering thread is often running off a system level audio callback, that executes in an isochronous fashion. This callback passes in a buffer that has to be filled with the audio that will be output. The size of the buffer is often larger than a rendering quantum. In this case, multiple invocations of the rendering algorithm will be called in a rapid succession, in the same callback, before returning. After some time, the underlying audio system will call the callback again, and the algorithm will be executed again. This is an implementation detail that should not be observable, apart from the latency implications.

OfflineAudioContext will execute the algorithm continuously, until length (as passed in the OfflineAudioContext contructor) frames have been rendered.

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

Mixer Gain Structure

Background

One of the most important considerations when dealing with audio processing graphs is how to adjust the gain (volume) at various points. For example, in a standard mixing board model, each input bus has pre-gain, post-gain, and send-gains. Submix and master out busses also have gain control. The gain control described here can be used to implement standard mixing boards as well as other architectures.

Summing Inputs

The inputs to AudioNodes have the ability to accept connections from multiple outputs. The input then acts as a unity gain summing junction with each output signal being added with the others:

unity gain summing junction
A graph showing Source 1 and Source 2 output summed at the input of Destination

In cases where the channel layouts of the outputs do not match, a mix (usually up-mix) will occur according to the mixing rules.

No clipping is applied at the inputs or outputs of the AudioNode to allow a maximum of dynamic range within the audio graph.

Gain Control

In many scenarios, it's important to be able to control the gain for each of the output signals. The GainNode gives this control:

mixer architecture new
A graph featuring volume control for each voice

Using these two concepts of unity gain summing junctions and GainNodes, it's possible to construct simple or complex mixing scenarios.

Example: Mixer with Send Busses

In a routing scenario involving multiple sends and submixes, explicit control is needed over the volume or "gain" of each connection to a mixer. Such routing topologies are very common and exist in even the simplest of electronic gear sitting around in a basic recording studio.

Here's an example with two send mixers and a main mixer. Although possible, for simplicity's sake, pre-gain control and insert effects are not illustrated:

mixer gain structure
A graph showing a full mixer with send busses.

This diagram is using a shorthand notation where "send 1", "send 2", and "main bus" are actually inputs to AudioNodes, but here are represented as summing busses, where the intersections g2_1, g3_1, etc. represent the "gain" or volume for the given source on the given mixer. In order to expose this gain, an GainNode is used:

Here's how the above diagram could be constructed in JavaScript:


  var context = 0;
  var compressor = 0;
  var reverb = 0;
  var delay = 0;
  var s1 = 0;
  var s2 = 0;

  var source1 = 0;
  var source2 = 0;
  var g1_1 = 0;
  var g2_1 = 0;
  var g3_1 = 0;
  var g1_2 = 0;
  var g2_2 = 0;
  var g3_2 = 0;

  // Setup routing graph
  function setupRoutingGraph() {
      context = new AudioContext();

      compressor = context.createDynamicsCompressor();

      // Send1 effect
      reverb = context.createConvolver();
      // Convolver impulse response may be set here or later

      // Send2 effect
      delay = context.createDelay();

      // Connect final compressor to final destination
      compressor.connect(context.destination);

      // Connect sends 1 & 2 through effects to main mixer
      s1 = context.createGain();
      reverb.connect(s1);
      s1.connect(compressor);

      s2 = context.createGain();
      delay.connect(s2);
      s2.connect(compressor);

      // Create a couple of sources
      source1 = context.createBufferSource();
      source2 = context.createBufferSource();
      source1.buffer = manTalkingBuffer;
      source2.buffer = footstepsBuffer;

      // Connect source1
      g1_1 = context.createGain();
      g2_1 = context.createGain();
      g3_1 = context.createGain();
      source1.connect(g1_1);
      source1.connect(g2_1);
      source1.connect(g3_1);
      g1_1.connect(compressor);
      g2_1.connect(reverb);
      g3_1.connect(delay);

      // Connect source2
      g1_2 = context.createGain();
      g2_2 = context.createGain();
      g3_2 = context.createGain();
      source2.connect(g1_2);
      source2.connect(g2_2);
      source2.connect(g3_2);
      g1_2.connect(compressor);
      g2_2.connect(reverb);
      g3_2.connect(delay);

      // We now have explicit control over all the volumes g1_1, g2_1, ..., s1, s2
      g2_1.gain.value = 0.2;  // For example, set source1 reverb gain

      // Because g2_1.gain is an "AudioParam",
      // an automation curve could also be attached to it.
      // A "mixing board" UI could be created in canvas or WebGL controlling these gains.
  }

   

Dynamic Lifetime

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

dynamic allocation
A graph featuring a subgraph that will be releases early.

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);
}

Channel up-mixing and down-mixing

This section is normative.

describes how an input to an AudioNode can be connected from one or more outputs of an AudioNode. Each of these connections from an output represents a stream with a specific non-zero number of channels. An input has mixing rules for combining the channels from all of the connections to it. As a simple example, if an input is connected from a mono output and a stereo output, then the mono connection will usually be up-mixed to stereo and summed with the stereo connection. But, of course, it's important to define the exact mixing rules for every input to every AudioNode. The default mixing rules for all of the inputs have been chosen so that things "just work" without worrying too much about the details, especially in the very common case of mono and stereo streams. Of course, the rules can be changed for advanced use cases, especially multi-channel.

To define some terms, up-mixing refers to the process of taking a stream with a smaller number of channels and converting it to a stream with a larger number of channels. down-mixing refers to the process of taking a stream with a larger number of channels and converting it to a stream with a smaller number of channels.

An AudioNode input use three basic pieces of information to determine how to mix all the outputs connected to it. As part of this process it computes an internal value computedNumberOfChannels representing the actual number of channels of the input at any given time:

The AudioNode attributes involved in channel up-mixing and down-mixing rules are defined above. The following is a more precise specification on what each of them mean.

For each input of an AudioNode, an implementation must:

  1. Compute computedNumberOfChannels.
  2. For each connection to the input:
    • up-mix or down-mix the connection to computedNumberOfChannels according to channelInterpretation.
    • Mix it together with all of the other mixed streams (from other connections). This is a straight-forward mixing together of each of the corresponding channels from each connection.

Speaker Channel Layouts

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

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

Channel ordering

    Mono
      0: M: mono

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

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

Up Mixing speaker layouts

Mono up-mix:

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

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

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

Stereo up-mix:

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

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

Quad up-mix:

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

Down Mixing speaker layouts

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

  Mono down-mix:

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

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

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


  Stereo down-mix:

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

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

  Quad down-mix:

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

  

Channel Rules Examples

  // Set gain node to explicit 2-channels (stereo).
  gain.channelCount = 2;
  gain.channelCountMode = "explicit";
  gain.channelInterpretation = "speakers";

  // Set "hardware output" to 4-channels for DJ-app with two stereo output busses.
  context.destination.channelCount = 4;
  context.destination.channelCountMode = "explicit";
  context.destination.channelInterpretation = "discrete";

  // Set "hardware output" to 8-channels for custom multi-channel speaker array
  // with custom matrix mixing.
  context.destination.channelCount = 8;
  context.destination.channelCountMode = "explicit";
  context.destination.channelInterpretation = "discrete";

  // Set "hardware output" to 5.1 to play an HTMLAudioElement.
  context.destination.channelCount = 6;
  context.destination.channelCountMode = "explicit";
  context.destination.channelInterpretation = "speakers";

  // Explicitly down-mix to mono.
  gain.channelCount = 1;
  gain.channelCountMode = "explicit";
  gain.channelInterpretation = "speakers";
  

Audio Signal Values

The range of all audio signals at a destination node of any audio graph is nominally [-1, 1]. The audio rendition of signal values outside this range, or of the values NaN, positive infinity or negative infinity, is undefined by this specification.

Spatialization/Panning

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.

Equal-power panning

This is a simple and relatively inexpensive algorithm which provides basic, but reasonable results. It is used for the StereoPannerNode, and for the PannerNode when the panningModel attribute is set to "equalpower", in which case the elevation value is ignored. This algorithm MUST be implemented using a-rate parameters.

  1. For each sample to be computed by this AudioNode:
    1. Let azimuth be the value computed in the azimuth and elevation section.

    2. The azimuth value is first contained to be within the range [-90, 90] according to:

        // First, clamp azimuth to allowed range of [-180, 180].
        azimuth = max(-180, azimuth);
        azimuth = min(180, azimuth);
      
        // Then wrap to range [-90, 90].
        if (azimuth < -90)
          azimuth = -180 - azimuth;
        else if (azimuth > 90)
          azimuth = 180 - azimuth;
      
                    
    3. A normalized value x is calculated from azimuth for a mono input as:

        x = (azimuth + 90) / 180;
      
                    

      Or for a stereo input as:

        if (azimuth <= 0) { // -90 ~ 0
          // Transform the azimuth value from [-90, 0] degrees into the range [-90, 90].
          x = (azimuth + 90) / 90;
        } else { // 0 ~ 90
          // Transform the azimuth value from [0, 90] degrees into the range [-90, 90].
          x = azimuth / 90;
        }
      
                        

For a StereoPannerNode, the following algorithm MUST be implemented.

  1. For each sample to be computed by this AudioNode
    1. Let pan be the computedValue of the pan AudioParam of this StereoPannerNode.

    2. Clamp pan to [-1, 1].

          pan = max(-1, pan);
          pan = min(1, pan);
      
                    
    3. Calculate x by normalizing pan value to [0, 1]. For mono input:

          x = (pan + 1) / 2;
      
                    

      For stereo input:

          if (pan <= 0)
            x = pan + 1;
          else
            x = pan;
      
                    

    Then following steps are used to achieve equal-power panning:

    1. Left and right gain values are calculated as:

          gainL = cos(x * Math.PI / 2);
          gainR = sin(x * Math.PI / 2);
      
                    
    2. For mono input, the stereo output is calculated as:

          outputL = input * gainL;
          outputR = input * gainR;
      
                    

      Else for stereo input, the output is calculated as:

          if (pan <= 0) {
            // Pass through inputL to outputL and equal-power pan inputR as in mono case.
            outputL = inputL + inputR * gainL;
            outputR = inputR * gainR;
          } else {
            // Pass through inputR to outputR and equal-power pan inputL as in mono case.
            outputL = inputL * gainL;
            outputR = inputR + inputL * gainR;
          }
      
                    

HRTF panning (stereo only)

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

HRTF panner
A diagram showing the process of panning a source using HRTF.

Distance Effects

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

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

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

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

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

Sound Cones

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

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

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

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

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

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

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

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

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

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

  return gain;
}  

Performance Considerations

Latency

latency
Use cases in which the latency can be important

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 cumulative. 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.

Implementations will generally seek to minimize overall latency.

Along with minimizing overall latency, implementations will generally seek to minimize the difference between an AudioContext's currentTime and an AudioProcessingEvent's playbackTime. Deprecation of ScriptProcessorNode will make this consideration less important over time.

Additionally, some AudioNodes can add latency to some paths of the audio graph, notably:

Audio Buffer Copying

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

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

AudioParam Transitions

While no automatic smoothing is done when directly setting the value attribute of an AudioParam, for certain parameters, smooth transition are preferable to directly setting the value.

Using the setTargetAtTime method with a low timeConstant allows authors to perform a smooth transition.

Audio Glitching

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

Security and Privacy Considerations

The W3C TAG is developing a Self-Review Questionnaire: Security and Privacy for editors of specifications to informatively answer.

Per the Questions to Consider

  1. Does this specification deal with personally-identifiable information?

    No.

  2. Does this specification deal with high-value data?

    No. Credit card information and the like is not used in Web Audio. It is possible to use Web Audio to process or analyze voice data, which might be a privacy concern, but access to the user's microphone is permission-based via getUserMedia.

  3. Does this specification introduce new state for an origin that persists across browsing sessions?

    No. AudioWorklet does not persist across browsing sessions. right?

  4. Does this specification expose persistent, cross-origin state to the web?

    Not sure. If audio sample data is loaded cross-origin, it exposes state (whether that sample data resolves or not) to the script origin.

  5. Does this specification expose any other data to an origin that it doesn’t currently have access to?

    Yes. When giving various information on available AudioNodes, the Web Audio API potentially exposes information on characteristic features of the client (such as audio hardware sample-rate) to any page that makes use of the AudioNode interface. Additionally, timing information can be collected through the AnalyserNode or ScriptProcessorNode interface. The information could subsequently be used to create a fingerprint of the client.

  6. Does this specification enable new script execution/loading mechanisms?

    No. However, it does use the worker script execution method, defined in that specification.

  7. Does this specification allow an origin access to a user’s location?

    No.

  8. Does this specification allow an origin access to sensors on a user’s device?

    Not directly. Currently audio input is not specified in this document, but it will involve gaining access to the client machine's audio input or microphone. This will require asking the user for permission in an appropriate way, probably via the getUserMedia() API.

  9. Does this specification allow an origin access to aspects of a user’s local computing environment?

    Not sure. Does it allow probing of supported sample rates? Supported audio codecs? We should mention denial of service attack by consuming CPU cycles.

  10. Does this specification allow an origin access to other devices?

    No.

  11. Does this specification allow an origin some measure of control over a user agent’s native UI?

    No?. Though it could be used to emulate system sounds to make an attack seem more like a local system event?

  12. Does this specification expose temporary identifiers to the web?

    No.

  13. Does this specification distinguish between behavior in first-party and third-party contexts?

    No.

  14. How should this specification work in the context of a user agent’s "incognito" mode?

    No differently.

  15. Does this specification persist data to a user’s local device?

    Maybe? Cached impulses or audio sample data stored locally?

  16. Does this specification have a "Security Considerations" and "Privacy Considerations" section?

    Yes.

  17. Does this specification allow downgrading default security characteristics?

    No.

Requirements and Use Cases

Please see [[webaudio-usecases]].

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); Hofmann, Bill (Dolby Laboratories); 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.