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<pre class=metadata>
Title: Web Audio API 1.1
Shortname: webaudio
Level: 1.1
Group: audiowg
Status: ED
ED: https://webaudio.github.io/web-audio-api/
TR: https://www.w3.org/TR/webaudio-11/
Favicon: favicon.png
Previous Version: https://www.w3.org/TR/2021/REC-webaudio-20210617/
Previous Version: https://www.w3.org/TR/2021/CR-webaudio-20210114/
Previous Version: https://www.w3.org/TR/2020/CR-webaudio-20200611/
Previous Version: https://www.w3.org/TR/2018/CR-webaudio-20180918/
Previous Version: https://www.w3.org/TR/2018/WD-webaudio-20180619/
Previous Version: https://www.w3.org/TR/2015/WD-webaudio-20151208/
Previous Version: https://www.w3.org/TR/2013/WD-webaudio-20131010/
Previous Version: https://www.w3.org/TR/2012/WD-webaudio-20121213/
Previous Version: https://www.w3.org/TR/2012/WD-webaudio-20120802/
Previous Version: https://www.w3.org/TR/2012/WD-webaudio-20120315/
Previous Version: https://www.w3.org/TR/2011/WD-webaudio-20111215/
Editor: Paul Adenot, Mozilla (https://www.mozilla.org/), [email protected], w3cid 62410
Editor: Hongchan Choi, Google (https://www.google.com/), [email protected], w3cid 74103
Former Editor: Raymond Toy (until Oct 2018)
Former Editor: Chris Wilson (Until Jan 2016)
Former Editor: Chris Rogers (Until Aug 2013)
Implementation Report: implementation-report.html
Test Suite: https://github.com/web-platform-tests/wpt/tree/master/webaudio
Repository: WebAudio/web-audio-api
Abstract: This specification describes a high-level Web <abbr title="Application Programming Interface">API</abbr>
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 [[#AudioWorklet|direct script 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 <code>responseType</code> and <code>response</code> attributes).
For games and interactive applications,
it is anticipated to be used with the <code>canvas</code> 2D [[2dcontext]]
and WebGL [[WEBGL]] 3D graphics APIs.
Markup Shorthands: markdown on, dfn on, css off
</pre>
<pre class=anchors>
spec: ECMAScript; url: https://tc39.github.io/ecma262/#sec-data-blocks; type: dfn; text: data block;
url: https://www.w3.org/TR/mediacapture-streams/#dom-mediadevices-getusermedia; type: method; for: MediaDevices; text: getUserMedia()
</pre>
<!-- We want {{object}} and {{Promise}} to go to the WebIDL object -->
<pre class=link-defaults>
spec:webidl; type:interface; text:object
spec:webidl; type:interface; text:Promise
</pre>
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<h2 id="introductory" class=no-num>
Introduction</h2>
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
<i>any</i> use case which could reasonably be implemented with an
optimized C++ engine controlled via script 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.
<h3 id="Features">
Features</h3>
The API supports these primary features:
* [[#ModularRouting|Modular routing]] for simple or complex
mixing/effect architectures.
* High dynamic range, using 32-bit floats for internal processing.
* [[#AudioParam|Sample-accurate scheduled sound playback]]
with low [[#latency|latency]] for musical applications
requiring a very high degree of rhythmic precision such as drum
machines and sequencers. This also includes the possibility of
[[#DynamicLifetime|dynamic creation]] of effects.
* Automation of audio parameters for envelopes, fade-ins /
fade-outs, granular effects, filter sweeps, LFOs etc.
* Flexible handling of channels in an audio stream, allowing them
to be split and merged.
* Processing of audio sources from an <{audio}> or <{video}>
{{MediaElementAudioSourceNode|media element}}.
* Processing live audio input using a {{MediaStreamTrackAudioSourceNode|MediaStream}} from
{{getUserMedia()}}.
* Integration with WebRTC
* Processing audio received from a remote peer using a
{{MediaStreamTrackAudioSourceNode}} and
[[!webrtc]].
* Sending a generated or processed audio stream to a remote
peer using a {{MediaStreamAudioDestinationNode}}
and [[!webrtc]].
* Audio stream synthesis and processing [[#AudioWorklet|directly using scripts]].
* [[#Spatialization|Spatialized audio]] supporting a wide
range of 3D games and immersive environments:
* Panning models: equalpower, HRTF, pass-through
* Distance Attenuation
* Sound Cones
* Obstruction / Occlusion
* Source / Listener based
* A convolution engine for a wide
range of linear effects, especially very high-quality room effects.
Here are some examples of possible effects:
* Small / large room
* Cathedral
* Concert hall
* Cave
* Tunnel
* Hallway
* Forest
* Amphitheater
* Sound of a distant room through a doorway
* Extreme filters
* Strange backwards effects
* Extreme comb filter effects
* Dynamics compression for overall control and sweetening of the mix
* Efficient [[#AnalyserNode|real-time time-domain and frequency-domain analysis / music visualizer support]].
* Efficient biquad filters for lowpass, highpass, and other common filters.
* A Waveshaping effect for distortion and other non-linear effects
* Oscillators
<h4 id="ModularRouting">
Modular Routing</h4>
Modular routing allows arbitrary connections between different
{{AudioNode}} objects. Each node can have
<dfn>inputs</dfn> and/or <dfn>outputs</dfn>.
A <dfn>source node</dfn> has no inputs and a single output.
A <dfn>destination node</dfn> 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;
<a href="#channel-up-mixing-and-down-mixing">the right thing just happens</a>.
For example, if a mono audio stream is connected to a
stereo input it should just mix to left and right channels
[[#channel-up-mixing-and-down-mixing|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}}:
<figure>
<img alt="modular routing" src="images/modular-routing1.png" width="305" height="128">
<figcaption>
A simple example of modular routing.
</figcaption>
</figure>
Illustrating this simple routing, here's a simple example playing a single sound:
<pre class="example" highlight="js">
const context = new AudioContext();
function playSound() {
const source = context.createBufferSource();
source.buffer = dogBarkingBuffer;
source.connect(context.destination);
source.start(0);
}
</pre>
Here's a more complex example with three sources and a convolution
reverb send with a dynamics compressor at the final output stage:
<figure>
<img alt="modular routing2" src="images/modular-routing2.png" width="561" height="411">
<figcaption>
A more complex example of modular routing.
</figcaption>
</figure>
<pre class="example" highlight="js" line-numbers>
let context;
let compressor;
let reverb;
let source1, source2, source3;
let lowpassFilter;
let waveShaper;
let panner;
let dry1, dry2, dry3;
let wet1, wet2, wet3;
let mainDry;
let mainWet;
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 main wet and dry.
mainDry = context.createGain();
mainWet = context.createGain();
// Connect final compressor to final destination.
compressor.connect(context.destination);
// Connect main dry and wet to compressor.
mainDry.connect(compressor);
mainWet.connect(compressor);
// Connect reverb to main wet.
reverb.connect(mainWet);
// 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(mainDry);
wet1.connect(reverb);
// Connect source2
dry2 = context.createGain();
wet2 = context.createGain();
source2.connect(waveShaper);
waveShaper.connect(dry2);
waveShaper.connect(wet2);
dry2.connect(mainDry);
wet2.connect(reverb);
// Connect source3
dry3 = context.createGain();
wet3 = context.createGain();
source3.connect(panner);
panner.connect(dry3);
panner.connect(wet3);
dry3.connect(mainDry);
wet3.connect(reverb);
// Start the sources now.
source1.start(0);
source2.start(0);
source3.start(0);
}
</pre>
Modular routing also permits the output of
{{AudioNode}}s 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.
<figure>
<img alt="modular routing3" src="images/modular-routing3.png"
width="346" height="337">
<figcaption>
Modular routing illustrating one Oscillator modulating the
frequency of another.
</figcaption>
</figure>
<pre class="example" highlight="js" line-numbers>
function setupRoutingGraph() {
const context = new AudioContext();
// Create the low frequency oscillator that supplies the modulation signal
const lfo = context.createOscillator();
lfo.frequency.value = 1.0;
// Create the high frequency oscillator to be modulated
const hfo = context.createOscillator();
hfo.frequency.value = 440.0;
// Create a gain node whose gain determines the amplitude of the modulation signal
const 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);
}
</pre>
<h3 id="APIOverview">
API Overview</h3>
The interfaces defined are:
* An <a class="dfnref" href="#AudioContext">AudioContext</a>
interface, which contains an audio signal graph representing
connections between {{AudioNode}}s.
* An {{AudioNode}} interface, which represents
audio sources, audio outputs, and intermediate processing modules.
{{AudioNode}}s can be dynamically connected together
in a [[#ModularRouting|modular fashion]].
{{AudioNode}}s exist in the context of an
{{AudioContext}}.
* An {{AnalyserNode}} interface, an
{{AudioNode}} for use with music visualizers, or
other visualization applications.
* An {{AudioBuffer}} interface, for working with
memory-resident audio assets. These can represent one-shot sounds, or
longer audio clips.
* An {{AudioBufferSourceNode}} interface, an
{{AudioNode}} which generates audio from an
AudioBuffer.
* An {{AudioDestinationNode}} interface, an
{{AudioNode}} subclass representing the final
destination for all rendered audio.
* An {{AudioParam}} interface, for controlling an
individual aspect of an {{AudioNode}}'s functioning,
such as volume.
* An {{AudioListener}} interface, which works with
a {{PannerNode}} for spatialization.
* An {{AudioWorklet}} interface representing a
factory for creating custom nodes that can process audio directly
using scripts.
* An {{AudioWorkletGlobalScope}} interface, the
context in which AudioWorkletProcessor processing scripts run.
* An {{AudioWorkletNode}} interface, an
{{AudioNode}} representing a node processed in an
AudioWorkletProcessor.
* An {{AudioWorkletProcessor}} interface,
representing a single node instance inside an audio worker.
* A {{BiquadFilterNode}} interface, an
{{AudioNode}} for common low-order filters such as:
* Low Pass
* High Pass
* Band Pass
* Low Shelf
* High Shelf
* Peaking
* Notch
* Allpass
* A {{ChannelMergerNode}} interface, an
{{AudioNode}} for combining channels from multiple
audio streams into a single audio stream.
* A {{ChannelSplitterNode}} interface, an {{AudioNode}} for accessing the individual channels of an
audio stream in the routing graph.
* A {{ConstantSourceNode}} interface, an
{{AudioNode}} for generating a nominally constant output value
with an {{AudioParam}} to allow automation of the value.
* A {{ConvolverNode}} interface, an
{{AudioNode}} for applying a
real-time linear effect (such as the sound of
a concert hall).
* A {{DelayNode}} interface, an
{{AudioNode}} which applies a dynamically adjustable
variable delay.
* A {{DynamicsCompressorNode}} interface, an
{{AudioNode}} for dynamics compression.
* A {{GainNode}} interface, an
{{AudioNode}} for explicit gain control.
* An {{IIRFilterNode}} interface, an
{{AudioNode}} for a general IIR filter.
* A {{MediaElementAudioSourceNode}} interface, an
{{AudioNode}} which is the audio source from an
<{audio}>, <{video}>, or other media element.
* A {{MediaStreamAudioSourceNode}} interface, an
{{AudioNode}} which is the audio source from a
{{MediaStream}} such as live audio input, or from a remote peer.
* A {{MediaStreamTrackAudioSourceNode}} interface,
an {{AudioNode}} which is the audio source from a
{{MediaStreamTrack}}.
* A {{MediaStreamAudioDestinationNode}} interface,
an {{AudioNode}} which is the audio destination to a
{{MediaStream}} sent to a remote peer.
* A {{PannerNode}} interface, an
{{AudioNode}} for spatializing / positioning audio in
3D space.
* A {{PeriodicWave}} interface for specifying
custom periodic waveforms for use by the
{{OscillatorNode}}.
* An {{OscillatorNode}} interface, an
{{AudioNode}} for generating a periodic waveform.
* A {{StereoPannerNode}} interface, an
{{AudioNode}} for equal-power positioning of audio
input in a stereo stream.
* A {{WaveShaperNode}} interface, an
{{AudioNode}} which applies a non-linear waveshaping
effect for distortion and other more subtle warming effects.
There are also several features that have been deprecated from the
Web Audio API but not yet removed, pending implementation experience
of their replacements:
* A {{ScriptProcessorNode}} interface, an {{AudioNode}} for generating or processing audio directly
using scripts.
* An {{AudioProcessingEvent}} interface, which is
an event type used with {{ScriptProcessorNode}}
objects.
<h2 id="audioapi">
The Audio API</h2>
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<h3 interface lt="BaseAudioContext" id="BaseAudioContext">
The {{BaseAudioContext}} Interface</h3>
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 [[#ModularRouting|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).
{{BaseAudioContext}} are created with an internal slot
<dfn attribute for="BaseAudioContext">[[pending promises]]</dfn> that is an
initially empty ordered list of promises.
Each {{BaseAudioContext}} has a unique
<a href="https://html.spec.whatwg.org/multipage/media.html#media-element-event-task-source">
media element event task source</a>.
Additionally, a {{BaseAudioContext}} has several private slots <dfn attribute
for="BaseAudioContext">[[rendering thread state]]</dfn> and <dfn attribute
for="BaseAudioContext">[[control thread state]]</dfn> that take values from
{{AudioContextState}}, and that are both initially set to <code>"suspended"
</code>, and a private slot <dfn attribute for="BaseAudioContext">
[[render quantum size]]</dfn> that is an unsigned integer.
<pre class="idl">
enum AudioContextState {
"suspended",
"running",
"closed",
"interrupted"
};
</pre>
<div class="enum-description">
<table class="simple" dfn-for="AudioContextState" dfn-type="enum-value">
<caption>{{AudioContextState}} enumeration description</caption>
<thead>
<tr>
<th>Enum value</th><th>Description</th>
<tbody>
<tr>
<td>
"<dfn>suspended</dfn>"
<td>
This context is currently suspended (context time is not
proceeding, audio hardware may be powered down/released).
<tr>
<td>
"<dfn>running</dfn>"
<td>
Audio is being processed.
<tr>
<td>
"<dfn>closed</dfn>"
<td>
This context has been released, and can no longer be used to
process audio. All system audio resources have been released.
<tr>
<td>
"<dfn>interrupted</dfn>"
<td>
This context is currently interrupted and cannot process audio
until the [=interruption=] ends.
</table>
</div>
<pre class="idl">
enum AudioContextRenderSizeCategory {
"default",
"hardware"
};
</pre>
<div class="enum-description">
<table class="simple" dfn-for="AudioContextRenderSizeCategory" dfn-type="enum-value">
<thead>
<tr>
<th scope="col" colspan="2">
Enumeration description
</th>
</tr>
<tbody>
<tr>
<td>
"<dfn>default</dfn>"
<td>
The AudioContext's <a>render quantum size</a> is the default value of 128
frames.
<tr>
<td>
"<dfn>hardware</dfn>"
</td>
<td>
The User-Agent picks a <a>render quantum size</a> that is best for the
current configuration.
Note: This exposes information about the host and can be used for fingerprinting.
</td>
</tr>
</table>
</div>
<xmp class="idl">
callback DecodeErrorCallback = undefined (DOMException error);
callback DecodeSuccessCallback = undefined (AudioBuffer decodedData);
[Exposed=Window]
interface BaseAudioContext : EventTarget {
readonly attribute AudioDestinationNode destination;
readonly attribute float sampleRate;
readonly attribute double currentTime;
readonly attribute AudioListener listener;
readonly attribute AudioContextState state;
readonly attribute unsigned long renderQuantumSize;
[SameObject, SecureContext]
readonly attribute AudioWorklet audioWorklet;
attribute EventHandler onstatechange;
AnalyserNode createAnalyser ();
BiquadFilterNode createBiquadFilter ();
AudioBuffer createBuffer (unsigned long numberOfChannels,
unsigned long length,
float sampleRate);
AudioBufferSourceNode createBufferSource ();
ChannelMergerNode createChannelMerger (optional unsigned long numberOfInputs = 6);
ChannelSplitterNode createChannelSplitter (
optional unsigned long numberOfOutputs = 6);
ConstantSourceNode createConstantSource ();
ConvolverNode createConvolver ();
DelayNode createDelay (optional double maxDelayTime = 1.0);
DynamicsCompressorNode createDynamicsCompressor ();
GainNode createGain ();
IIRFilterNode createIIRFilter (sequence<double> feedforward,
sequence<double> feedback);
OscillatorNode createOscillator ();
PannerNode createPanner ();
PeriodicWave createPeriodicWave (sequence<float> real,
sequence<float> imag,
optional PeriodicWaveConstraints constraints = {});
ScriptProcessorNode createScriptProcessor(
optional unsigned long bufferSize = 0,
optional unsigned long numberOfInputChannels = 2,
optional unsigned long numberOfOutputChannels = 2);
StereoPannerNode createStereoPanner ();
WaveShaperNode createWaveShaper ();
Promise<AudioBuffer> decodeAudioData (
ArrayBuffer audioData,
optional DecodeSuccessCallback? successCallback,
optional DecodeErrorCallback? errorCallback);
};
</xmp>
<h4 id='BaseAudioContext-attributes'>
Attributes</h4>
<dl dfn-type=attribute dfn-for=BaseAudioContext>
: <dfn>audioWorklet</dfn>
::
Allows access to the <code>Worklet</code> object that can import
a script containing {{AudioWorkletProcessor}}
class definitions via the algorithms defined by [[!HTML]]
and {{AudioWorklet}}.
: <dfn>currentTime</dfn>
::
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 {{BaseAudioContext/currentTime}} has a value of
zero.
In the time coordinate system of {{BaseAudioContext/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 {{BaseAudioContext/currentTime}}.
When the {{BaseAudioContext}} is in the
"{{AudioContextState/running}}" state, the
value of this attribute is monotonically increasing and is
updated by the rendering thread in uniform increments,
corresponding to one <a>render quantum</a>. Thus, for a running
context, <code>currentTime</code> 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.
<code>currentTime</code> MUST be read <a>atomically</a> on the control thread before being
returned.
: <dfn>destination</dfn>
::
An {{AudioDestinationNode}}
with a single input representing the final destination for all
audio. Usually this will represent the actual audio hardware.
All {{AudioNode}}s actively rendering audio
will directly or indirectly connect to {{BaseAudioContext/destination}}.
: <dfn>listener</dfn>
::
An {{AudioListener}}
which is used for 3D <a href="#Spatialization">spatialization</a>.
: <dfn>onstatechange</dfn>
::
A property used to set an [=event handler=] 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). The event type of this event handler is
<dfn event>statechange</dfn>. An event that uses the
{{Event}} interface will be dispatched to the event
handler, which can query the AudioContext's state directly. A
newly-created AudioContext will always begin in the
<code>suspended</code> state, and a state change event will be
fired whenever the state changes to a different state. This
event is fired before the {{complete}} event
is fired.
: <dfn>sampleRate</dfn>
::
The sample rate (in sample-frames per second) at which the
{{BaseAudioContext}} handles audio. It is assumed that all
{{AudioNode}}s 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 <dfn dfn for="/">Nyquist frequency</dfn> is half this sample-rate value.
: <dfn>state</dfn>
::
Describes the current state of the {{BaseAudioContext}}. Getting this
attribute returns the contents of the {{[[control thread state]]}} slot.
: <dfn>renderQuantumSize</dfn>
::
Getting this attribute returns the value of {{BaseAudioContext/[[render
quantum size]]}} slot.
</dl>
<h4 id="BaseAudioContent-methods">
Methods</h4>
<dl dfn-type=method dfn-for=BaseAudioContext>
: <dfn>createAnalyser()</dfn>
::
<a>Factory method</a> for an {{AnalyserNode}}.
<div>
<em>No parameters.</em>
</div>
<div>
<em>Return type:</em> {{AnalyserNode}}
</div>
: <dfn>createBiquadFilter()</dfn>
::
<a>Factory method</a> for a {{BiquadFilterNode}}
representing a second order filter which can be configured as one
of several common filter types.
<div>
<em>No parameters.</em>
</div>
<div>
<em>Return type:</em> {{BiquadFilterNode}}
</div>
: <dfn>createBuffer(numberOfChannels, length, sampleRate)</dfn>
::
Creates an AudioBuffer of the given size. The audio data in the
buffer will be zero-initialized (silent).
<span class="synchronous">A {{NotSupportedError}} exception MUST be
thrown if any of the arguments is negative, zero, or outside its
nominal range.</span>
<pre class=argumentdef for="BaseAudioContext/createBuffer()">
numberOfChannels: Determines how many channels the buffer will have. An implementation MUST support at least 32 channels.
length: Determines the size of the buffer in sample-frames. This MUST be at least 1.
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.
</pre>
<div>
<em>Return type:</em> {{AudioBuffer}}
</div>
: <dfn>createBufferSource()</dfn>
::
<a>Factory method</a> for a
{{AudioBufferSourceNode}}.
<div>
<em>No parameters.</em>
</div>
<div>
<em>Return type:</em> {{AudioBufferSourceNode}}
</div>
: <dfn>createChannelMerger(numberOfInputs)</dfn>
::
<a>Factory method</a> for a
{{ChannelMergerNode}} representing a channel
merger. <span class="synchronous">An