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|
<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"
"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8" />
<title>Web Audio API</title>
<meta name="revision"
content="$Id: Overview.html,v 1.4 2012/07/30 11:44:57 tmichel Exp $" />
<link rel="stylesheet" href="style.css" type="text/css" />
<!--
<script src="section-links.js" type="application/ecmascript"></script>
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<style type='text/css'>
.ignore {
-ms-filter:"progid:DXImageTransform.Microsoft.Alpha(Opacity=50)";
filter: alpha(opacity=50);
}
</style>
<![endif]-->
<link rel="stylesheet" href="//www.w3.org/StyleSheets/TR/W3C-ED"
type="text/css" />
</head>
<body>
<div class="head">
<p><a href="http://www.w3.org/"><img width="72" height="48" alt="W3C"
src="http://www.w3.org/Icons/w3c_home" /></a> </p>
<h1 id="title" class="title">Web Audio API </h1>
<h2 id="w3c-date-document"><acronym
title="World Wide Web Consortium">W3C</acronym> Editor's Draft
</h2>
<dl>
<dt>This version: </dt>
<dd><a
href="https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html">https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html</a>
</dd>
<dt>Latest published version: </dt>
<dd><a
href="http://www.w3.org/TR/webaudio/">http://www.w3.org/TR/webaudio/</a>
</dd>
<dt>Previous version: </dt>
<dd><a
href="http://www.w3.org/TR/2012/WD-webaudio-20120315/">http://www.w3.org/TR/2012/WD-webaudio-20120315/</a>
</dd>
</dl>
<dl>
<dt>Editor: </dt>
<dd>Chris Rogers, Google <crogers@google.com></dd>
</dl>
<p class="copyright"><a
href="http://www.w3.org/Consortium/Legal/ipr-notice#Copyright">Copyright</a> ©
2012 <a href="http://www.w3.org/"><acronym
title="World Wide Web Consortium">W3C</acronym></a><sup>®</sup> (<a
href="http://www.csail.mit.edu/"><acronym
title="Massachusetts Institute of Technology">MIT</acronym></a>, <a
href="http://www.ercim.eu/"><acronym
title="European Research Consortium for Informatics and Mathematics">ERCIM</acronym></a>,
<a href="http://www.keio.ac.jp/">Keio</a>), All Rights Reserved. W3C <a
href="http://www.w3.org/Consortium/Legal/ipr-notice#Legal_Disclaimer">liability</a>,
<a
href="http://www.w3.org/Consortium/Legal/ipr-notice#W3C_Trademarks">trademark</a>
and <a href="http://www.w3.org/Consortium/Legal/copyright-documents">document
use</a> rules apply.</p>
<hr />
</div>
<div id="abstract-section" class="section">
<h2 id="abstract">Abstract</h2>
<p>This specification describes a high-level JavaScript <acronym
title="Application Programming Interface">API</acronym> for processing and
synthesizing audio in web applications. The primary paradigm is of an audio
routing graph, where a number of <a
href="#AudioNode-section"><code>AudioNode</code></a> 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 <a href="#JavaScriptProcessing-section">direct
JavaScript processing and synthesis</a> is also supported. </p>
<p>The <a href="#introduction">introductory</a> section covers the motivation
behind this specification.</p>
<p>This API is designed to be used in conjunction with other APIs and elements
on the web platform, notably: XMLHttpRequest
(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 and WebGL 3D graphics APIs. </p>
</div>
<div id="sotd-section" class="section">
<h2 id="sotd">Status of this Document</h2>
<p><em>This section describes the status of this document at the time of its
publication. Other documents may supersede this document. A list of current W3C
publications and the latest revision of this technical report can be found in
the <a href="http://www.w3.org/TR/">W3C technical reports index</a> at
http://www.w3.org/TR/. </em></p>
<p>This is the Editor's Draft of the <cite>Web Audio API</cite>
specification. It has been produced by the <a
href="http://www.w3.org/2011/audio/"><b>W3C Audio Working Group</b></a> , which
is part of the W3C WebApps Activity.</p>
<p></p>
<p>Please send comments about this document to <<a
href="mailto:public-audio@w3.org">public-audio@w3.org</a>> (<a
href="http://lists.w3.org/Archives/Public/public-audio/">public archives</a> of
the W3C audio mailing list). Web content and browser developers are encouraged
to review this draft. </p>
<p>Publication as a Working Draft does not imply endorsement by the W3C
Membership. This is a draft document and may be updated, replaced or obsoleted
by other documents at any time. It is inappropriate to cite this document as
other than work in progress.</p>
<p> This document was produced by a group operating under the <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/">5 February 2004 W3C Patent Policy</a>. W3C maintains a <a rel="disclosure" href="http://www.w3.org/2004/01/pp-impl/46884/status">public list of any patent disclosures</a> made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes contains <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/#def-essential">Essential Claim(s)</a> must disclose the information in accordance with <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/#sec-Disclosure">section 6 of the W3C Patent Policy</a>. </p>
</div>
<div id="toc">
<h2 id="L13522">Table of Contents</h2>
<div class="toc">
<ul>
<li><a href="#introduction">1. Introduction</a>
<ul>
<li><a href="#Features">1.1. Features</a></li>
<li><a href="#ModularRouting">1.2. Modular Routing</a></li>
<li><a href="#APIOverview">1.3. API Overview</a></li>
</ul>
</li>
<li><a href="#conformance">2. Conformance</a></li>
<li><a href="#API-section">4. The Audio API</a>
<ul>
<li><a href="#AudioContext-section">4.1. The AudioContext Interface</a>
<ul>
<li><a href="#attributes-AudioContext">4.1.1. Attributes</a></li>
<li><a href="#methodsandparams-AudioContext">4.1.2. Methods and
Parameters</a></li>
<li><a href="#lifetime-AudioContext">4.1.3. Lifetime</a></li>
</ul>
</li>
<li><a href="#OfflineAudioContext-section">4.1b. The OfflineAudioContext Interface</a>
</li>
<li><a href="#AudioNode-section">4.2. The AudioNode Interface</a>
<ul>
<li><a href="#attributes-AudioNode">4.2.1. Attributes</a></li>
<li><a href="#methodsandparams-AudioNode">4.2.2. Methods and
Parameters</a></li>
<li><a href="#lifetime-AudioNode">4.2.3. Lifetime</a></li>
</ul>
</li>
<li><a href="#AudioDestinationNode">4.4. The AudioDestinationNode
Interface</a>
<ul>
<li><a href="#attributes-AudioDestinationNode">4.4.1. Attributes</a></li>
</ul>
</li>
<li><a href="#AudioParam">4.5. The AudioParam Interface</a>
<ul>
<li><a href="#attributes-AudioParam">4.5.1. Attributes</a></li>
<li><a href="#methodsandparams-AudioParam">4.5.2. Methods and
Parameters</a></li>
<li><a href="#computedValue-AudioParam-section">4.5.3. Computation of Value</a></li>
<li><a href="#example1-AudioParam-section">4.5.4. AudioParam Automation Example</a></li>
</ul>
</li>
<li><a href="#GainNode">4.7. The GainNode Interface</a>
<ul>
<li><a href="#attributes-GainNode">4.7.1. Attributes</a></li>
</ul>
</li>
<li><a href="#DelayNode">4.8. The DelayNode Interface</a>
<ul>
<li><a href="#attributes-GainNode_2">4.8.1. Attributes</a></li>
</ul>
</li>
<li><a href="#AudioBuffer">4.9. The AudioBuffer Interface</a>
<ul>
<li><a href="#attributes-AudioBuffer">4.9.1. Attributes</a></li>
<li><a href="#methodsandparams-AudioBuffer">4.9.2. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#AudioBufferSourceNode">4.10. The AudioBufferSourceNode
Interface</a>
<ul>
<li><a href="#attributes-AudioBufferSourceNode">4.10.1.
Attributes</a></li>
<li><a href="#methodsandparams-AudioBufferSourceNode">4.10.2. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#MediaElementAudioSourceNode">4.11. The
MediaElementAudioSourceNode Interface</a></li>
<li><a href="#ScriptProcessorNode">4.12. The ScriptProcessorNode
Interface</a>
<ul>
<li><a href="#attributes-ScriptProcessorNode">4.12.1. Attributes</a></li>
</ul>
</li>
<li><a href="#AudioProcessingEvent">4.13. The AudioProcessingEvent
Interface</a>
<ul>
<li><a href="#attributes-AudioProcessingEvent">4.13.1. Attributes</a></li>
</ul>
</li>
<li><a href="#PannerNode">4.14. The PannerNode Interface</a>
<ul>
<li><a href="#attributes-PannerNode_attributes">4.14.2.
Attributes</a></li>
<li><a href="#Methods_and_Parameters">4.14.3. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#AudioListener">4.15. The AudioListener Interface</a>
<ul>
<li><a href="#attributes-AudioListener">4.15.1. Attributes</a></li>
<li><a href="#L15842">4.15.2. Methods and Parameters</a></li>
</ul>
</li>
<li><a href="#ConvolverNode">4.16. The ConvolverNode Interface</a>
<ul>
<li><a href="#attributes-ConvolverNode">4.16.1. Attributes</a></li>
</ul>
</li>
<li><a href="#AnalyserNode">4.17. The AnalyserNode
Interface</a>
<ul>
<li><a href="#attributes-ConvolverNode_2">4.17.1. Attributes</a></li>
<li><a href="#methods-and-parameters">4.17.2. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#ChannelSplitterNode">4.18. The ChannelSplitterNode
Interface</a>
<ul>
<li><a href="#example-1">Example:</a></li>
</ul>
</li>
<li><a href="#ChannelMergerNode">4.19. The ChannelMergerNode Interface</a>
<ul>
<li><a href="#example-2">Example:</a></li>
</ul>
</li>
<li><a href="#DynamicsCompressorNode">4.20. The DynamicsCompressorNode
Interface</a>
<ul>
<li><a href="#attributes-DynamicsCompressorNode">4.20.1.
Attributes</a></li>
</ul>
</li>
<li><a href="#BiquadFilterNode">4.21. The BiquadFilterNode Interface</a>
<ul>
<li><a href="#BiquadFilterNode-description">4.21.1 Lowpass</a></li>
<li><a href="#HIGHPASS">4.21.2 Highpass</a></li>
<li><a href="#BANDPASS">4.21.3 Bandpass</a></li>
<li><a href="#LOWSHELF">4.21.4 Lowshelf</a></li>
<li><a href="#L16352">4.21.5 Highshelf</a></li>
<li><a href="#PEAKING">4.21.6 Peaking</a></li>
<li><a href="#NOTCH">4.21.7 Notch</a></li>
<li><a href="#ALLPASS">4.21.8 Allpass</a></li>
<li><a href="#Methods">4.21.9. Methods</a></li>
</ul>
</li>
<li><a href="#WaveShaperNode">4.22. The WaveShaperNode Interface</a>
<ul>
<li><a href="#attributes-WaveShaperNode">4.22.1.
Attributes</a></li>
</ul>
</li>
<li><a href="#OscillatorNode">4.23. The OscillatorNode Interface</a>
<ul>
<li><a href="#attributes-OscillatorNode">4.23.1.
Attributes</a></li>
<li><a href="#methodsandparams-OscillatorNode-section">4.23.2. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#PeriodicWave">4.24. The PeriodicWave Interface</a>
</li>
<li><a href="#MediaStreamAudioSourceNode">4.25. The
MediaStreamAudioSourceNode Interface</a></li>
<li><a href="#MediaStreamAudioDestinationNode">4.26. The
MediaStreamAudioDestinationNode Interface</a></li>
</ul>
</li>
<li><a href="#MixerGainStructure">6. Mixer Gain Structure</a>
<ul>
<li><a href="#background">Background</a></li>
<li><a href="#SummingJunction">Summing Inputs</a></li>
<li><a href="#gain-Control">Gain Control</a></li>
<li><a href="#Example-mixer-with-send-busses">Example: Mixer with Send
Busses</a></li>
</ul>
</li>
<li><a href="#DynamicLifetime">7. Dynamic Lifetime</a>
<ul>
<li><a href="#DynamicLifetime-background">Background</a></li>
<li><a href="#Example-DynamicLifetime">Example</a></li>
</ul>
</li>
<li><a href="#UpMix">9. Channel up-mixing and down-mixing</a>
<ul>
<li><a href="#ChannelLayouts">9.1. Speaker Channel Layouts</a>
<ul>
<li><a href="#ChannelOrdering">9.1.1. Channel Ordering</a></li>
<li><a href="#UpMix-sub">9.1.2. Up Mixing</a></li>
<li><a href="#down-mix">9.1.3. Down Mixing</a></li>
</ul>
</li>
<li><a href="#ChannelRules-section">9.2. Channel Rules Examples</a>
</ul>
</li>
<li><a href="#Spatialization">11. Spatialization / Panning </a>
<ul>
<li><a href="#Spatialization-background">Background</a></li>
<li><a href="#Spatialization-panning-algorithm">Panning Algorithm</a></li>
<li><a href="#Spatialization-distance-effects">Distance Effects</a></li>
<li><a href="#Spatialization-sound-cones">Sound Cones</a></li>
<li><a href="#Spatialization-doppler-shift">Doppler Shift</a></li>
</ul>
</li>
<li><a href="#Convolution">12. Linear Effects using Convolution</a>
<ul>
<li><a href="#Convolution-background">Background</a></li>
<li><a href="#Convolution-motivation">Motivation for use as a
Standard</a></li>
<li><a href="#Convolution-implementation-guide">Implementation Guide</a></li>
<li><a href="#Convolution-reverb-effect">Reverb Effect (with
matrixing)</a></li>
<li><a href="#recording-impulse-responses">Recording Impulse
Responses</a></li>
<li><a href="#tools">Tools</a></li>
<li><a href="#recording-setup">Recording Setup</a></li>
<li><a href="#warehouse">The Warehouse Space</a></li>
</ul>
</li>
<li><a href="#JavaScriptProcessing">13. JavaScript Synthesis and
Processing</a>
<ul>
<li><a href="#custom-DSP-effects">Custom DSP Effects</a></li>
<li><a href="#educational-applications">Educational Applications</a></li>
<li><a href="#javaScript-performance">JavaScript Performance</a></li>
</ul>
</li>
<li><a href="#Performance">15. Performance Considerations</a>
<ul>
<li><a href="#Latency">15.1. Latency: What it is and Why it's
Important</a></li>
<li><a href="#audio-glitching">15.2. Audio Glitching</a></li>
<li><a href="#hardware-scalability">15.3. Hardware Scalability</a>
<ul>
<li><a href="#CPU-monitoring">15.3.1. CPU monitoring</a></li>
<li><a href="#Voice-dropping">15.3.2. Voice Dropping</a></li>
<li><a href="#Simplification-of-Effects-Processing">15.3.3.
Simplification of Effects Processing</a></li>
<li><a href="#Sample-rate">15.3.4. Sample Rate</a></li>
<li><a href="#pre-flighting">15.3.5. Pre-flighting</a></li>
<li><a href="#Authoring-for-different-user-agents">15.3.6. Authoring
for different user agents</a></li>
<li><a href="#Scalability-of-Direct-JavaScript-Synthesis">15.3.7.
Scalability of Direct JavaScript Synthesis / Processing</a></li>
</ul>
</li>
<li><a href="#JavaScriptPerformance">15.4. JavaScript Issues with
real-time Processing and Synthesis: </a></li>
</ul>
</li>
<li><a href="#ExampleApplications">16. Example Applications</a>
<ul>
<li><a href="#basic-sound-playback">Basic Sound Playback</a></li>
<li><a href="#threeD-environmentse-and-games">3D Environments and
Games</a></li>
<li><a href="#musical-applications">Musical Applications</a></li>
<li><a href="#music-visualizers">Music Visualizers</a></li>
<li><a href="#educational-applications_2">Educational
Applications</a></li>
<li><a href="#artistic-audio-exploration">Artistic Audio
Exploration</a></li>
</ul>
</li>
<li><a href="#SecurityConsiderations">17. Security Considerations</a></li>
<li><a href="#PrivacyConsiderations">18. Privacy Considerations</a></li>
<li><a href="#requirements">19. Requirements and Use Cases</a></li>
<li><a href="#OldNames">20. Old Names</a></li>
<li><a href="#L17310">A.References</a>
<ul>
<li><a href="#Normative-references">A.1 Normative references</a></li>
<li><a href="#Informative-references">A.2 Informative references</a></li>
</ul>
</li>
<li><a href="#L17335">B.Acknowledgements</a></li>
<li><a href="#ChangeLog">C. Web Audio API Change Log</a></li>
</ul>
</div>
</div>
<div id="sections">
<div id="div-introduction" class="section">
<h2 id="introduction">1. Introduction</h2>
<p class="norm">This section is informative.</p>
<p>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 <code>audio</code> 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. </p>
<p>The APIs have been designed with a wide variety of <a
href="#ExampleApplications-section">use cases</a> 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 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. </p>
<div id="Features-section" class="section">
<h2 id="Features">1.1. Features</h2>
</div>
<p>The API supports these primary features: </p>
<ul>
<li><a href="#ModularRouting-section">Modular routing</a> for simple or
complex mixing/effect architectures, including <a
href="#MixerGainStructure-section">multiple sends and submixes</a>.</li>
<li><a href="#AudioParam">Sample-accurate scheduled sound
playback</a> with low <a href="#Latency-section">latency</a> for musical
applications requiring a very high degree of rhythmic precision such as
drum machines and sequencers. This also includes the possibility of <a
href="#DynamicLifetime-section">dynamic creation</a> of effects. </li>
<li>Automation of audio parameters for envelopes, fade-ins / fade-outs,
granular effects, filter sweeps, LFOs etc. </li>
<li>Flexible handling of channels in an audio stream, allowing them to be split and merged.</li>
<li>Processing of audio sources from an <code>audio</code> or
<code>video</code> <a href="#MediaElementAudioSourceNode">media
element</a>. </li>
<li>Processing live audio input using a <a href="#MediaStreamAudioSourceNode">MediaStream</a>
from getUserMedia().
</li>
<li>Integration with WebRTC
<ul>
<li>Processing audio received from a remote peer using a <a href="#MediaStreamAudioSourceNode">MediaStream</a>.
</li>
<li>Sending a generated or processed audio stream to a remote peer using a <a href="#MediaStreamAudioDestinationNode">MediaStream</a>.
</li>
</ul>
</li>
<li>Audio stream synthesis and processing <a
href="#JavaScriptProcessing-section">directly in JavaScript</a>. </li>
<li><a href="#Spatialization-section">Spatialized audio</a> supporting a wide
range of 3D games and immersive environments:
<ul>
<li>Panning models: equal-power, HRTF, pass-through </li>
<li>Distance Attenuation </li>
<li>Sound Cones </li>
<li>Obstruction / Occlusion </li>
<li>Doppler Shift </li>
<li>Source / Listener based</li>
</ul>
</li>
<li>A <a href="#Convolution-section">convolution engine</a> for a wide range
of linear effects, especially very high-quality room effects. Here are some
examples of possible effects:
<ul>
<li>Small / large room </li>
<li>Cathedral </li>
<li>Concert hall </li>
<li>Cave </li>
<li>Tunnel </li>
<li>Hallway </li>
<li>Forest </li>
<li>Amphitheater </li>
<li>Sound of a distant room through a doorway </li>
<li>Extreme filters</li>
<li>Strange backwards effects</li>
<li>Extreme comb filter effects </li>
</ul>
</li>
<li>Dynamics compression for overall control and sweetening of the mix </li>
<li>Efficient <a href="#AnalyserNode">real-time time-domain and
frequency analysis / music visualizer support</a></li>
<li>Efficient biquad filters for lowpass, highpass, and other common filters.
</li>
<li>A Waveshaping effect for distortion and other non-linear effects</li>
<li>Oscillators</li>
</ul>
<div id="ModularRouting-section">
<h2 id="ModularRouting">1.2. Modular Routing</h2>
<p>Modular routing allows arbitrary connections between different <a
href="#AudioNode-section"><code>AudioNode</code></a> 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, the most common example being <a
href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a> the final destination to the audio
hardware. 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="#UpMix-section">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 <a
href="#UpMix-section">appropriately</a>. </p>
<p>In the simplest case, a single source can be routed directly to the output.
All routing occurs within an <a
href="#AudioContext-section"><code>AudioContext</code></a> containing a single
<a href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a>:
</p>
<img alt="modular routing" src="images/modular-routing1.png" />
<p>Illustrating this simple routing, here's a simple example playing a single
sound: </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span> </div>
<div class="blockContent">
<pre class="code"><code class="es-code">
var context = new AudioContext();
function playSound() {
var source = context.createBufferSource();
source.buffer = dogBarkingBuffer;
source.connect(context.destination);
source.start(0);
}
</code></pre>
</div>
</div>
<p>Here's a more complex example with three sources and a convolution reverb
send with a dynamics compressor at the final output stage: </p>
<img alt="modular routing2" src="images/modular-routing2.png" />
<div class="example">
<div class="exampleHeader">
Example</div>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span></div>
<div class="blockContent">
<pre class="code"><code class="es-code">
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);
}
</code></pre>
</div>
</div>
</div>
</div>
</div>
<div id="APIOverview-section" class="section">
<h2 id="APIOverview">1.3. API Overview</h2>
</div>
<p>The interfaces defined are: </p>
<ul>
<li>An <a class="dfnref" href="#AudioContext-section">AudioContext</a>
interface, which contains an audio signal graph representing connections
betweens AudioNodes. </li>
<li>An <a class="dfnref" href="#AudioNode-section">AudioNode</a> interface,
which represents audio sources, audio outputs, and intermediate processing
modules. AudioNodes can be dynamically connected together in a <a
href="#ModularRouting-section">modular fashion</a>. <code>AudioNodes</code>
exist in the context of an <code>AudioContext</code> </li>
<li>An <a class="dfnref"
href="#AudioDestinationNode-section">AudioDestinationNode</a> interface, an
AudioNode subclass representing the final destination for all rendered
audio. </li>
<li>An <a class="dfnref" href="#AudioBuffer-section">AudioBuffer</a>
interface, for working with memory-resident audio assets. These can
represent one-shot sounds, or longer audio clips. </li>
<li>An <a class="dfnref"
href="#AudioBufferSourceNode-section">AudioBufferSourceNode</a> interface,
an AudioNode which generates audio from an AudioBuffer. </li>
<li>A <a class="dfnref"
href="#MediaElementAudioSourceNode-section">MediaElementAudioSourceNode</a>
interface, an AudioNode which is the audio source from an
<code>audio</code>, <code>video</code>, or other media element. </li>
<li>A <a class="dfnref"
href="#MediaStreamAudioSourceNode-section">MediaStreamAudioSourceNode</a>
interface, an AudioNode which is the audio source from a
MediaStream such as live audio input, or from a remote peer. </li>
<li>A <a class="dfnref"
href="#MediaStreamAudioDestinationNode-section">MediaStreamAudioDestinationNode</a>
interface, an AudioNode which is the audio destination to a
MediaStream sent to a remote peer. </li>
<li>A <a class="dfnref"
href="#ScriptProcessorNode-section">ScriptProcessorNode</a> interface, an
AudioNode for generating or processing audio directly in JavaScript. </li>
<li>An <a class="dfnref"
href="#AudioProcessingEvent-section">AudioProcessingEvent</a> interface,
which is an event type used with <code>ScriptProcessorNode</code> objects.
</li>
<li>An <a class="dfnref" href="#AudioParam-section">AudioParam</a> interface,
for controlling an individual aspect of an AudioNode's functioning, such as
volume. </li>
<li>An <a class="dfnref" href="#GainNode-section">GainNode</a>
interface, for explicit gain control. Because inputs to AudioNodes support
multiple connections (as a unity-gain summing junction), mixers can be <a
href="#MixerGainStructure-section">easily built</a> with GainNodes.
</li>
<li>A <a class="dfnref" href="#BiquadFilterNode-section">BiquadFilterNode</a>
interface, an AudioNode for common low-order filters such as:
<ul>
<li>Low Pass</li>
<li>High Pass </li>
<li>Band Pass </li>
<li>Low Shelf </li>
<li>High Shelf </li>
<li>Peaking </li>
<li>Notch </li>
<li>Allpass </li>
</ul>
</li>
<li>A <a class="dfnref" href="#DelayNode-section">DelayNode</a> interface, an
AudioNode which applies a dynamically adjustable variable delay. </li>
<li>An <a class="dfnref" href="#PannerNode-section">PannerNode</a>
interface, for spatializing / positioning audio in 3D space. </li>
<li>An <a class="dfnref" href="#AudioListener-section">AudioListener</a>
interface, which works with an <code>PannerNode</code> for
spatialization. </li>
<li>A <a class="dfnref" href="#ConvolverNode-section">ConvolverNode</a>
interface, an AudioNode for applying a <a
href="#Convolution-section">real-time linear effect</a> (such as the sound
of a concert hall). </li>
<li>A <a class="dfnref"
href="#AnalyserNode-section">AnalyserNode</a> interface,
for use with music visualizers, or other visualization applications. </li>
<li>A <a class="dfnref"
href="#ChannelSplitterNode-section">ChannelSplitterNode</a> interface,
for accessing the individual channels of an audio stream in the routing
graph. </li>
<li>A <a class="dfnref"
href="#ChannelMergerNode-section">ChannelMergerNode</a> interface, for
combining channels from multiple audio streams into a single audio stream.
</li>
<li>A <a
href="#DynamicsCompressorNode-section">DynamicsCompressorNode</a> interface, an
AudioNode for dynamics compression. </li>
<li>A <a class="dfnref" href="#dfn-WaveShaperNode">WaveShaperNode</a>
interface, an AudioNode which applies a non-linear waveshaping effect for
distortion and other more subtle warming effects. </li>
<li>A <a class="dfnref" href="#dfn-OscillatorNode">OscillatorNode</a>
interface, an audio source generating a periodic waveform. </li>
</ul>
</div>
<div id="conformance-section" class="section">
<h2 id="conformance">2. Conformance</h2>
<p>Everything in this specification is normative except for examples and
sections marked as being informative. </p>
<p>The keywords “<span class="rfc2119">MUST</span>”, “<span
class="rfc2119">MUST NOT</span>”, “<span
class="rfc2119">REQUIRED</span>”, “<span class="rfc2119">SHALL</span>”,
“<span class="rfc2119">SHALL NOT</span>”, “<span
class="rfc2119">RECOMMENDED</span>”, “<span class="rfc2119">MAY</span>”
and “<span class="rfc2119">OPTIONAL</span>” in this document are to be
interpreted as described in <cite><a href="http://www.ietf.org/rfc/rfc2119">Key
words for use in RFCs to Indicate Requirement Levels</a></cite> <a
href="#RFC2119">[RFC2119]</a>. </p>
<p>The following conformance classes are defined by this specification: </p>
<dl>
<dt><dfn id="dfn-conforming-implementation">conforming
implementation</dfn></dt>
<dd><p>A user agent is considered to be a <a class="dfnref"
href="#dfn-conforming-implementation">conforming implementation</a> if it
satisfies all of the <span class="rfc2119">MUST</span>-, <span
class="rfc2119">REQUIRED</span>- and <span
class="rfc2119">SHALL</span>-level criteria in this specification that
apply to implementations. </p>
</dd>
</dl>
</div>
<div id="terminology-section" class="section">
<div id="API-section-section" class="section">
<h2 id="API-section">4. The Audio API</h2>
</div>
<div id="AudioContext-section-section" class="section">
<h2 id="AudioContext-section">4.1. The AudioContext Interface</h2>
<p>This interface represents a set of <a
href="#AudioNode-section"><code>AudioNode</code></a> objects and their
connections. It allows for arbitrary routing of signals to the <a
href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a>
(what the user ultimately hears). Nodes are created from the context and are
then <a href="#ModularRouting-section">connected</a> together. In most use
cases, only a single AudioContext is used per document.</p>
<br>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-context-idl">
callback DecodeSuccessCallback = void (AudioBuffer decodedData);
callback DecodeErrorCallback = void ();
[Constructor]
interface <dfn id="dfn-AudioContext">AudioContext</dfn> : EventTarget {
readonly attribute AudioDestinationNode destination;
readonly attribute float sampleRate;
readonly attribute double currentTime;
readonly attribute AudioListener listener;
AudioBuffer createBuffer(unsigned long numberOfChannels, unsigned long length, float sampleRate);
void decodeAudioData(ArrayBuffer audioData,
DecodeSuccessCallback successCallback,
optional DecodeErrorCallback errorCallback);
<span class="comment">// AudioNode creation </span>
AudioBufferSourceNode createBufferSource();
MediaElementAudioSourceNode createMediaElementSource(HTMLMediaElement mediaElement);
MediaStreamAudioSourceNode createMediaStreamSource(MediaStream mediaStream);
MediaStreamAudioDestinationNode createMediaStreamDestination();
ScriptProcessorNode createScriptProcessor(optional unsigned long bufferSize = 0,
optional unsigned long numberOfInputChannels = 2,
optional unsigned long numberOfOutputChannels = 2);
AnalyserNode createAnalyser();
GainNode createGain();
DelayNode createDelay(optional double maxDelayTime = 1.0);
BiquadFilterNode createBiquadFilter();
WaveShaperNode createWaveShaper();
PannerNode createPanner();
ConvolverNode createConvolver();
ChannelSplitterNode createChannelSplitter(optional unsigned long numberOfOutputs = 6);
ChannelMergerNode createChannelMerger(optional unsigned long numberOfInputs = 6);
DynamicsCompressorNode createDynamicsCompressor();
OscillatorNode createOscillator();
PeriodicWave createPeriodicWave(Float32Array real, Float32Array imag);
};
</code></pre>
</div>
</div>
<div id="attributes-AudioContext-section" class="section">
<h3 id="attributes-AudioContext">4.1.1. Attributes</h3>
<dl>
<dt id="dfn-destination"><code>destination</code></dt>
<dd><p>An <a
href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a>
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 <code>destination</code>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-sampleRate"><code>sampleRate</code></dt>
<dd><p>The sample rate (in sample-frames per second) at which the
AudioContext handles audio. It is assumed that all AudioNodes in the
context run at this rate. In making this assumption, sample-rate
converters or "varispeed" processors are not supported in real-time
processing.</p>
</dd>
</dl>
<dl>
<dt id="dfn-currentTime"><code>currentTime</code></dt>
<dd><p>This is a time in seconds which starts at zero when the context is
created and increases in real-time. All scheduled times are relative to
it. This is not a "transport" time which can be started, paused, and
re-positioned. It is always moving forward. A GarageBand-like timeline
transport system can be very easily built on top of this (in JavaScript).
This time corresponds to an ever-increasing hardware timestamp. </p>
</dd>
</dl>
<dl>
<dt id="dfn-listener"><code>listener</code></dt>
<dd><p>An <a href="#AudioListener-section"><code>AudioListener</code></a>
which is used for 3D <a
href="#Spatialization-section">spatialization</a>.</p>
</dd>
</dl>
</div>
<div id="methodsandparams-AudioContext-section" class="section">
<h3 id="methodsandparams-AudioContext">4.1.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-createBuffer">The <code>createBuffer</code> method</dt>
<dd><p>Creates an AudioBuffer of the given size. The audio data in the
buffer will be zero-initialized (silent). An NOT_SUPPORTED_ERR exception will be thrown if
the <code>numberOfChannels</code> or <code>sampleRate</code> are out-of-bounds,
or if length is 0.</p>
<p>The <dfn id="dfn-numberOfChannels">numberOfChannels</dfn> parameter
determines how many channels the buffer will have. An implementation must support at least 32 channels. </p>
<p>The <dfn id="dfn-length">length</dfn> parameter determines the size of
the buffer in sample-frames. </p>
<p>The <dfn id="dfn-sampleRate_2">sampleRate</dfn> parameter describes
the sample-rate of the linear PCM audio data in the buffer in
sample-frames per second. An implementation must support sample-rates in at least the range 22050 to 96000.</p>
</dd>
</dl>
<dl>
<dt id="dfn-decodeAudioData">The <code>decodeAudioData</code> method</dt>
<dd><p>Asynchronously decodes the audio file data contained in the
ArrayBuffer. The ArrayBuffer can, for example, be loaded from an XMLHttpRequest's
<code>response</code> attribute after setting the <code>responseType</code> to "arraybuffer".
Audio file data can be in any of the
formats supported by the <code>audio</code> element. </p>
<p><dfn id="dfn-audioData">audioData</dfn> is an ArrayBuffer containing
audio file data.</p>
<p><dfn id="dfn-successCallback">successCallback</dfn> is 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.</p>
<p><dfn id="dfn-errorCallback">errorCallback</dfn> is a callback function
which will be invoked if there is an error decoding the audio file
data.</p>
<p>
The following steps must be performed:
</p>
<ol>
<li>Temporarily neuter the <dfn>audioData</dfn> ArrayBuffer in such a way that JavaScript code may not
access or modify the data.</li>
<li>Queue a decoding operation to be performed on another thread.</li>
<li>The decoding thread will attempt to decode the encoded <dfn>audioData</dfn> into linear PCM.
If a decoding error is encountered due to the audio format not being recognized or supported, or
because of corrupted/unexpected/inconsistent data then the <dfn>audioData</dfn> neutered state
will be restored to normal and the <dfn>errorCallback</dfn> will be
scheduled to run on the main thread's event loop and these steps will be terminated.</li>
<li>The decoding thread will 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 <dfn>audioData</dfn>. The final result (after possibly sample-rate converting) will be stored
in an AudioBuffer.
</li>
<li>The <dfn>audioData</dfn> neutered state will be restored to normal
</li>
<li>
The <dfn>successCallback</dfn> function will be scheduled to run on the main thread's event loop
given the AudioBuffer from step (4) as an argument.
</li>
</ol>
</dd>
</dl>
<dl>
<dt id="dfn-createBufferSource">The <code>createBufferSource</code>
method</dt>
<dd><p>Creates an <a
href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createMediaElementSource">The <code>createMediaElementSource</code>
method</dt>
<dd><p>Creates a <a
href="#MediaElementAudioSourceNode-section"><code>MediaElementAudioSourceNode</code></a> 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.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createMediaStreamSource">The <code>createMediaStreamSource</code>
method</dt>
<dd><p>Creates a <a
href="#MediaStreamAudioSourceNode-section"><code>MediaStreamAudioSourceNode</code></a> given a MediaStream.
As a consequence of calling this method, audio playback from the MediaStream will be re-routed
into the processing graph of the AudioContext.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createMediaStreamDestination">The <code>createMediaStreamDestination</code>
method</dt>
<dd><p>Creates a <a
href="#MediaStreamAudioDestinationNode-section"><code>MediaStreamAudioDestinationNode</code></a>.
</p>
</dd>
</dl>
<dl>
<dt id="dfn-createScriptProcessor">The <code>createScriptProcessor</code>
method</dt>
<dd><p>Creates a <a
href="#ScriptProcessorNode"><code>ScriptProcessorNode</code></a> for
direct audio processing using JavaScript. An INDEX_SIZE_ERR exception MUST be thrown if <code>bufferSize</code> or <code>numberOfInputChannels</code> or <code>numberOfOutputChannels</code>
are outside the valid range. </p>
<p>The <dfn id="dfn-bufferSize">bufferSize</dfn> 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 <code>audioprocess</code> event is dispatched and
how many sample-frames need to be processed each call. Lower values for
<code>bufferSize</code> will result in a lower (better) <a
href="#Latency-section">latency</a>. Higher values will be necessary to
avoid audio breakup and <a href="#Glitching-section">glitches</a>.
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.
</p>
<p>The <dfn id="dfn-numberOfInputChannels">numberOfInputChannels</dfn> parameter (defaults to 2) and
determines the number of channels for this node's input. Values of up to 32 must be supported. </p>
<p>The <dfn id="dfn-numberOfOutputChannels">numberOfOutputChannels</dfn> parameter (defaults to 2) and
determines the number of channels for this node's output. Values of up to 32 must be supported.</p>
<p>It is invalid for both <code>numberOfInputChannels</code> and
<code>numberOfOutputChannels</code> to be zero. </p>
</dd>
</dl>
<dl>
<dt id="dfn-createAnalyser">The <code>createAnalyser</code> method</dt>
<dd><p>Creates a <a
href="#AnalyserNode-section"><code>AnalyserNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createGain">The <code>createGain</code> method</dt>
<dd><p>Creates a <a
href="#GainNode-section"><code>GainNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createDelay">The <code>createDelay</code> method</dt>
<dd><p>Creates a <a href="#DelayNode-section"><code>DelayNode</code></a>
representing a variable delay line. The initial default delay time will
be 0 seconds.</p>
<p>The <dfn id="dfn-maxDelayTime">maxDelayTime</dfn> 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 NOT_SUPPORTED_ERR exception will be thrown.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createBiquadFilter">The <code>createBiquadFilter</code>
method</dt>
<dd><p>Creates a <a
href="#BiquadFilterNode-section"><code>BiquadFilterNode</code></a>
representing a second order filter which can be configured as one of
several common filter types.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createWaveShaper">The <code>createWaveShaper</code>
method</dt>
<dd><p>Creates a <a
href="#WaveShaperNode-section"><code>WaveShaperNode</code></a>
representing a non-linear distortion.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createPanner">The <code>createPanner</code> method</dt>
<dd><p>Creates an <a
href="#PannerNode-section"><code>PannerNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createConvolver">The <code>createConvolver</code> method</dt>
<dd><p>Creates a <a
href="#ConvolverNode-section"><code>ConvolverNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createChannelSplitter">The <code>createChannelSplitter</code>
method</dt>
<dd><p>Creates an <a
href="#ChannelSplitterNode-section"><code>ChannelSplitterNode</code></a>
representing a channel splitter. An exception will be thrown for invalid parameter values.</p>
<p>The <dfn id="dfn-numberOfOutputs">numberOfOutputs</dfn> parameter
determines the number of outputs. Values of up to 32 must be supported. If not specified, then 6 will be used. </p>
</dd>
</dl>
<dl>
<dt id="dfn-createChannelMerger">The <code>createChannelMerger</code>
method</dt>
<dd><p>Creates an <a
href="#ChannelMergerNode-section"><code>ChannelMergerNode</code></a>
representing a channel merger. An exception will be thrown for invalid parameter values.</p>
<p>The <dfn id="dfn-numberOfInputs">numberOfInputs</dfn> parameter
determines the number of inputs. Values of up to 32 must be supported. If not specified, then 6 will be used. </p>
</dd>
</dl>
<dl>
<dt id="dfn-createDynamicsCompressor">The
<code>createDynamicsCompressor</code> method</dt>
<dd><p>Creates a <a
href="#DynamicsCompressorNode-section"><code>DynamicsCompressorNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createOscillator">The
<code>createOscillator</code> method</dt>
<dd><p>Creates an <a
href="#OscillatorNode-section"><code>OscillatorNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createPeriodicWave">The
<code>createPeriodicWave</code> method</dt>
<dd><p>Creates a <a
href="#PeriodicWave-section"><code>PeriodicWave</code></a> representing a waveform containing arbitrary harmonic content.
The <code>real</code> and <code>imag</code> parameters must be of type <code>Float32Array</code> of equal
lengths greater than zero and less than or equal to 4096 or an exception will be thrown.
These parameters specify the Fourier coefficients of a
<a href="http://en.wikipedia.org/wiki/Fourier_series">Fourier series</a> representing the partials of a periodic waveform.
The created PeriodicWave will be used with an <a href="#OscillatorNode-section"><code>OscillatorNode</code></a>
and will represent a <em>normalized</em> time-domain waveform having maximum absolute peak value of 1.
Another way of saying this is that the generated waveform of an <a href="#OscillatorNode-section"><code>OscillatorNode</code></a>
will have maximum peak value at 0dBFS. Conveniently, this corresponds to the full-range of the signal values used by the Web Audio API.
Because the PeriodicWave will be normalized on creation, the <code>real</code> and <code>imag</code> parameters
represent <em>relative</em> values.
</p>
<p>The <dfn id="dfn-real">real</dfn> parameter represents an array of <code>cosine</code> terms (traditionally the A terms).
In audio terminology, the first element (index 0) is the DC-offset of the periodic waveform and is usually set to zero.
The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.</p>
<p>The <dfn id="dfn-imag">imag</dfn> parameter represents an array of <code>sine</code> 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.</p>
</dd>
</dl>
</div>
</div>
<h3 id="lifetime-AudioContext">4.1.3. Lifetime</h3>
<p class="norm">This section is informative.</p>
<p>
Once created, an <code>AudioContext</code> will continue to play sound until it has no more sound to play, or
the page goes away.
</p>
<div id="OfflineAudioContext-section-section" class="section">
<h2 id="OfflineAudioContext-section">4.1b. The OfflineAudioContext Interface</h2>
<p>
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, calling a completion event handler
with the result provided as an AudioBuffer.
</p>
<p>
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="offline-audio-context-idl">
[Constructor(unsigned long numberOfChannels, unsigned long length, float sampleRate)]
interface <dfn id="dfn-OfflineAudioContext">OfflineAudioContext</dfn> : AudioContext {
void startRendering();
attribute EventHandler oncomplete;
};
</code></pre>
</div>
</div>
<div id="attributes-OfflineAudioContext-section" class="section">
<h3 id="attributes-OfflineAudioContext">4.1b.1. Attributes</h3>
<dl>
<dt id="dfn-oncomplete"><code>oncomplete</code></dt>
<dd><p>An EventHandler of type <a href="#OfflineAudioCompletionEvent-section">OfflineAudioCompletionEvent</a>.</p>
</dd>
</dl>
</div>
<div id="methodsandparams-OfflineAudioContext-section" class="section">
<h3 id="methodsandparams-OfflineAudioContext">4.1b.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-startRendering">The <code>startRendering</code>
method</dt>
<dd><p>Given the current connections and scheduled changes, starts rendering audio. The
<code>oncomplete</code> handler will be called once the rendering has finished.
This method must only be called one time or an exception will be thrown.</p>
</dd>
</dl>
</div>
<div id="OfflineAudioCompletionEvent-section" class="section">
<h2 id="OfflineAudioCompletionEvent">4.1c. The OfflineAudioCompletionEvent Interface</h2>
<p>This is an <code>Event</code> object which is dispatched to <a
href="#OfflineAudioContext-section"><code>OfflineAudioContext</code></a>. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="offline-audio-completion-event-idl">
interface <dfn id="dfn-OfflineAudioCompletionEvent">OfflineAudioCompletionEvent</dfn> : Event {
readonly attribute AudioBuffer renderedBuffer;
};
</code></pre>
</div>
</div>
<div id="attributes-OfflineAudioCompletionEvent-section" class="section">
<h3 id="attributes-OfflineAudioCompletionEvent">4.1c.1. Attributes</h3>
<dl>
<dt id="dfn-renderedBuffer"><code>renderedBuffer</code></dt>
<dd><p>An AudioBuffer containing the rendered audio data once an OfflineAudioContext has finished rendering.
It will have a number of channels equal to the <code>numberOfChannels</code> parameter
of the OfflineAudioContext constructor.</p>
</dd>
</dl>
</div>
</div>
<div id="AudioNode-section-section" class="section">
<h2 id="AudioNode-section">4.2. The AudioNode Interface</h2>
<p>AudioNodes are the building blocks of an <a
href="#AudioContext-section"><code>AudioContext</code></a>. This interface
represents audio sources, the audio destination, and intermediate processing
modules. These modules can be connected together to form <a
href="#ModularRouting-section">processing graphs</a> for rendering audio to the
audio hardware. Each node can have <dfn>inputs</dfn> and/or <dfn>outputs</dfn>.
A <dfn>source node</dfn> has no inputs
and a single output. An <a
href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a> has
one input and no outputs and represents the final destination to the audio
hardware. Most processing nodes such as filters will have one input and one
output. Each type of <code>AudioNode</code> differs in the details of how it processes or synthesizes audio. But, in general, <code>AudioNodes</code>
will process its inputs (if it has any), and generate audio for its outputs (if it has any).
</p>
<p>
Each <dfn>output</dfn> has one or more <dfn>channels</dfn>. The exact number of channels depends on the details of the specific AudioNode.
</p>
<p>
An output may connect to one or more <code>AudioNode</code> inputs, thus <em>fan-out</em> is supported. An input initially has no connections,
but may be connected from one
or more <code>AudioNode</code> outputs, thus <em>fan-in</em> is supported. When the <code>connect()</code> method is called to connect
an output of an AudioNode to an input of an AudioNode, we call that a <dfn>connection</dfn> to the input.
</p>
<p>
Each AudioNode <dfn>input</dfn> has a specific number of channels at any given time. This number can change depending on the <dfn>connection(s)</dfn>
made to the input. If the input has no connections then it has one channel which is silent.
</p>
<p>
For each <dfn>input</dfn>, an <code>AudioNode</code> performs a mixing (usually an up-mixing) of all connections to that input.
Please see <a href="#MixerGainStructure-section">Mixer Gain Structure</a> for more informative details, and the <a href="#UpMix-section">Channel up-mixing and down-mixing</a>
section for normative requirements.
</p>
<p>
For performance reasons, practical implementations will need to use block processing, with each <code>AudioNode</code> processing a
fixed number of sample-frames of size <em>block-size</em>. In order to get uniform behavior across implementations, we will define this
value explicitly. <em>block-size</em> is defined to be 128 sample-frames which corresponds to roughly 3ms at a sample-rate of 44.1KHz.
</p>
<p>
AudioNodes are <em>EventTarget</em>s, as described in <cite><a href="http://dom.spec.whatwg.org/">DOM</a></cite>
<a href="#DOM">[DOM]</a>. This means that it is possible to dispatch events to AudioNodes the same
way that other EventTargets accept events.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-node-idl">
enum <dfn>ChannelCountMode</dfn> {
"max",
"clamped-max",
"explicit"
};
enum <dfn>ChannelInterpretation</dfn> {
"speakers",
"discrete"
};
interface <dfn id="dfn-AudioNode">AudioNode</dfn> : EventTarget {
void connect(AudioNode destination, optional unsigned long output = 0, optional unsigned long input = 0);
void connect(AudioParam destination, optional unsigned long output = 0);
void disconnect(optional unsigned long output = 0);
readonly attribute AudioContext context;
readonly attribute unsigned long numberOfInputs;
readonly attribute unsigned long numberOfOutputs;
// Channel up-mixing and down-mixing rules for all inputs.
attribute unsigned long channelCount;
attribute ChannelCountMode channelCountMode;
attribute ChannelInterpretation channelInterpretation;
};
</code></pre>
</div>
</div>
<div id="attributes-AudioNode-section" class="section">
<h3 id="attributes-AudioNode">4.2.1. Attributes</h3>
<dl>
<dt id="dfn-context"><code>context</code></dt>
<dd><p>The AudioContext which owns this AudioNode.</p>
</dd>
</dl>
<dl>
<dt id="dfn-numberOfInputs_2"><code>numberOfInputs</code></dt>
<dd><p>The number of inputs feeding into the AudioNode. For <dfn>source nodes</dfn>,
this will be 0.</p>
</dd>
</dl>
<dl>
<dt id="dfn-numberOfOutputs_2"><code>numberOfOutputs</code></dt>
<dd><p>The number of outputs coming out of the AudioNode. This will be 0
for an AudioDestinationNode.</p>
</dd>
</dl>
<dl>
<dt id="dfn-channelCount"><code>channelCount</code><dt>
<dd><p>The number of channels used when up-mixing and down-mixing connections to any inputs to the node. The default value is 2
except for specific nodes where its value is specially determined.
This attribute has no effect for nodes with no inputs.
If this value is set to zero, the implementation MUST raise the
NOT_SUPPORTED_ERR exception.</p>
<p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a>
section for more information on this attribute.</p>
</dd>
</dl>
<dl>
<dt id="dfn-channelCountMode"><code>channelCountMode</code><dt>
<dd><p>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.</p>
<p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a>
section for more information on this attribute.</p>
</dd>
</dl>
<dl>
<dt id="dfn-channelInterpretation"><code>channelInterpretation</code><dt>
<dd><p>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.</p>
<p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a>
section for more information on this attribute.</p>
</dd>
</dl>
</div>
<div id="methodsandparams-AudioNode-section" class="section">
<h3 id="methodsandparams-AudioNode">4.2.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-connect-AudioNode">The <code>connect</code> to AudioNode method</dt>
<dd><p>Connects the AudioNode to another AudioNode.</p>
<p>The <dfn id="dfn-destination_2">destination</dfn> parameter is the
AudioNode to connect to.</p>
<p>The <dfn id="dfn-output_2">output</dfn> parameter is an index
describing which output of the AudioNode from which to connect. An
out-of-bound value throws an exception.</p>
<p>The <dfn id="dfn-input_2">input</dfn> parameter is an index describing
which input of the destination AudioNode to connect to. An out-of-bound
value throws an exception. </p>
<p>It is possible to connect an AudioNode output to more than one input
with multiple calls to connect(). Thus, "fan-out" is supported. </p>
<p>
It is possible to connect an AudioNode to another AudioNode which creates a <em>cycle</em>.
In other words, 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
<a class="dfnref" href="#DelayNode-section">DelayNode</a> in the <em>cycle</em> or an exception will
be thrown.
</p>
<p>
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:
</p>
<pre>
nodeA.connect(nodeB);
nodeA.connect(nodeB);
will have the same effect as
nodeA.connect(nodeB);
</pre>
</dd>
</dl>
<dl>
<dt id="dfn-connect-AudioParam">The <code>connect</code> to AudioParam method</dt>
<dd><p>Connects the AudioNode to an AudioParam, controlling the parameter
value with an audio-rate signal.
</p>
<p>The <dfn id="dfn-destination_3">destination</dfn> parameter is the
AudioParam to connect to.</p>
<p>The <dfn id="dfn-output_3-destination">output</dfn> parameter is an index
describing which output of the AudioNode from which to connect. An
out-of-bound value throws an exception.</p>
<p>It is possible to connect an AudioNode output to more than one AudioParam
with multiple calls to connect(). Thus, "fan-out" is supported. </p>
<p>It is possible to connect more than one AudioNode output to a single AudioParam
with multiple calls to connect(). Thus, "fan-in" is supported. </p>
<p>An AudioParam will take the rendered audio data from any AudioNode output connected to it and <a href="#down-mix">convert it to mono</a> by down-mixing if it is not
already mono, then mix it together with other such outputs and finally will mix with the <em>intrinsic</em>
parameter value (the value the AudioParam would normally have without any audio connections), including any timeline changes
scheduled for the parameter. </p>
<p>
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:
</p>
<pre>
nodeA.connect(param);
nodeA.connect(param);
will have the same effect as
nodeA.connect(param);
</pre>
</dd>
</dl>
<dl>
<dt id="dfn-disconnect">The <code>disconnect</code> method</dt>
<dd><p>Disconnects an AudioNode's output.</p>
<p>The <dfn id="dfn-output_3-disconnect">output</dfn> parameter is an index
describing which output of the AudioNode to disconnect. An out-of-bound
value throws an exception.</p>
</dd>
</dl>
</div>
</div>
<h3 id="lifetime-AudioNode">4.2.3. Lifetime</h3>
<p class="norm">This section is informative.</p>
<p>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.
</p>
<p>
An <code>AudioNode</code> will live as long as there are any references to it. There are several types of references:
</p>
<ol>
<li>A <em>normal</em> JavaScript reference obeying normal garbage collection rules. </li>
<li>A <em>playing</em> reference for both <code>AudioBufferSourceNodes</code> and <code>OscillatorNodes</code>.
These nodes maintain a <em>playing</em>
reference to themselves while they are currently playing.</li>
<li>A <em>connection</em> reference which occurs if another <code>AudioNode</code> is connected to it. </li>
<li>A <em>tail-time</em> reference which an <code>AudioNode</code> maintains on itself as long as it has
any internal processing state which has not yet been emitted. For example, a <code>ConvolverNode</code> 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 <code>AudioNodes</code> have this
property. Please see details for specific nodes.</li>
</ol>
<p>
Any <code>AudioNodes</code> which are connected in a cycle <em>and</em> are directly or indirectly connected to the
<code>AudioDestinationNode</code> of the <code>AudioContext</code> will stay alive as long as the <code>AudioContext</code> is alive.
</p>
<p>
When an <code>AudioNode</code> has no references it will be deleted. But before it is deleted, it will disconnect itself
from any other <code>AudioNodes</code> which it is connected to. In this way it releases all connection references (3) it has to other nodes.
</p>
<p>
Regardless of any of the above references, it can be assumed that the <code>AudioNode</code> will be deleted when its <code>AudioContext</code> is deleted.
</p>
<div id="AudioDestinationNode-section" class="section">
<h2 id="AudioDestinationNode">4.4. The AudioDestinationNode Interface</h2>
<p>This is an <a href="#AudioNode-section"><code>AudioNode</code></a>
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
<code>destination</code> attribute of <a
href="#AudioContext-section"><code>AudioContext</code></a>. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 0
channelCount = 2;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-destination-node-idl">
interface <dfn id="dfn-AudioDestinationNode">AudioDestinationNode</dfn> : AudioNode {
readonly attribute unsigned long maxChannelCount;
};
</code></pre>
</div>
</div>
<div id="attributes-AudioDestinationNode-section" class="section">
<h3 id="attributes-AudioDestinationNode">4.4.1. Attributes</h3>
<dl>
<dt id="dfn-maxChannelCount"><code>maxChannelCount</code></dt>
<dd><p>The maximum number of channels that the <code>channelCount</code> attribute can be set to.
An <code>AudioDestinationNode</code> 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. <code>maxChannelCount</code> is the maximum number of channels that
this hardware is capable of supporting. If this value is 0, then this indicates that <code>channelCount</code> may not be
changed. This will be the case for an <code>AudioDestinationNode</code> in an <code>OfflineAudioContext</code> and also for
basic implementations with hardware support for stereo output only.</p>
<p><code>channelCount</code> defaults to 2 for a destination in a normal AudioContext, and may be set to any non-zero value less than or equal
to <code>maxChannelCount</code>. An exception will 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 <code>numberOfChannels</code> to 8, and render 8-channels of output.
</p>
<p>
For an AudioDestinationNode in an OfflineAudioContext, the <code>channelCount</code> is determined when the offline context is created and this value
may not be changed.
</p>
</dd>
</dl>
</div>
</div>
<div id="AudioParam-section" class="section">
<h2 id="AudioParam">4.5. The AudioParam Interface</h2>
<p>AudioParam controls an individual aspect of an <a
href="#AudioNode-section"><code>AudioNode</code></a>'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.currentTime), 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 <code>AudioNodes</code> can be connected
to an <code>AudioParam</code>, summing with the <em>intrinsic</em> parameter value.
</p>
<p>
Some synthesis and processing <code>AudioNodes</code> have <code>AudioParams</code> as attributes whose values must
be taken into account on a per-audio-sample basis.
For other <code>AudioParams</code>, sample-accuracy is not important and the value changes can be sampled more coarsely.
Each individual <code>AudioParam</code> will specify that it is either an <em>a-rate</em> parameter
which means that its values must be taken into account on a per-audio-sample basis, or it is a <em>k-rate</em> parameter.
</p>
<p>
Implementations must use block processing, with each <code>AudioNode</code>
processing 128 sample-frames in each block.
</p>
<p>
For each 128 sample-frame block, the value of a <em>k-rate</em> parameter must
be sampled at the time of the very first sample-frame, and that value must be
used for the entire block. <em>a-rate</em> parameters must be sampled for each
sample-frame of the block.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-param-idl">
interface <dfn id="dfn-AudioParam">AudioParam</dfn> {
attribute float value;
readonly attribute float defaultValue;
<span class="comment">// Parameter automation. </span>
void setValueAtTime(float value, double startTime);
void linearRampToValueAtTime(float value, double endTime);
void exponentialRampToValueAtTime(float value, double endTime);
<span class="comment">// Exponentially approach the target value with a rate having the given time constant. </span>
void setTargetAtTime(float target, double startTime, double timeConstant);
<span class="comment">// Sets an array of arbitrary parameter values starting at time for the given duration. </span>
<span class="comment">// The number of values will be scaled to fit into the desired duration. </span>
void setValueCurveAtTime(Float32Array values, double startTime, double duration);
<span class="comment">// Cancels all scheduled parameter changes with times greater than or equal to startTime. </span>
void cancelScheduledValues(double startTime);
};
</code></pre>
</div>
</div>
<div id="attributes-AudioParam-section" class="section">
<h3 id="attributes-AudioParam">4.5.1. Attributes</h3>
<dl>
<dt id="dfn-value"><code>value</code></dt>
<dd><p>The parameter's floating-point value. This attribute is initialized to the
<code>defaultValue</code>. If a value is set during a time when there are any automation events scheduled then
it will be ignored and no exception will be thrown.</p>
</dd>
</dl>
<dl>
<dt id="dfn-defaultValue"><code>defaultValue</code></dt>
<dd><p>Initial value for the value attribute</p>
</dd>
</dl>
</div>
<div id="methodsandparams-AudioParam-section" class="section">
<h3 id="methodsandparams-AudioParam">4.5.2. Methods and Parameters</h3>
<p>
An <code>AudioParam</code> maintains a time-ordered event list which is initially empty. The times are in
the time coordinate system of AudioContext.currentTime. 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 <em>automation</em> methods:</p>
<ul>
<li>setValueAtTime() - <em>SetValue</em></li>
<li>linearRampToValueAtTime() - <em>LinearRampToValue</em></li>
<li>exponentialRampToValueAtTime() - <em>ExponentialRampToValue</em></li>
<li>setTargetAtTime() - <em>SetTarget</em></li>
<li>setValueCurveAtTime() - <em>SetValueCurve</em></li>
</ul>
<p>
The following rules will apply when calling these methods:
</p>
<ul>
<li>If one of these events is added at a time where there is already an event of the exact same type, then the new event will replace the old
one.</li>
<li>If one of these events is added at a time where there is already one or more events of a different type, then it will be
placed in the list after them, but before events whose times are after the event. </li>
<li>If setValueCurveAtTime() is called for time T and duration D and there are any events having a time greater than T, but less than
T + D, then an exception will be thrown. In other words, it's not ok to schedule a value curve during a time period containing other events.</li>
<li>Similarly an exception will be thrown if any <em>automation</em> method is called at a time which is inside of the time interval
of a <em>SetValueCurve</em> event at time T and duration D.</li>
</ul>
<p>
</p>
<dl>
<dt id="dfn-setValueAtTime">The <code>setValueAtTime</code> method</dt>
<dd><p>Schedules a parameter value change at the given time.</p>
<p>The <dfn id="dfn-value_2">value</dfn> parameter is the value the
parameter will change to at the given time.</p>
<p>The <dfn id="dfn-startTime_2">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>
If there are no more events after this <em>SetValue</em> event, then for t >= startTime, v(t) = value. In other words, the value will remain constant.
</p>
<p>
If the next event (having time T1) after this <em>SetValue</em> event is not of type <em>LinearRampToValue</em> or <em>ExponentialRampToValue</em>,
then, for t: startTime <= t < T1, v(t) = value.
In other words, the value will remain constant during this time interval, allowing the creation of "step" functions.
</p>
<p>
If the next event after this <em>SetValue</em> event is of type <em>LinearRampToValue</em> or <em>ExponentialRampToValue</em> then please
see details below.
</p>
</dd>
</dl>
<dl>
<dt id="dfn-linearRampToValueAtTime">The <code>linearRampToValueAtTime</code>
method</dt>
<dd><p>Schedules a linear continuous change in parameter value from the
previous scheduled parameter value to the given value.</p>
<p>The <dfn id="dfn-value_3">value</dfn> parameter is the value the
parameter will linearly ramp to at the given time.</p>
<p>The <dfn id="dfn-endTime_3">endTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>
The value during the time interval T0 <= t < T1 (where T0 is the time of the previous event and T1 is the endTime parameter passed into this method)
will be calculated as:
</p>
<pre>
v(t) = V0 + (V1 - V0) * ((t - T0) / (T1 - T0))
</pre>
<p>
Where V0 is the value at the time T0 and V1 is the value parameter passed into this method.
</p>
<p>
If there are no more events after this LinearRampToValue event then for t >= T1, v(t) = V1
</p>
</dd>
</dl>
<dl>
<dt id="dfn-exponentialRampToValueAtTime">The
<code>exponentialRampToValueAtTime</code> method</dt>
<dd><p>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. </p>
<p>The <dfn id="dfn-value_4">value</dfn> parameter is the value the
parameter will exponentially ramp to at the given time. An exception will be thrown if this value is less than
or equal to 0, or if the value at the time of the previous event is less than or equal to 0.</p>
<p>The <dfn id="dfn-endTime_4">endTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>
The value during the time interval T0 <= t < T1 (where T0 is the time of the previous event and T1 is the endTime parameter passed into this method)
will be calculated as:
</p>
<pre>
v(t) = V0 * (V1 / V0) ^ ((t - T0) / (T1 - T0))
</pre>
<p>
Where V0 is the value at the time T0 and V1 is the value parameter passed into this method.
</p>
<p>
If there are no more events after this ExponentialRampToValue event then for t >= T1, v(t) = V1
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setTargetAtTime">The <code>setTargetAtTime</code>
method</dt>
<dd><p>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.</p>
<p>The <dfn id="dfn-target">target</dfn> parameter is the value
the parameter will <em>start</em> changing to at the given time.</p>
<p>The <dfn id="dfn-startTime">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>The <dfn id="dfn-timeConstant">timeConstant</dfn> parameter is the
time-constant value of first-order filter (exponential) approach to the
target value. The larger this value is, the slower the transition will
be.</p>
<p>
More precisely, <em>timeConstant</em> 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).
</p>
<p>
During the time interval: <em>T0</em> <= t < <em>T1</em>, where T0 is the <em>startTime</em> parameter and T1 represents the time of the event following this
event (or <em>infinity</em> if there are no following events):
</p>
<pre>
v(t) = V1 + (V0 - V1) * exp(-(t - T0) / <em>timeConstant</em>)
</pre>
<p>
Where V0 is the initial value (the .value attribute) at T0 (the <em>startTime</em> parameter) and V1 is equal to the <em>target</em>
parameter.
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setValueCurveAtTime">The <code>setValueCurveAtTime</code>
method</dt>
<dd><p>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. </p>
<p>The <dfn id="dfn-values">values</dfn> parameter is a Float32Array
representing a parameter value curve. These values will apply starting at
the given time and lasting for the given duration. </p>
<p>The <dfn id="dfn-startTime_5">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>The <dfn id="dfn-duration_5">duration</dfn> parameter is the
amount of time in seconds (after the <em>time</em> parameter) where values will be calculated according to the <em>values</em> parameter..</p>
<p>
During the time interval: <em>startTime</em> <= t < <em>startTime</em> + <em>duration</em>, values will be calculated:
</p>
<pre>
v(t) = values[N * (t - startTime) / duration], where <em>N</em> is the length of the <em>values</em> array.
</pre>
<p>
After the end of the curve time interval (t >= <em>startTime</em> + <em>duration</em>), the value will remain constant at the final curve value,
until there is another automation event (if any).
</p>
</dd>
</dl>
<dl>
<dt id="dfn-cancelScheduledValues">The <code>cancelScheduledValues</code>
method</dt>
<dd><p>Cancels all scheduled parameter changes with times greater than or
equal to startTime.</p>
<p>The <dfn>startTime</dfn> parameter is the starting
time at and after which any previously scheduled parameter changes will
be cancelled. It is a time in the same time coordinate system as AudioContext.currentTime.</p>
</dd>
</dl>
</div>
</div>
<div id="computedValue-AudioParam-section" class="section">
<h3>4.5.3. Computation of Value</h3>
<p>
<dfn>computedValue</dfn> 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:
</p>
<ol>
<li>An <em>intrinsic</em> 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. If the .value attribute
is set after any automation events have been scheduled, then these events will be removed. When read, the .value attribute
always returns the <em>intrinsic</em> value for the current time. If automation events are removed from a given time range, then the
<em>intrinsic</em> 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.
</li>
<li>
An AudioParam will take the rendered audio data from any AudioNode output connected to it and <a href="#down-mix">convert it to mono</a> by down-mixing if it is not
already mono, then mix it together with other such outputs. If there are no AudioNodes connected to it, then this value is 0, having no
effect on the <em>computedValue</em>.
</li>
<li>
The <em>computedValue</em> is the sum of the <em>intrinsic</em> value and the value calculated from (2).
</li>
</ol>
</div>
<div id="example1-AudioParam-section" class="section">
<h3 id="example1-AudioParam">4.5.4. AudioParam Automation Example</h3>
<div class="example">
<div class="exampleHeader">
Example</div>
<img alt="AudioParam automation" src="images/audioparam-automation1.png" />
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span></div>
<div class="blockContent">
<pre class="code"><code class="es-code">
var t0 = 0;
var t1 = 0.1;
var t2 = 0.2;
var t3 = 0.3;
var t4 = 0.4;
var t5 = 0.6;
var t6 = 0.7;
var t7 = 1.0;
var curveLength = 44100;
var curve = new Float32Array(curveLength);
for (var i = 0; i < curveLength; ++i)
curve[i] = Math.sin(Math.PI * i / curveLength);
param.setValueAtTime(0.2, t0);
param.setValueAtTime(0.3, t1);
param.setValueAtTime(0.4, t2);
param.linearRampToValueAtTime(1, t3);
param.linearRampToValueAtTime(0.15, t4);
param.exponentialRampToValueAtTime(0.75, t5);
param.exponentialRampToValueAtTime(0.05, t6);
param.setValueCurveAtTime(curve, t6, t7 - t6);
</code></pre>
</div>
</div>
</div>
</div>
<div id="GainNode-section" class="section">
<h2 id="GainNode">4.7. The GainNode Interface</h2>
<p>Changing the gain of an audio signal is a fundamental operation in audio
applications. The <code>GainNode</code> is one of the building blocks for creating <a
href="#MixerGainStructure-section">mixers</a>.
This interface is an AudioNode with a single input and single
output: </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>It multiplies the input audio signal by the (possibly time-varying) <code>gain</code> attribute, copying the result to the output.
By default, it will take the input and pass it through to the output unchanged, which represents a constant gain change
of 1.
</p>
<p>
As with other <code>AudioParams</code>, the <code>gain</code> parameter represents a mapping from time
(in the coordinate system of AudioContext.currentTime) to floating-point value.
Every PCM audio sample in the input is multiplied by the <code>gain</code> parameter's value for the specific time
corresponding to that audio sample. This multiplied value represents the PCM audio sample for the output.
</p>
<p>
The number of channels of the output will always equal the number of channels of the input, with each channel
of the input being multiplied by the <code>gain</code> values and being copied into the corresponding channel
of the output.
</p>
<p>
The implementation must make
gain changes to the audio stream smoothly, without introducing noticeable
clicks or glitches. This process is called "de-zippering". </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="gain-node-idl">
interface <dfn id="dfn-GainNode">GainNode</dfn> : AudioNode {
readonly attribute AudioParam gain;
};
</code></pre>
</div>
</div>
<div id="attributes-GainNode-section" class="section">
<h3 id="attributes-GainNode">4.7.1. Attributes</h3>
<dl>
<dt id="dfn-gain"><code>gain</code></dt>
<dd><p>Represents the amount of gain to apply. Its
default <code>value</code> is 1 (no gain change). The nominal <code>minValue</code> is 0, but may be
set negative for phase inversion. The nominal <code>maxValue</code> is 1, but higher values are allowed (no
exception thrown).This parameter is <em>a-rate</em> </p>
</dd>
</dl>
</div>
</div>
<div id="DelayNode-section" class="section">
<h2 id="DelayNode">4.8. The DelayNode Interface</h2>
<p>A delay-line is a fundamental building block in audio applications. This
interface is an AudioNode with a single input and single output: </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>
The number of channels of the output always equals the number of channels of the input.
</p>
<p>It delays the incoming audio signal by a certain amount. The default
amount is 0 seconds (no delay). When the delay time is changed, the
implementation must make the transition smoothly, without introducing
noticeable clicks or glitches to the audio stream. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="delay-node-idl">
interface <dfn id="dfn-DelayNode">DelayNode</dfn> : AudioNode {
readonly attribute AudioParam delayTime;
};
</code></pre>
</div>
</div>
<div id="attributes-GainNode-section_2" class="section">
<h3 id="attributes-GainNode_2">4.8.1. Attributes</h3>
<dl>
<dt id="dfn-delayTime_2"><code>delayTime</code></dt>
<dd><p>An AudioParam object representing the amount of delay (in seconds)
to apply. The default value (<code>delayTime.value</code>) is 0 (no
delay). The minimum value is 0 and the maximum value is determined by the <em>maxDelayTime</em>
argument to the <code>AudioContext</code> method <code>createDelay</code>. This parameter is <em>a-rate</em></p>
</dd>
</dl>
</div>
</div>
<div id="AudioBuffer-section" class="section">
<h2 id="AudioBuffer">4.9. The AudioBuffer Interface</h2>
<p>This interface represents a memory-resident audio asset (for one-shot sounds
and other short audio clips). Its format is non-interleaved IEEE 32-bit linear PCM with a
nominal range of -1 -> +1. It can contain one or more channels. Typically, it would be expected that the length
of the PCM data would be fairly short (usually somewhat less than a minute).
For longer sounds, such as music soundtracks, streaming should be used with the
<code>audio</code> element and <code>MediaElementAudioSourceNode</code>. </p>
<p>
An AudioBuffer may be used by one or more AudioContexts.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-buffer-idl">
interface <dfn id="dfn-AudioBuffer">AudioBuffer</dfn> {
readonly attribute float sampleRate;
readonly attribute long length;
<span class="comment">// in seconds </span>
readonly attribute double duration;
readonly attribute long numberOfChannels;
Float32Array getChannelData(unsigned long channel);
};
</code></pre>
</div>
</div>
<div id="attributes-AudioBuffer-section" class="section">
<h3 id="attributes-AudioBuffer">4.9.1. Attributes</h3>
<dl>
<dt id="dfn-sampleRate_AudioBuffer"><code>sampleRate</code></dt>
<dd><p>The sample-rate for the PCM audio data in samples per second.</p>
</dd>
</dl>
<dl>
<dt id="dfn-length_AudioBuffer"><code>length</code></dt>
<dd><p>Length of the PCM audio data in sample-frames.</p>
</dd>
</dl>
<dl>
<dt id="dfn-duration_AudioBuffer"><code>duration</code></dt>
<dd><p>Duration of the PCM audio data in seconds.</p>
</dd>
</dl>
<dl>
<dt id="dfn-numberOfChannels_AudioBuffer"><code>numberOfChannels</code></dt>
<dd><p>The number of discrete audio channels.</p>
</dd>
</dl>
</div>
<div id="methodsandparams-AudioBuffer-section" class="section">
<h3 id="methodsandparams-AudioBuffer">4.9.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-getChannelData">The <code>getChannelData</code> method</dt>
<dd><p>Returns the <code>Float32Array</code> representing the PCM audio data for the specific channel.</p>
<p>The <dfn id="dfn-channel">channel</dfn> 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 <code>numberOfChannels</code>
or an exception will be thrown.</p>
</dd>
</dl>
</div>
</div>
<div id="AudioBufferSourceNode-section" class="section">
<h2 id="AudioBufferSourceNode">4.10. The AudioBufferSourceNode Interface</h2>
<p>This interface represents an audio source from an in-memory audio asset in
an <code>AudioBuffer</code>. It is useful for playing short audio assets
which require a high degree of scheduling flexibility (can playback in
rhythmically perfect ways). The start() method is used to schedule when
sound playback will happen. The playback will stop automatically when
the buffer's audio data has been completely
played (if the <code>loop</code> attribute is false), or when the stop()
method has been called and the specified time has been reached. Please see more
details in the start() and stop() description. start() and stop() may not be issued
multiple times for a given
AudioBufferSourceNode. </p>
<pre> numberOfInputs : 0
numberOfOutputs : 1
</pre>
<p>
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.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-buffer-source-node-idl">
interface <dfn id="dfn-AudioBufferSourceNode">AudioBufferSourceNode</dfn> : AudioNode {
attribute AudioBuffer? buffer;
readonly attribute AudioParam playbackRate;
attribute boolean loop;
attribute double loopStart;
attribute double loopEnd;
void start(optional double when = 0, optional double offset = 0, optional double duration);
void stop(optional double when = 0);
attribute EventHandler onended;
};
</code></pre>
</div>
</div>
<div id="attributes-AudioBufferSourceNode-section" class="section">
<h3 id="attributes-AudioBufferSourceNode">4.10.1. Attributes</h3>
<dl>
<dt id="dfn-buffer_AudioBufferSourceNode"><code>buffer</code></dt>
<dd><p>Represents the audio asset to be played. </p>
</dd>
</dl>
<dl>
<dt id="dfn-playbackRate_AudioBufferSourceNode"><code>playbackRate</code></dt>
<dd><p>The speed at which to render the audio stream. The default
playbackRate.value is 1. This parameter is <em>a-rate</em> </p>
</dd>
</dl>
<dl>
<dt id="dfn-loop_AudioBufferSourceNode"><code>loop</code></dt>
<dd><p>Indicates if the audio data should play in a loop. The default value is false. </p>
</dd>
</dl>
<dl>
<dt id="dfn-loopStart_AudioBufferSourceNode"><code>loopStart</code></dt>
<dd><p>An optional value in seconds where looping should begin if the <code>loop</code> 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.</p>
</dd>
</dl>
<dl>
<dt id="dfn-loopEnd_AudioBufferSourceNode"><code>loopEnd</code></dt>
<dd><p>An optional value in seconds where looping should end if the <code>loop</code> 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.</p>
</dd>
</dl>
<dl>
<dt id="dfn-onended_AudioBufferSourceNode"><code>onended</code></dt>
<dd><p>A property used to set the <code>EventHandler</code> (described in <cite><a
href="http://www.whatwg.org/specs/web-apps/current-work/#eventhandler">HTML</a></cite>)
for the ended event that is dispatched to <a
href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>
node types. When the playback of the buffer for an <code>AudioBufferSourceNode</code>
is finished, an event of type <code>Event</code> (described in <cite><a
href="http://www.whatwg.org/specs/web-apps/current-work/#event">HTML</a></cite>)
will be dispatched to the event handler. </p>
</dd>
</dl>
</div>
</div>
<div id="methodsandparams-AudioBufferSourceNode-section" class="section">
<h3 id="methodsandparams-AudioBufferSourceNode">4.10.2. Methods and
Parameters</h3>
<dl>
<dt id="dfn-start">The <code>start</code> method</dt>
<dd><p>Schedules a sound to playback at an exact time.</p>
<p>The <dfn id="dfn-when">when</dfn> parameter describes at what time (in
seconds) the sound should start playing. It is in the same
time coordinate system as AudioContext.currentTime. If 0 is passed in for
this value or if the value is less than <b>currentTime</b>, then the
sound will start playing immediately. <code>start</code> may only be called one time
and must be called before <code>stop</code> is called or an exception will be thrown.</p>
<p>The <dfn id="dfn-offset">offset</dfn> 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.</p>
<p>The <dfn id="dfn-duration">duration</dfn> 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 <code>offset</code> parameter.
Thus if neither <code>offset</code> nor <code>duration</code> are specified then the implied duration is
the total duration of the AudioBuffer.
</p>
</dd>
</dl>
<dl>
<dt id="dfn-stop">The <code>stop</code> method</dt>
<dd><p>Schedules a sound to stop playback at an exact time.</p>
<p>The <dfn id="dfn-when_AudioBufferSourceNode_2">when</dfn> parameter
describes at what time (in seconds) the sound should stop playing.
It is in the same time coordinate system as AudioContext.currentTime.
If 0 is passed in for this value or if the value is less than
<b>currentTime</b>, then the sound will stop playing immediately.
<code>stop</code> must only be called one time and only after a call to <code>start</code> or <code>stop</code>,
or an exception will be thrown.</p>
</dd>
</dl>
</div>
<div id="looping-AudioBufferSourceNode-section" class="section">
<h3 id="looping-AudioBufferSourceNode">4.10.3. Looping</h3>
<p>
If the <code>loop</code> attribute is true when <code>start()</code> is called, then playback will continue indefinitely
until <code>stop()</code> 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 <code>offset</code> argument of <code>start()</code>, and in <em>loop</em> mode will continue playing until it reaches the <em>actualLoopEnd</em> position
in the buffer (or the end of the buffer), at which point it will wrap back around to the <em>actualLoopStart</em> position in the buffer, and continue
playing according to this pattern.
</p>
<p>
In <em>loop</em> mode then the <em>actual</em> loop points are calculated as follows from the <code>loopStart</code> and <code>loopEnd</code> attributes:
</p>
<blockquote>
<pre>
if ((loopStart || loopEnd) && loopStart >= 0 && loopEnd > 0 && loopStart < loopEnd) {
actualLoopStart = loopStart;
actualLoopEnd = min(loopEnd, buffer.length);
} else {
actualLoopStart = 0;
actualLoopEnd = buffer.length;
}
</pre>
</blockquote>
<p>
Note that the default values for <code>loopStart</code> and <code>loopEnd</code> are both 0, which indicates that looping should occur from the very start
to the very end of the buffer.
</p>
<p>
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.
</p>
</div>
<div id="MediaElementAudioSourceNode-section" class="section">
<h2 id="MediaElementAudioSourceNode">4.11. The MediaElementAudioSourceNode
Interface</h2>
<p>This interface represents an audio source from an <code>audio</code> or
<code>video</code> element. </p>
<pre> numberOfInputs : 0
numberOfOutputs : 1
</pre>
<p>
The number of channels of the output corresponds to the number of channels of the media referenced by the HTMLMediaElement.
Thus, changes to the media element's .src attribute can change the number of channels output by this node.
If the .src attribute is not set, then the number of channels output will be one silent channel.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="media-element-audio-source-node-idl">
interface <dfn id="dfn-MediaElementAudioSourceNode">MediaElementAudioSourceNode</dfn> : AudioNode {
};
</code></pre>
</div>
</div>
</div>
<p>A MediaElementAudioSourceNode
is created given an HTMLMediaElement using the AudioContext <a href="#dfn-createMediaElementSource">createMediaElementSource()</a> method. </p>
<p>
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.
</p>
<p>
The HTMLMediaElement must behave in an identical fashion after the MediaElementAudioSourceNode has
been created, <em>except</em> 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, <code>.src</code> attribute changes, and other aspects of the HTMLMediaElement must behave as they normally would
if <em>not</em> used with a MediaElementAudioSourceNode.
</p>
<div class="example">
<div class="exampleHeader">
Example</div>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span></div>
<div class="blockContent">
<pre class="code"><code class="es-code">
var mediaElement = document.getElementById('mediaElementID');
var sourceNode = context.createMediaElementSource(mediaElement);
sourceNode.connect(filterNode);
</code></pre>
</div>
</div>
</div>
</div>
<div id="ScriptProcessorNode-section" class="section">
<h2 id="ScriptProcessorNode">4.12. The ScriptProcessorNode Interface</h2>
<p>This interface is an AudioNode which can generate, process, or analyse audio
directly using JavaScript. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCount = numberOfInputChannels;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<p>The ScriptProcessorNode is constructed with a <code>bufferSize</code> which
must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384.
This value controls how frequently the <code>audioprocess</code> event
is dispatched and how many sample-frames need to be processed each call.
Lower numbers for <code>bufferSize</code> will result in a lower (better) <a
href="#Latency-section">latency</a>. Higher numbers will be necessary to avoid
audio breakup and <a href="#Glitching-section">glitches</a>.
This value will be picked by the implementation if the bufferSize argument
to <code>createScriptProcessor</code> is not passed in, or is set to 0.</p>
<p><code>numberOfInputChannels</code> and <code>numberOfOutputChannels</code>
determine the number of input and output channels. It is invalid for both
<code>numberOfInputChannels</code> and <code>numberOfOutputChannels</code> to
be zero. </p>
<pre> var node = context.createScriptProcessor(bufferSize, numberOfInputChannels, numberOfOutputChannels);
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="script-processor-node-idl">
interface <dfn id="dfn-ScriptProcessorNode">ScriptProcessorNode</dfn> : AudioNode {
attribute EventHandler onaudioprocess;
readonly attribute long bufferSize;
};
</code></pre>
</div>
</div>
<div id="attributes-ScriptProcessorNode-section" class="section">
<h3 id="attributes-ScriptProcessorNode">4.12.1. Attributes</h3>
<dl>
<dt id="dfn-onaudioprocess"><code>onaudioprocess</code></dt>
<dd><p>A property used to set the <code>EventHandler</code> (described in <cite><a
href="http://www.whatwg.org/specs/web-apps/current-work/#eventhandler">HTML</a></cite>)
for the audioprocess event that is dispatched to <a
href="#ScriptProcessorNode-section"><code>ScriptProcessorNode</code></a>
node types. An event of type <a
href="#AudioProcessingEvent-section"><code>AudioProcessingEvent</code></a>
will be dispatched to the event handler. </p>
</dd>
</dl>
<dl>
<dt id="dfn-bufferSize_ScriptProcessorNode"><code>bufferSize</code></dt>
<dd><p>The size of the buffer (in sample-frames) which needs to be
processed each time <code>onprocessaudio</code> is called. Legal values
are (256, 512, 1024, 2048, 4096, 8192, 16384). </p>
</dd>
</dl>
</div>
</div>
<div id="AudioProcessingEvent-section" class="section">
<h2 id="AudioProcessingEvent">4.13. The AudioProcessingEvent Interface</h2>
<p>This is an <code>Event</code> object which is dispatched to <a
href="#ScriptProcessorNode-section"><code>ScriptProcessorNode</code></a> nodes. </p>
<p>The event handler processes audio from the input (if any) by accessing the
audio data from the <code>inputBuffer</code> 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 <code>outputBuffer</code>. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-processing-event-idl">
interface <dfn id="dfn-AudioProcessingEvent">AudioProcessingEvent</dfn> : Event {
readonly attribute double playbackTime;
readonly attribute AudioBuffer inputBuffer;
readonly attribute AudioBuffer outputBuffer;
};
</code></pre>
</div>
</div>
<div id="attributes-AudioProcessingEvent-section" class="section">
<h3 id="attributes-AudioProcessingEvent">4.13.1. Attributes</h3>
<dl>
<dt id="dfn-playbackTime"><code>playbackTime</code></dt>
<dd><p>The time when the audio will be played in the same time coordinate system as AudioContext.currentTime.
<code>playbackTime</code> allows for very tight synchronization between
processing directly in JavaScript with the other events in the context's
rendering graph. </p>
</dd>
</dl>
<dl>
<dt id="dfn-inputBuffer"><code>inputBuffer</code></dt>
<dd><p>An AudioBuffer containing the input audio data. It will have a number of channels equal to the <code>numberOfInputChannels</code> parameter
of the createScriptProcessor() method. This AudioBuffer is only valid while in the scope of the <code>onaudioprocess</code>
function. Its values will be meaningless outside of this scope.</p>
</dd>
</dl>
<dl>
<dt id="dfn-outputBuffer"><code>outputBuffer</code></dt>
<dd><p>An AudioBuffer where the output audio data should be written. It will have a number of channels equal to the
<code>numberOfOutputChannels</code> parameter of the createScriptProcessor() method.
Script code within the scope of the <code>onaudioprocess</code> function is expected to modify the
<code>Float32Array</code> arrays representing channel data in this AudioBuffer.
Any script modifications to this AudioBuffer outside of this scope will not produce any audible effects.</p>
</dd>
</dl>
</div>
</div>
<div id="PannerNode-section" class="section">
<h2 id="PannerNode">4.14. The PannerNode Interface</h2>
<p>This interface represents a processing node which <a
href="#Spatialization-section">positions / spatializes</a> an incoming audio
stream in three-dimensional space. The spatialization is in relation to the <a
href="#AudioContext-section"><code>AudioContext</code></a>'s <a
href="#AudioListener-section"><code>AudioListener</code></a>
(<code>listener</code> attribute). </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCount = 2;
channelCountMode = "clamped-max";
channelInterpretation = "speakers";
</pre>
<p>
The audio stream from the input will be either mono or stereo, depending on the connection(s) to the input.
</p>
<p>
The output of this node is hard-coded to stereo (2 channels) and <em>currently</em> cannot be configured.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="panner-node-idl">
enum <dfn>PanningModelType</dfn> {
"equalpower",
"HRTF"
};
enum <dfn>DistanceModelType</dfn> {
"linear",
"inverse",
"exponential"
};
interface <dfn id="dfn-PannerNode">PannerNode</dfn> : AudioNode {
<span class="comment">// Default for stereo is HRTF </span>
attribute PanningModelType panningModel;
<span class="comment">// Uses a 3D cartesian coordinate system </span>
void setPosition(double x, double y, double z);
void setOrientation(double x, double y, double z);
void setVelocity(double x, double y, double z);
<span class="comment">// Distance model and attributes </span>
attribute DistanceModelType distanceModel;
attribute double refDistance;
attribute double maxDistance;
attribute double rolloffFactor;
<span class="comment">// Directional sound cone </span>
attribute double coneInnerAngle;
attribute double coneOuterAngle;
attribute double coneOuterGain;
};
</code></pre>
</div>
</div>
</div>
<div id="attributes-PannerNode_attributes-section" class="section">
<h3 id="attributes-PannerNode_attributes">4.14.2. Attributes</h3>
<dl>
<dt id="dfn-panningModel"><code>panningModel</code></dt>
<dd><p>Determines which spatialization algorithm will be used to position
the audio in 3D space. The default is "HRTF". </p>
<dl>
<dt id="dfn-EQUALPOWER"><code>"equalpower"</code></dt>
<dd><p>A simple and efficient spatialization algorithm using equal-power
panning. </p>
</dd>
</dl>
<dl>
<dt id="dfn-HRTF"><code>"HRTF"</code></dt>
<dd><p>A higher quality spatialization algorithm using a convolution with
measured impulse responses from human subjects. This panning method
renders stereo output. </p>
</dd>
</dl>
</dd>
</dl>
<dl>
<dt id="dfn-distanceModel"><code>distanceModel</code></dt>
<dd><p>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".
</p>
<dl>
<dt id="dfn-LINEAR_DISTANCE"><code>"linear"</code></dt>
<dd><p>A linear distance model which calculates <em>distanceGain</em> according to: </p>
<pre>
1 - rolloffFactor * (distance - refDistance) / (maxDistance - refDistance)
</pre>
</dd>
</dl>
<dl>
<dt id="dfn-INVERSE_DISTANCE"><code>"inverse"</code></dt>
<dd><p>An inverse distance model which calculates <em>distanceGain</em> according to: </p>
<pre>
refDistance / (refDistance + rolloffFactor * (distance - refDistance))
</pre>
</dd>
</dl>
<dl>
<dt id="dfn-EXPONENTIAL_DISTANCE"><code>"exponential"</code></dt>
<dd><p>An exponential distance model which calculates <em>distanceGain</em> according to: </p>
<pre>
pow(distance / refDistance, -rolloffFactor)
</pre>
</dd>
</dl>
</dd>
</dl>
<dl>
<dt id="dfn-refDistance"><code>refDistance</code></dt>
<dd><p>A reference distance for reducing volume as source move further from
the listener. The default value is 1. </p>
</dd>
</dl>
<dl>
<dt id="dfn-maxDistance"><code>maxDistance</code></dt>
<dd><p>The maximum distance between source and listener, after which the
volume will not be reduced any further. The default value is 10000. </p>
</dd>
</dl>
<dl>
<dt id="dfn-rolloffFactor"><code>rolloffFactor</code></dt>
<dd><p>Describes how quickly the volume is reduced as source moves away
from listener. The default value is 1. </p>
</dd>
</dl>
<dl>
<dt id="dfn-coneInnerAngle"><code>coneInnerAngle</code></dt>
<dd><p>A parameter for directional audio sources, this is an angle, inside
of which there will be no volume reduction. The default value is 360. </p>
</dd>
</dl>
<dl>
<dt id="dfn-coneOuterAngle"><code>coneOuterAngle</code></dt>
<dd><p>A parameter for directional audio sources, this is an angle, outside
of which the volume will be reduced to a constant value of
<b>coneOuterGain</b>. The default value is 360. </p>
</dd>
</dl>
<dl>
<dt id="dfn-coneOuterGain"><code>coneOuterGain</code></dt>
<dd><p>A parameter for directional audio sources, this is the amount of
volume reduction outside of the <b>coneOuterAngle</b>. The default value is 0. </p>
</dd>
</dl>
</div>
<h3 id="Methods_and_Parameters">4.14.3. Methods and Parameters</h3>
<dl>
<dt id="dfn-setPosition">The <code>setPosition</code> method</dt>
<dd><p>Sets the position of the audio source relative to the
<b>listener</b> attribute. A 3D cartesian coordinate system is used.</p>
<p>The <dfn id="dfn-x">x, y, z</dfn> parameters represent the coordinates
in 3D space. </p>
<p>The default value is (0,0,0)
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setOrientation">The <code>setOrientation</code> method</dt>
<dd><p>Describes which direction the audio source is pointing in the 3D
cartesian coordinate space. Depending on how directional the sound is
(controlled by the <b>cone</b> attributes), a sound pointing away from
the listener can be very quiet or completely silent.</p>
<p>The <dfn id="dfn-x_2">x, y, z</dfn> parameters represent a direction
vector in 3D space. </p>
<p>The default value is (1,0,0)
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setVelocity">The <code>setVelocity</code> method</dt>
<dd><p>Sets the velocity vector of the audio source. This vector controls
both the direction of travel and the speed in 3D space. This velocity
relative to the listener's velocity is used to determine how much doppler
shift (pitch change) to apply. The units used for this vector is <em>meters / second</em>
and is independent of the units used for position and orientation vectors.</p>
<p>The <dfn id="dfn-x_3">x, y, z</dfn> parameters describe a direction
vector indicating direction of travel and intensity. </p>
<p>The default value is (0,0,0)
</p>
</dd>
</dl>
<div id="AudioListener-section" class="section">
<h2 id="AudioListener">4.15. The AudioListener Interface</h2>
<p>This interface represents the position and orientation of the person
listening to the audio scene. All <a
href="#PannerNode-section"><code>PannerNode</code></a> objects
spatialize in relation to the AudioContext's <code>listener</code>. See <a
href="#Spatialization-section">this</a> section for more details about
spatialization. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-listener-idl">
interface <dfn id="dfn-AudioListener">AudioListener</dfn> {
attribute double dopplerFactor;
attribute double speedOfSound;
<span class="comment">// Uses a 3D cartesian coordinate system </span>
void setPosition(double x, double y, double z);
void setOrientation(double x, double y, double z, double xUp, double yUp, double zUp);
void setVelocity(double x, double y, double z);
};
</code></pre>
</div>
</div>
</div>
<div id="attributes-AudioListener-section" class="section">
<h3 id="attributes-AudioListener">4.15.1. Attributes</h3>
<dl>
<dt id="dfn-dopplerFactor"><code>dopplerFactor</code></dt>
<dd><p>A constant used to determine the amount of pitch shift to use when
rendering a doppler effect. The default value is 1. </p>
</dd>
</dl>
<dl>
<dt id="dfn-speedOfSound"><code>speedOfSound</code></dt>
<dd><p>The speed of sound used for calculating doppler shift. The default
value is 343.3. </p>
</dd>
</dl>
</div>
<h3 id="L15842">4.15.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-setPosition_2">The <code>setPosition</code> method</dt>
<dd><p>Sets the position of the listener in a 3D cartesian coordinate
space. <code>PannerNode</code> objects use this position relative to
individual audio sources for spatialization.</p>
<p>The <dfn id="dfn-x_AudioListener">x, y, z</dfn> parameters represent
the coordinates in 3D space. </p>
<p>The default value is (0,0,0)
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setOrientation_2">The <code>setOrientation</code> method</dt>
<dd><p>Describes which direction the listener is pointing in the 3D
cartesian coordinate space. Both a <b>front</b> vector and an <b>up</b>
vector are provided. In simple human terms, the <b>front</b> vector represents which
direction the person's nose is pointing. The <b>up</b> 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
<a href="#Spatialization-section">spatialization section</a>.
</p>
<p>The <dfn id="dfn-x_setOrientation">x, y, z</dfn> parameters represent
a <b>front</b> direction vector in 3D space, with the default value being (0,0,-1) </p>
<p>The <dfn id="dfn-x_setOrientation_2">xUp, yUp, zUp</dfn> parameters
represent an <b>up</b> direction vector in 3D space, with the default value being (0,1,0) </p>
</dd>
</dl>
<dl>
<dt id="dfn-setVelocity_4">The <code>setVelocity</code> method</dt>
<dd><p>Sets the velocity vector of the listener. This vector controls both
the direction of travel and the speed in 3D space. This velocity relative to
an audio source's velocity is used to determine how much doppler shift
(pitch change) to apply. The units used for this vector is <em>meters / second</em>
and is independent of the units used for position and orientation vectors.</p>
<p>The <dfn id="dfn-x_setVelocity_5">x, y, z</dfn> parameters describe a
direction vector indicating direction of travel and intensity. </p>
<p>The default value is (0,0,0)
</p>
</dd>
</dl>
<div id="ConvolverNode-section" class="section">
<h2 id="ConvolverNode">4.16. The ConvolverNode Interface</h2>
<p>This interface represents a processing node which applies a <a
href="#Convolution-section">linear convolution effect</a> given an impulse
response. Normative requirements for multi-channel convolution matrixing are described
<a href="#Convolution-reverb-effect">here</a>. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCount = 2;
channelCountMode = "clamped-max";
channelInterpretation = "speakers";
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="convolver-node-idl">
interface <dfn id="dfn-ConvolverNode">ConvolverNode</dfn> : AudioNode {
attribute AudioBuffer? buffer;
attribute boolean normalize;
};
</code></pre>
</div>
</div>
</div>
<div id="attributes-ConvolverNode-section" class="section">
<h3 id="attributes-ConvolverNode">4.16.1. Attributes</h3>
<dl>
<dt id="dfn-buffer_ConvolverNode"><code>buffer</code></dt>
<dd><p>A mono, stereo, or 4-channel <code>AudioBuffer</code> containing the (possibly multi-channel) impulse response
used by the ConvolverNode. This <code>AudioBuffer</code> must be of the same sample-rate as the AudioContext or an exception will
be thrown. At the time when this attribute is set, the <em>buffer</em> and the state of the <em>normalize</em>
attribute will be used to configure the ConvolverNode with this impulse response having the given normalization.
The initial value of this attribute is null.</p>
</dd>
</dl>
<dl>
<dt id="dfn-normalize"><code>normalize</code></dt>
<dd><p>Controls whether the impulse response from the buffer will be scaled
by an equal-power normalization when the <code>buffer</code> atttribute
is set. Its default value is <code>true</code> in order to achieve a more
uniform output level from the convolver when loaded with diverse impulse
responses. If <code>normalize</code> is set to <code>false</code>, 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 <em>buffer</em> attribute is set. </p>
</dd>
</dl>
<p>
If the <em>normalize</em> attribute is false when the <em>buffer</em> attribute is set then the
ConvolverNode will perform a linear convolution given the exact impulse response contained within the <em>buffer</em>.
</p>
<p>
Otherwise, if the <em>normalize</em> attribute is true when the <em>buffer</em> attribute is set then the
ConvolverNode will first perform a scaled RMS-power analysis of the audio data contained within <em>buffer</em> to calculate a
<em>normalizationScale</em> given this algorithm:
</p>
<div class="block">
<div class="blockTitleDiv">
<div class="blockContent">
<pre class="code"><code class="es-code">
float calculateNormalizationScale(buffer)
{
const float GainCalibration = 0.00125;
const float GainCalibrationSampleRate = 44100;
const float MinPower = 0.000125;
// Normalize by RMS power.
size_t numberOfChannels = buffer->numberOfChannels();
size_t length = buffer->length();
float power = 0;
for (size_t i = 0; i < numberOfChannels; ++i) {
float* sourceP = buffer->channel(i)->data();
float channelPower = 0;
int n = length;
while (n--) {
float sample = *sourceP++;
channelPower += sample * sample;
}
power += channelPower;
}
power = sqrt(power / (numberOfChannels * length));
// Protect against accidental overload.
if (isinf(power) || isnan(power) || power < MinPower)
power = MinPower;
float scale = 1 / power;
// Calibrate to make perceived volume same as unprocessed.
scale *= GainCalibration;
// Scale depends on sample-rate.
if (buffer->sampleRate())
scale *= GainCalibrationSampleRate / buffer->sampleRate();
// True-stereo compensation.
if (buffer->numberOfChannels() == 4)
scale *= 0.5;
return scale;
}
</code></pre>
</div>
</div>
</div>
<p>
During processing, the ConvolverNode will then take this calculated <em>normalizationScale</em> value and multiply it by the result of the linear convolution
resulting from processing the input with the impulse response (represented by the <em>buffer</em>) to produce the
final output. Or any mathematically equivalent operation may be used, such as pre-multiplying the
input by <em>normalizationScale</em>, or pre-multiplying a version of the impulse-response by <em>normalizationScale</em>.
</p>
</div>
<div id="AnalyserNode-section" class="section">
<h2 id="AnalyserNode">4.17. The AnalyserNode Interface</h2>
<p>This interface represents a node which is able to provide real-time
frequency and time-domain <a href="#AnalyserNode">analysis</a>
information. The audio stream will be passed un-processed from input to output.
</p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1 <em>Note that this output may be left unconnected.</em>
channelCount = 1;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="analyser-node-idl">
interface <dfn id="dfn-AnalyserNode">AnalyserNode</dfn> : AudioNode {
<span class="comment">// Real-time frequency-domain data </span>
void getFloatFrequencyData(Float32Array array);
void getByteFrequencyData(Uint8Array array);
<span class="comment">// Real-time waveform data </span>
void getByteTimeDomainData(Uint8Array array);
attribute unsigned long fftSize;
readonly attribute unsigned long frequencyBinCount;
attribute double minDecibels;
attribute double maxDecibels;
attribute double smoothingTimeConstant;
};
</code></pre>
</div>
</div>
</div>
<div id="attributes-ConvolverNode-section_2" class="section">
<h3 id="attributes-ConvolverNode_2">4.17.1. Attributes</h3>
<dl>
<dt id="dfn-fftSize"><code>fftSize</code></dt>
<dd><p>The size of the FFT used for frequency-domain analysis. This must be
a non-zero power of two in the range 32 to 2048, otherwise an INDEX_SIZE_ERR exception MUST be thrown.
The default value is 2048.</p>
</dd>
</dl>
<dl>
<dt id="dfn-frequencyBinCount"><code>frequencyBinCount</code></dt>
<dd><p>Half the FFT size. </p>
</dd>
</dl>
<dl>
<dt id="dfn-minDecibels"><code>minDecibels</code></dt>
<dd><p>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 <code>maxDecibels</code>,
an INDEX_SIZE_ERR exception MUST be thrown.</p>
</dd>
</dl>
<dl>
<dt id="dfn-maxDecibels"><code>maxDecibels</code></dt>
<dd><p>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 <code>minDecibels</code>,
an INDEX_SIZE_ERR exception MUST be thrown.</p>
</dd>
</dl>
<dl>
<dt id="dfn-smoothingTimeConstant"><code>smoothingTimeConstant</code></dt>
<dd><p>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 INDEX_SIZE_ERR exception MUST be thrown.</p>
</dd>
</dl>
</div>
<h3 id="methods-and-parameters">4.17.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-getFloatFrequencyData">The <code>getFloatFrequencyData</code>
method</dt>
<dd><p>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.</p>
<p>The <dfn id="dfn-array">array</dfn> parameter is where
frequency-domain analysis data will be copied. </p>
</dd>
</dl>
<dl>
<dt id="dfn-getByteFrequencyData">The <code>getByteFrequencyData</code>
method</dt>
<dd><p>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.</p>
<p>The <dfn id="dfn-array_2">array</dfn> parameter is where
frequency-domain analysis data will be copied. </p>
</dd>
</dl>
<dl>
<dt id="dfn-getByteTimeDomainData">The <code>getByteTimeDomainData</code>
method</dt>
<dd><p>Copies the current time-domain (waveform) data into the passed
unsigned byte array. If the array has fewer elements than the
fftSize, the excess elements will be dropped. If the array has more
elements than fftSize, the excess elements will be ignored.</p>
<p>The <dfn id="dfn-array_3">array</dfn> parameter is where time-domain
analysis data will be copied. </p>
</dd>
</dl>
<div id="ChannelSplitterNode-section" class="section">
<h2 id="ChannelSplitterNode">4.18. The ChannelSplitterNode Interface</h2>
<p>The <code>ChannelSplitterNode</code> is for use in more advanced
applications and would often be used in conjunction with <a
href="#ChannelMergerNode-section"><code>ChannelMergerNode</code></a>. </p>
<pre>
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";
</pre>
<p>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
<code>ChannelSplitterNode</code> 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 <code>numberOfOutputs</code> parameter to the AudioContext method <code>createChannelSplitter()</code>),
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. </p>
<h3 id="example-1">Example:</h3>
<img alt="channel splitter" src="images/channel-splitter.png" />
<p>Please note that in this example, the splitter does <b>not</b> interpret the channel identities (such as left, right, etc.), but
simply splits out channels in the order that they are input.</p>
<p>One application for <code>ChannelSplitterNode</code> is for doing "matrix
mixing" where individual gain control of each channel is desired. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="channel-splitter-node-idl">
interface <dfn id="dfn-ChannelSplitterNode">ChannelSplitterNode</dfn> : AudioNode {
};
</code></pre>
</div>
</div>
</div>
<div id="ChannelMergerNode-section" class="section">
<h2 id="ChannelMergerNode">4.19. The ChannelMergerNode Interface</h2>
<p>The <code>ChannelMergerNode</code> is for use in more advanced applications
and would often be used in conjunction with <a
href="#ChannelSplitterNode-section"><code>ChannelSplitterNode</code></a>. </p>
<pre>
numberOfInputs : Variable N (default to 6) // number of connected inputs may be less than this
numberOfOutputs : 1
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>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 sum of the numbers of channels of all the connected
inputs. For example, if an <code>ChannelMergerNode</code> has two connected
inputs (both stereo), then the output will be four channels, the first two from
the first input and the second two from the second input. In another example
with two connected inputs (both mono), the output will be two channels
(stereo), with the left channel coming from the first input and the right
channel coming from the second input. </p>
<h3 id="example-2">Example:</h3>
<img alt="channel merger" src="images/channel-merger.png" />
<p>Please note that in this example, the merger does <b>not</b> interpret the channel identities (such as left, right, etc.), but
simply combines channels in the order that they are input.</p>
<p>Be aware that it is possible to connect an <code>ChannelMergerNode</code>
in such a way that it outputs an audio stream with a large number of channels
greater than the maximum supported by the audio hardware. In this case where such an output is connected
to the AudioContext .destination (the audio hardware), then the extra channels will be ignored.
Thus, the <code>ChannelMergerNode</code> should be used in situations where the number
of channels is well understood. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="channel-merger-node-idl">
interface <dfn id="dfn-ChannelMergerNode">ChannelMergerNode</dfn> : AudioNode {
};
</code></pre>
</div>
</div>
</div>
<div id="DynamicsCompressorNode-section" class="section">
<h2 id="DynamicsCompressorNode">4.20. The DynamicsCompressorNode Interface</h2>
<p>DynamicsCompressorNode is an AudioNode processor implementing a dynamics
compression effect. </p>
<p>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. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCount = 2;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="dynamics-compressor-node-idl">
interface <dfn id="dfn-DynamicsCompressorNode">DynamicsCompressorNode</dfn> : AudioNode {
readonly attribute AudioParam threshold; // in Decibels
readonly attribute AudioParam knee; // in Decibels
readonly attribute AudioParam ratio; // unit-less
readonly attribute AudioParam reduction; // in Decibels
readonly attribute AudioParam attack; // in Seconds
readonly attribute AudioParam release; // in Seconds
};
</code>
</pre>
</div>
</div>
<div id="attributes-DynamicsCompressorNode-section" class="section">
<h3 id="attributes-DynamicsCompressorNode">4.20.1. Attributes</h3>
<p>
All parameters are <em>k-rate</em>
</p>
<dl>
<dt id="dfn-threshold"><code>threshold</code></dt>
<dd><p>The decibel value above which the compression will start taking
effect. Its default value is -24, with a nominal range of -100 to 0. </p>
</dd>
</dl>
<dl>
<dt id="dfn-knee"><code>knee</code></dt>
<dd><p>A decibel value representing the range above the threshold where the
curve smoothly transitions to the "ratio" portion. Its default value is 30, with a nominal range of 0 to 40. </p>
</dd>
</dl>
<dl>
<dt id="dfn-ratio"><code>ratio</code></dt>
<dd><p>The amount of dB change in input for a 1 dB change in output. Its default value is 12, with a nominal range of 1 to 20. </p>
</dd>
</dl>
<dl>
<dt id="dfn-reduction"><code>reduction</code></dt>
<dd><p>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). The nominal range is -20 to 0. </p>
</dd>
</dl>
<dl>
<dt id="dfn-attack"><code>attack</code></dt>
<dd><p>The amount of time (in seconds) to reduce the gain by 10dB. Its default value is 0.003, with a nominal range of 0 to 1. </p>
</dd>
</dl>
<dl>
<dt id="dfn-release"><code>release</code></dt>
<dd><p>The amount of time (in seconds) to increase the gain by 10dB. Its default value is 0.250, with a nominal range of 0 to 1. </p>
</dd>
</dl>
</div>
</div>
<div id="BiquadFilterNode-section" class="section">
<h2 id="BiquadFilterNode">4.21. The BiquadFilterNode Interface</h2>
<p>BiquadFilterNode is an AudioNode processor implementing very common
low-order filters. </p>
<p>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".</p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>
The number of channels of the output always equals the number of channels of the input.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="biquad-filter-node-idl">
enum <dfn>BiquadFilterType</dfn> {
"lowpass",
"highpass",
"bandpass",
"lowshelf",
"highshelf",
"peaking",
"notch",
"allpass"
};
interface <dfn id="dfn-BiquadFilterNode">BiquadFilterNode</dfn> : AudioNode {
attribute BiquadFilterType type;
readonly attribute AudioParam frequency; // in Hertz
readonly attribute AudioParam detune; // in Cents
readonly attribute AudioParam Q; // Quality factor
readonly attribute AudioParam gain; // in Decibels
void getFrequencyResponse(Float32Array frequencyHz,
Float32Array magResponse,
Float32Array phaseResponse);
};
</code></pre>
</div>
</div>
</div>
<p>The filter types are briefly described below. We note that all of these
filters are very commonly used in audio processing. In terms of implementation,
they have all been derived from standard analog filter prototypes. For more
technical details, we refer the reader to the excellent <a
href="http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt">reference</a> by
Robert Bristow-Johnson.</p>
<p>
All parameters are <em>k-rate</em> with the following default parameter values:
</p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>350Hz, with a nominal range of 10 to the Nyquist frequency (half the sample-rate).
</dd>
<dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt>
<dd>1, with a nominal range of 0.0001 to 1000.</dd>
<dt>gain</dt>
<dd>0, with a nominal range of -40 to 40.</dd>
</dl>
</blockquote>
<div id="BiquadFilterNode-description-section" class="section">
<h3 id="BiquadFilterNode-description">4.21.1 "lowpass"</h3>
<p>A <a href="http://en.wikipedia.org/wiki/Low-pass_filter">lowpass filter</a>
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.</p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>The cutoff frequency</dd>
<dt>Q</dt>
<dd>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.</dd>
<dt>gain</dt>
<dd>Not used in this filter type</dd>
</dl>
</blockquote>
<h3 id="HIGHPASS">4.21.2 "highpass"</h3>
<p>A <a href="http://en.wikipedia.org/wiki/High-pass_filter">highpass
filter</a> 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.</p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>The cutoff frequency below which the frequencies are attenuated</dd>
<dt>Q</dt>
<dd>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.</dd>
<dt>gain</dt>
<dd>Not used in this filter type</dd>
</dl>
</blockquote>
<h3 id="BANDPASS">4.21.3 "bandpass"</h3>
<p>A <a href="http://en.wikipedia.org/wiki/Band-pass_filter">bandpass
filter</a> 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.</p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>The center of the frequency band</dd>
<dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt>
<dd>Controls the width of the band. The width becomes narrower as the Q
value increases.</dd>
<dt>gain</dt>
<dd>Not used in this filter type</dd>
</dl>
</blockquote>
<h3 id="LOWSHELF">4.21.4 "lowshelf"</h3>
<p>The lowshelf filter allows all frequencies through, but adds a boost (or
attenuation) to the lower frequencies. It implements a second-order
lowshelf filter.</p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>The upper limit of the frequences where the boost (or attenuation) is
applied.</dd>
<dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt>
<dd>Not used in this filter type.</dd>
<dt>gain</dt>
<dd>The boost, in dB, to be applied. If the value is negative, the
frequencies are attenuated.</dd>
</dl>
</blockquote>
<h3 id="L16352">4.21.5 "highshelf"</h3>
<p>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</p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>The lower limit of the frequences where the boost (or attenuation) is
applied.</dd>
<dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt>
<dd>Not used in this filter type.</dd>
<dt>gain</dt>
<dd>The boost, in dB, to be applied. If the value is negative, the
frequencies are attenuated.</dd>
</dl>
</blockquote>
<h3 id="PEAKING">4.21.6 "peaking"</h3>
<p>The peaking filter allows all frequencies through, but adds a boost (or
attenuation) to a range of frequencies. </p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>The center frequency of where the boost is applied.</dd>
<dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt>
<dd>Controls the width of the band of frequencies that are boosted. A
large value implies a narrow width.</dd>
<dt>gain</dt>
<dd>The boost, in dB, to be applied. If the value is negative, the
frequencies are attenuated.</dd>
</dl>
</blockquote>
<h3 id="NOTCH">4.21.7 "notch"</h3>
<p>The notch filter (also known as a <a
href="http://en.wikipedia.org/wiki/Band-stop_filter">band-stop or
band-rejection filter</a>) is the opposite of a bandpass filter. It allows all
frequencies through, except for a set of frequencies.</p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>The center frequency of where the notch is applied.</dd>
<dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt>
<dd>Controls the width of the band of frequencies that are attenuated. A
large value implies a narrow width.</dd>
<dt>gain</dt>
<dd>Not used in this filter type.</dd>
</dl>
</blockquote>
<h3 id="ALLPASS">4.21.8 "allpass"</h3>
<p>An <a
href="http://en.wikipedia.org/wiki/All-pass_filter#Digital_Implementation">allpass
filter</a> allows all frequencies through, but changes the phase relationship
between the various frequencies. It implements a second-order allpass
filter</p>
<blockquote>
<dl>
<dt>frequency</dt>
<dd>The frequency where the center of the phase transition occurs. Viewed
another way, this is the frequency with maximal <a
href="http://en.wikipedia.org/wiki/Group_delay">group delay</a>.</dd>
<dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt>
<dd>Controls how sharp the phase transition is at the center frequency. A
larger value implies a sharper transition and a larger group delay.</dd>
<dt>gain</dt>
<dd>Not used in this filter type.</dd>
</dl>
</blockquote>
<h3 id="Methods">4.21.9. Methods</h3>
<dl>
<dt id="dfn-getFrequencyResponse">The <code>getFrequencyResponse</code>
method</dt>
<dd><p>Given the current filter parameter settings, calculates the
frequency response for the specified frequencies. </p>
<p>The <dfn id="dfn-frequencyHz">frequencyHz</dfn> parameter specifies an
array of frequencies at which the response values will be calculated.</p>
<p>The <dfn id="dfn-magResponse">magResponse</dfn> parameter specifies an
output array receiving the linear magnitude response values.</p>
<p>The <dfn id="dfn-phaseResponse">phaseResponse</dfn> parameter
specifies an output array receiving the phase response values in
radians.</p>
</dd>
</dl>
</div>
<div id="WaveShaperNode-section" class="section">
<h2 id="WaveShaperNode">4.22. The WaveShaperNode Interface</h2>
<p>WaveShaperNode is an AudioNode processor implementing non-linear distortion
effects. </p>
<p>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.</p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>
The number of channels of the output always equals the number of channels of the input.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="wave-shaper-node-idl">
enum <dfn>OverSampleType</dfn> {
"none",
"2x",
"4x"
};
interface <dfn id="dfn-WaveShaperNode">WaveShaperNode</dfn> : AudioNode {
attribute Float32Array? curve;
attribute OverSampleType oversample;
};
</code></pre>
</div>
</div>
<div id="attributes-WaveShaperNode-section" class="section">
<h3 id="attributes-WaveShaperNode">4.22.1. Attributes</h3>
<dl>
<dt id="dfn-curve"><code>curve</code></dt>
<dd><p>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. 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.</p>
</dd>
</dl>
<dl>
<dt id="dfn-oversample"><code>oversample</code></dt>
<dd><p>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.
</p>
<p>
A value of "2x" or "4x" means that the following steps must be performed:
<ol>
<li>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.</li>
<li>Apply the shaping curve.</li>
<li>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.
</ol>
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.
</p>
</dd>
</dl>
</div>
</div>
<div id="OscillatorNode-section" class="section">
<h2 id="OscillatorNode">4.23. The OscillatorNode Interface</h2>
<p>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 <a href="#PeriodicWave-section"><code>PeriodicWave</code></a> object. </p>
<p>Oscillators are common foundational building blocks in audio synthesis. An OscillatorNode will start emitting sound at the time
specified by the <code>start()</code> method. </p>
<p>
Mathematically speaking, a <em>continuous-time</em> 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 <em>Nyquist</em> frequency (half the sample-rate)
before converting the waveform to a digital form. If this is not done, then <em>aliasing</em> 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.
</p>
<p>
There are several practical approaches that an implementation may take to avoid this aliasing.
But regardless of approach, the <em>idealized</em> 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.
</p>
<p>
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.
</p>
<p>
Both .frequency and .detune are <em>a-rate</em> parameters and are used together to determine a <em>computedFrequency</em> value:
</p>
<pre>
computedFrequency(t) = frequency(t) * pow(2, detune(t) / 1200)
</pre>
<p>
The OscillatorNode's instantaneous phase at each time is the time integral of <em>computedFrequency</em>.
</p>
<pre> numberOfInputs : 0
numberOfOutputs : 1 (mono output)
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="oscillator-node-idl">
enum <dfn>OscillatorType</dfn> {
"sine",
"square",
"sawtooth",
"triangle",
"custom"
};
interface <dfn id="dfn-OscillatorNode">OscillatorNode</dfn> : AudioNode {
attribute OscillatorType type;
readonly attribute AudioParam frequency; // in Hertz
readonly attribute AudioParam detune; // in Cents
void start(double when);
void stop(double when);
void setPeriodicWave(PeriodicWave periodicWave);
attribute EventHandler onended;
};
</code></pre>
</div>
</div>
<div id="attributes-OscillatorNode-section" class="section">
<h3 id="attributes-OscillatorNode">4.23.1. Attributes</h3>
<dl>
<dt id="dfn-type"><code>type</code></dt>
<dd><p>The shape of the periodic waveform. It may directly be set to any of the type constant values except for "custom".
The <a href="#dfn-setPeriodicWave"><code>setPeriodicWave()</code></a> method can be used to set a custom waveform, which results in this attribute
being set to "custom". The default value is "sine". </p>
</dd>
</dl>
<dl>
<dt id="dfn-frequency"><code>frequency</code></dt>
<dd><p>The frequency (in Hertz) of the periodic waveform. This parameter is <em>a-rate</em> </p>
</dd>
</dl>
<dl>
<dt id="dfn-detune"><code>detune</code></dt>
<dd><p>A detuning value (in Cents) which will offset the <code>frequency</code> by the given amount.
This parameter is <em>a-rate</em> </p>
</dd>
</dl>
<dl>
<dt id="dfn-onended"><code>onended</code></dt>
<dd><p>A property used to set the <code>EventHandler</code> (described in <cite><a
href="http://www.whatwg.org/specs/web-apps/current-work/#eventhandler">HTML</a></cite>)
for the ended event that is dispatched to <a
href="#OscillatorNode-section"><code>OscillatorNode</code></a>
node types. When the playback of the buffer for an <code>OscillatorNode</code>
is finished, an event of type <code>Event</code> (described in <cite><a
href="http://www.whatwg.org/specs/web-apps/current-work/#event">HTML</a></cite>)
will be dispatched to the event handler. </p>
</dd>
</dl>
</div>
</div>
<div id="methodsandparams-OscillatorNode-section" class="section">
<h3 id="methodsandparams-OscillatorNode">4.23.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-setPeriodicWave">The <code>setPeriodicWave</code>
method</dt>
<dd><p>Sets an arbitrary custom periodic waveform given a <a href="#PeriodicWave-section"><code>PeriodicWave</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-start-AudioBufferSourceNode">The <code>start</code>
method</dt>
<dd><p>defined as in <a href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>. </p>
</dd>
</dl>
<dl>
<dt id="dfn-stop-AudioBufferSourceNode">The <code>stop</code>
method</dt>
<dd><p>defined as in <a href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>. </p>
</dd>
</dl>
</div>
<div id="PeriodicWave-section" class="section">
<h2 id="PeriodicWave">4.24. The PeriodicWave Interface</h2>
<p>PeriodicWave represents an arbitrary periodic waveform to be used with an <a href="#OscillatorNode-section"><code>OscillatorNode</code></a>.
Please see <a href="#dfn-createPeriodicWave">createPeriodicWave()</a> and <a href="#dfn-setPeriodicWave">setPeriodicWave()</a> and for more details. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="wavetable-idl">
interface <dfn id="dfn-PeriodicWave">PeriodicWave</dfn> {
};
</code></pre>
</div>
</div>
</div>
<div id="MediaStreamAudioSourceNode-section" class="section">
<h2 id="MediaStreamAudioSourceNode">4.25. The MediaStreamAudioSourceNode
Interface</h2>
<p>This interface represents an audio source from a <code>MediaStream</code>.
The first <code>AudioMediaStreamTrack</code> from the <code>MediaStream</code> will be
used as a source of audio.</p>
<pre> numberOfInputs : 0
numberOfOutputs : 1
</pre>
<p>
The number of channels of the output corresponds to the number of channels of the <code>AudioMediaStreamTrack</code>.
If there is no valid audio track, then the number of channels output will be one silent channel.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="media-stream-audio-source-node-idl">
interface <dfn id="dfn-MediaStreamAudioSourceNode">MediaStreamAudioSourceNode</dfn> : AudioNode {
};
</code></pre>
</div>
</div>
</div>
<div id="MediaStreamAudioDestinationNode-section" class="section">
<h2 id="MediaStreamAudioDestinationNode">4.26. The MediaStreamAudioDestinationNode
Interface</h2>
<p>This interface is an audio destination representing a <code>MediaStream</code> with a single <code>AudioMediaStreamTrack</code>.
This MediaStream is created when the node is created and is accessible via the <dfn>stream</dfn> 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 addStream() method.
</p>
<pre>
numberOfInputs : 1
numberOfOutputs : 0
channelCount = 2;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<p>
The number of channels of the input is by default 2 (stereo). Any connections to the input
are up-mixed/down-mixed to the number of channels of the input.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="media-stream-audio-destination-node-idl">
interface <dfn id="dfn-MediaStreamAudioDestinationNode">MediaStreamAudioDestinationNode</dfn> : AudioNode {
readonly attribute MediaStream stream;
};
</code></pre>
</div>
</div>
<div id="attributes-MediaStreamAudioDestinationNode-section" class="section">
<h3 id="attributes-MediaStreamAudioDestinationNode">4.26.1. Attributes</h3>
<dl>
<dt id="dfn-stream"><code>stream</code></dt>
<dd><p>A MediaStream containing a single AudioMediaStreamTrack with the same number of channels
as the node itself.</p>
</dd>
</dl>
</div>
</div>
<div id="MixerGainStructure-section" class="section">
<h2 id="MixerGainStructure">6. Mixer Gain Structure</h2>
<p class="norm">This section is informative.</p>
<h3 id="background">Background</h3>
<p>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. </p>
<div id="SummingJunction-section" class="section">
<h3 id="SummingJunction">Summing Inputs</h3>
</div>
<p>The inputs to <a href="#AudioNode-section"><code>AudioNodes</code></a> 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: </p>
<img alt="unity gain summing junction"
src="images/unity-gain-summing-junction.png" />
<p>In cases where the channel layouts of the outputs do not match, a mix (usually up-mix) will occur according to the <a
href="#UpMix-section">mixing rules</a>.
</p>
<h3 id="gain-Control">Gain Control</h3>
<p>But many times, it's important to be able to control the gain for each of
the output signals. The <a
href="#GainNode-section"><code>GainNode</code></a> gives this
control: </p>
<img alt="mixer architecture new" src="images/mixer-architecture-new.png" />
<p>Using these two concepts of unity gain summing junctions and GainNodes,
it's possible to construct simple or complex mixing scenarios. </p>
<h3 id="Example-mixer-with-send-busses">Example: Mixer with Send Busses</h3>
<p>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. </p>
<p>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:
</p>
<img alt="mixer gain structure" src="images/mixer-gain-structure.png" />
<p>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 <a href="#dfn-GainNode"><code>GainNode</code></a> is used:
</p>
<p>Here's how the above diagram could be constructed in JavaScript: </p>
<div class="example">
<div class="exampleHeader">
Example</div>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span></div>
<div class="blockContent">
<pre class="code"><code class="es-code">
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;
<span class="comment">// Setup routing graph </span>
function setupRoutingGraph() {
context = new AudioContext();
compressor = context.createDynamicsCompressor();
<span class="comment">// Send1 effect </span>
reverb = context.createConvolver();
<span class="comment">// Convolver impulse response may be set here or later </span>
<span class="comment">// Send2 effect </span>
delay = context.createDelay();
<span class="comment">// Connect final compressor to final destination </span>
compressor.connect(context.destination);
<span class="comment">// Connect sends 1 & 2 through effects to main mixer </span>
s1 = context.createGain();
reverb.connect(s1);
s1.connect(compressor);
s2 = context.createGain();
delay.connect(s2);
s2.connect(compressor);
<span class="comment">// Create a couple of sources </span>
source1 = context.createBufferSource();
source2 = context.createBufferSource();
source1.buffer = manTalkingBuffer;
source2.buffer = footstepsBuffer;
<span class="comment">// Connect source1 </span>
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);
<span class="comment">// Connect source2 </span>
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);
<span class="comment">// We now have explicit control over all the volumes g1_1, g2_1, ..., s1, s2 </span>
g2_1.gain.value = 0.2; <span class="comment"> // For example, set source1 reverb gain </span>
<span class="comment"> // Because g2_1.gain is an "AudioParam", </span>
<span class="comment"> // an automation curve could also be attached to it. </span>
<span class="comment"> // A "mixing board" UI could be created in canvas or WebGL controlling these gains. </span>
}
</code></pre>
</div>
</div>
</div>
</div>
<br />
<div id="DynamicLifetime-section">
<h2 id="DynamicLifetime">7. Dynamic Lifetime</h2>
<h3 id="DynamicLifetime-background">Background</h3>
<p class="norm">This section is informative. Please see <a href="#lifetime-AudioContext">AudioContext lifetime</a>
and <a href="#lifetime-AudioNode">AudioNode lifetime</a> for normative requirements
</p>
<p>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. </p>
<p>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. </p>
<p>The audio system automatically deals with tearing-down the part of the
routing graph for individual "note" events. A "note" is represented by an
<code>AudioBufferSourceNode</code>, which can be directly connected to other
processing nodes. When the note has finished playing, the context will
automatically release the reference to the <code>AudioBufferSourceNode</code>,
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. </p>
<h3 id="Example-DynamicLifetime">Example</h3>
<div class="example">
<div class="exampleHeader">
Example</div>
<img alt="dynamic allocation" src="images/dynamic-allocation.png" />
<p>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: </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span></div>
<div class="blockContent">
<pre class="code"><code class="es-code">
var context = 0;
var compressor = 0;
var gainNode1 = 0;
var streamingAudioSource = 0;
<span class="comment">// Initial setup of the "long-lived" part of the routing graph </span>
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);
}
<span class="comment">// Later in response to some user action (typically mouse or key event) </span>
<span class="comment">// a one-shot sound can be played. </span>
function playSound() {
var oneShotSound = context.createBufferSource();
oneShotSound.buffer = dogBarkingBuffer;
<span class="comment">// Create a filter, panner, and gain node. </span>
var lowpass = context.createBiquadFilter();
var panner = context.createPanner();
var gainNode2 = context.createGain();
<span class="comment">// Make connections </span>
oneShotSound.connect(lowpass);
lowpass.connect(panner);
panner.connect(gainNode2);
gainNode2.connect(compressor);
<span class="comment">// Play 0.75 seconds from now (to play immediately pass in 0)</span>
oneShotSound.start(context.currentTime + 0.75);
}
</code></pre>
</div>
</div>
</div>
</div>
<div id="UpMix-section" class="section">
<h2 id="UpMix">9. Channel up-mixing and down-mixing</h2>
<p class="norm">This section is normative.</p>
<img src="images/unity-gain-summing-junction.png">
<p>
<a href="#MixerGainStructure-section">Mixer Gain Structure</a>
describes how an <dfn>input</dfn> to an AudioNode can be connected from one or more <dfn>outputs</dfn>
of an AudioNode. Each of these connections from an output represents a stream with
a specific non-zero number of channels. An input has <em>mixing rules</em> 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 <em>mixing rules</em> 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. But the rules can be changed for advanced use cases, especially
multi-channel.
</p>
<p>
To define some terms, <em>up-mixing</em> 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. <em>down-mixing</em>
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.
</p>
<p>
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 <dfn>computedNumberOfChannels</dfn>
representing the actual number of channels of the input at any given time:
</p>
<p>
The AudioNode attributes involved in channel up-mixing and down-mixing rules are defined
<a href="#attributes-AudioNode-section">above</a>. The following is a more precise specification
on what each of them mean.
</p>
<ul>
<li><dfn>channelCount</dfn> is used to help compute <dfn>computedNumberOfChannels</dfn>.</li>
<li><dfn>channelCountMode</dfn> determines how <dfn>computedNumberOfChannels</dfn> will be computed.
Once this number is computed, all of the connections will be up or down-mixed to that many channels. For most nodes,
the default value is "max".
<ul>
<li>“max”: <dfn>computedNumberOfChannels</dfn> is computed as the maximum of the number of channels of all connections.
In this mode <dfn>channelCount</dfn> is ignored.</li>
<li>“clamped-max”: same as “max” up to a limit of the <dfn>channelCount</dfn></li>
<li>“explicit”: <dfn>computedNumberOfChannels</dfn> is the exact value as specified in <dfn>channelCount</dfn></li>
</ul>
</li>
<li><dfn>channelInterpretation</dfn> determines how the individual channels will be treated.
For example, will they be treated as speakers having a specific layout, or will they
be treated as simple discrete channels? This value influences exactly how the up and down mixing is
performed. The default value is "speakers".
<ul>
<li>“speakers”: use <a href="#ChannelLayouts">up-down-mix equations for mono/stereo/quad/5.1</a>.
In cases where the number of channels do not match any of these basic speaker layouts, revert
to "discrete".
</li>
<li>“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</li>
</ul>
</li>
</ul>
<p>
For each input of an AudioNode, an implementation must:
</p>
<ol>
<li>Compute <dfn>computedNumberOfChannels</dfn>.</li>
<li>For each connection to the input:
<ul>
<li> up-mix or down-mix the connection to <dfn>computedNumberOfChannels</dfn> according to <dfn>channelInterpretation</dfn>.</li>
<li> 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.</li>
</ul>
</li>
</ol>
<div id="ChannelLayouts-section" class="section">
<h3 id="ChannelLayouts">9.1. Speaker Channel Layouts</h3>
<p class="norm">This section is normative.</p>
<p>
When <dfn>channelInterpretation</dfn> is "speakers" then the up-mixing and down-mixing
is defined for specific channel layouts.
</p>
<p>It's important to define the channel ordering (and define some
abbreviations) for these speaker layouts.</p>
<p>
For now, only considers cases for mono, stereo, quad, 5.1. Later other channel
layouts can be defined.
</p>
<h4 id ="ChannelOrdering">9.1.1. Channel ordering</h4>
<pre> Mono
0: M: mono
Stereo
0: L: left
1: R: right
</pre>
<pre> 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
</pre>
</div>
<h4 id="UpMix-sub">9.1.2. Up Mixing speaker layouts</h4>
<pre>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 stereo 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;</pre>
<h4 id="down-mix">9.1.3. Down Mixing speaker layouts</h4>
<p>A down-mix will be necessary, for example, if processing 5.1 source
material, but playing back stereo. </p>
<pre>
Mono down-mix:
2 -> 1 : stereo to mono
output = 0.5 * (input.L + input.R);
4 -> 1 : quad to mono
output = 0.25 * (input.L + input.R + input.SL + input.SR);
5.1 -> 1 : 5.1 to mono
output = 0.7071 * (input.L + input.R) + input.C + 0.5 * (input.SL + input.SR)
Stereo down-mix:
4 -> 2 : quad to stereo
output.L = 0.5 * (input.L + input.SL);
output.R = 0.5 * (input.R + input.SR);
5.1 -> 2 : 5.1 to stereo
output.L = L + 0.7071 * (input.C + input.SL)
output.R = R + 0.7071 * (input.C + input.SR)
Quad down-mix:
5.1 -> 4 : 5.1 to quad
output.L = L + 0.7071 * input.C
output.R = R + 0.7071 * input.C
output.SL = input.SL
output.SR = input.SR
</pre>
</div>
<h3 id="ChannelRules-section">9.2. Channel Rules Examples</h3>
<p class="norm">This section is informative.</p>
<div class="block">
<div class="blockTitleDiv">
<div class="blockContent">
<pre class="code"><code class="idl-code">
// 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";
</code></pre>
</div>
</div>
</div>
<div id="Spatialization-section" class="section">
<h2 id="Spatialization">11. Spatialization / Panning </h2>
<h3 id="Spatialization-background">Background</h3>
<p>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. </p>
<p>Using an <code>PannerNode</code>, an audio stream can be spatialized or
positioned in space relative to an <code>AudioListener</code>. An <a
href="#AudioContext-section"><code>AudioContext</code></a> will contain a
single <code>AudioListener</code>. 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. <code>PannerNode</code>
objects (representing the source stream) have an <code>orientation</code>
vector representing in which direction the sound is projecting. Additionally,
they have a <code>sound cone</code> 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. <code>AudioListener</code> objects
(representing a person's ears) have an <code>orientation</code> and
<code>up</code> vector representing in which direction the person is facing.
Because both the source stream and the listener can be moving, they both have a
<code>velocity</code> 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. </p>
<p>
During rendering, the <code>PannerNode</code> calculates an <em>azimuth</em>
and <em>elevation</em>. These values are used internally by the implementation in
order to render the spatialization effect. See the <a href="#Spatialization-panning-algorithm">Panning Algorithm</a> section
for details of how these values are used.
</p>
<p>
The following algorithm must be used to calculate the <em>azimuth</em>
and <em>elevation</em>:
</p>
<div class="block">
<div class="blockTitleDiv">
<div class="blockContent">
<pre class="code"><code class="es-code">
// 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.
double 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;
</code></pre>
</div>
</div>
</div>
<h3 id="Spatialization-panning-algorithm">Panning Algorithm</h3>
<p>
<em>mono->stereo</em> and <em>stereo->stereo</em> panning must be supported.
<em>mono->stereo</em> processing is used when all connections to the input are mono.
Otherwise <em>stereo->stereo</em> processing is used.</p>
<p>The following algorithms must be implemented: </p>
<ul>
<li>Equal-power (Vector-based) panning
<p>This is a simple and relatively inexpensive algorithm which provides
basic, but reasonable results. It is commonly used when panning musical sources.
</p>
The <em>elevation</em> value is ignored in this panning algorithm.
<p>
The following steps are used for processing:
</p>
<ol>
<li>
<p>
The <em>azimuth</em> value is first contained to be within the range -90 <= <em>azimuth</em> <= +90 according to:
</p>
<pre>
// Clamp azimuth to allowed range of -180 -> +180.
azimuth = max(-180, azimuth);
azimuth = min(180, azimuth);
// Now wrap to range -90 -> +90.
if (azimuth < -90)
azimuth = -180 - azimuth;
else if (azimuth > 90)
azimuth = 180 - azimuth;
</pre>
</li>
<li>
<p>
A 0 -> 1 normalized value <em>x</em> is calculated from <em>azimuth</em> for <em>mono->stereo</em> as:
</p>
<pre>
x = (azimuth + 90) / 180
</pre>
<p>
Or for <em>stereo->stereo</em> as:
</p>
<pre>
if (azimuth <= 0) { // from -90 -> 0
// inputL -> outputL and "equal-power pan" inputR as in mono case
// by transforming the "azimuth" value from -90 -> 0 degrees into the range -90 -> +90.
x = (azimuth + 90) / 90;
} else { // from 0 -> +90
// inputR -> outputR and "equal-power pan" inputL as in mono case
// by transforming the "azimuth" value from 0 -> +90 degrees into the range -90 -> +90.
x = azimuth / 90;
}
</pre>
</li>
<li>
<p>
Left and right gain values are then calculated:
</p>
<pre>
gainL = cos(0.5 * PI * x);
gainR = sin(0.5 * PI * x);
</pre>
</li>
<li>
<p>For <em>mono->stereo</em>, the output is calculated as:</p>
<pre>
outputL = input * gainL
outputR = input * gainR
</pre>
<p>Else for <em>stereo->stereo</em>, the output is calculated as:</p>
<pre>
if (azimuth <= 0) { // from -90 -> 0
outputL = inputL + inputR * gainL;
outputR = inputR * gainR;
} else { // from 0 -> +90
outputL = inputL * gainL;
outputR = inputR + inputL * gainR;
}
</pre>
</li>
</ol>
</li>
<li><a
href="http://en.wikipedia.org/wiki/Head-related_transfer_function">HRTF</a>
panning (stereo only)
<p>This requires a set of HRTF impulse responses recorded at a variety of
azimuths and elevations. There are a small number of open/free impulse
responses available. The implementation requires a highly optimized
convolution function. It is somewhat more costly than "equal-power", but
provides a more spatialized sound. </p>
<img alt="HRTF panner" src="images/HRTF_panner.png" /></li>
</ul>
<h3 id="Spatialization-distance-effects">Distance Effects</h3>
<p>
Sounds which are closer are louder, while sounds further away are quieter.
Exactly <em>how</em> a sound's volume changes according to distance from the listener
depends on the <em>distanceModel</em> attribute.
</p>
<p>
During audio rendering, a <em>distance</em> value will be calculated based on the panner and listener positions according to:
</p>
<pre>
v = panner.position - listener.position
</pre>
<pre>
distance = sqrt(dot(v, v))
</pre>
<p>
<em>distance</em> will then be used to calculate <em>distanceGain</em> which depends
on the <em>distanceModel</em> attribute. See the <a href="#dfn-distanceModel">distanceModel</a> section for details of
how this is calculated for each distance model.
</p>
<p>As part of its processing, the <code>PannerNode</code> scales/multiplies the input audio signal by <em>distanceGain</em>
to make distant sounds quieter and nearer ones louder.
</p>
<h3 id="Spatialization-sound-cones">Sound Cones</h3>
<p>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. </p>
<p>
The following algorithm must be used to calculate the gain contribution due
to the cone effect, given the source (the <code>PannerNode</code>) and the listener:
</p>
<div class="block">
<div class="blockTitleDiv">
<div class="blockContent">
<pre class="code"><code class="idl-code">
if (source.orientation.isZero() || ((source.coneInnerAngle == 360) && (source.coneOuterAngle == 360)))
return 1; // no cone specified - unity gain
// Normalized source-listener vector
vec3 sourceToListener = listener.position - source.position;
sourceToListener.normalize();
vec3 normalizedSourceOrientation = source.orientation;
normalizedSourceOrientation.normalize();
// Angle between the source orientation vector and the source-listener vector
double dotProduct = sourceToListener.dot(normalizedSourceOrientation);
double angle = 180 * acos(dotProduct) / PI;
double absAngle = fabs(angle);
// Divide by 2 here since API is entire angle (not half-angle)
double absInnerAngle = fabs(source.coneInnerAngle) / 2;
double absOuterAngle = fabs(source.coneOuterAngle) / 2;
double 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
double x = (absAngle - absInnerAngle) / (absOuterAngle - absInnerAngle);
gain = (1 - x) + source.coneOuterGain * x;
}
return gain;
</code></pre>
</div>
</div>
</div>
<h3 id="Spatialization-doppler-shift">Doppler Shift</h3>
<ul>
<li>Introduces a pitch shift which can realistically simulate moving
sources.</li>
<li>Depends on: source / listener velocity vectors, speed of sound, doppler
factor.</li>
</ul>
<p>
The following algorithm must be used to calculate the doppler shift value which is used
as an additional playback rate scalar for all AudioBufferSourceNodes connecting directly or
indirectly to the AudioPannerNode:
</p>
<div class="block">
<div class="blockTitleDiv">
<div class="blockContent">
<pre class="code"><code class="idl-code">
double dopplerShift = 1; // Initialize to default value
double dopplerFactor = listener.dopplerFactor;
if (dopplerFactor > 0) {
double speedOfSound = listener.speedOfSound;
// Don't bother if both source and listener have no velocity.
if (!source.velocity.isZero() || !listener.velocity.isZero()) {
// Calculate the source to listener vector.
vec3 sourceToListener = source.position - listener.position;
double sourceListenerMagnitude = sourceToListener.length();
double listenerProjection = sourceToListener.dot(listener.velocity) / sourceListenerMagnitude;
double sourceProjection = sourceToListener.dot(source.velocity) / sourceListenerMagnitude;
listenerProjection = -listenerProjection;
sourceProjection = -sourceProjection;
double scaledSpeedOfSound = speedOfSound / dopplerFactor;
listenerProjection = min(listenerProjection, scaledSpeedOfSound);
sourceProjection = min(sourceProjection, scaledSpeedOfSound);
dopplerShift = ((speedOfSound - dopplerFactor * listenerProjection) / (speedOfSound - dopplerFactor * sourceProjection));
fixNANs(dopplerShift); // Avoid illegal values
// Limit the pitch shifting to 4 octaves up and 3 octaves down.
dopplerShift = min(dopplerShift, 16);
dopplerShift = max(dopplerShift, 0.125);
}
}
</code></pre>
</div>
</div>
</div>
</div>
<div id="Convolution-section" class="section">
<h2 id="Convolution">12. Linear Effects using Convolution</h2>
<h3 id="Convolution-background">Background</h3>
<p><a href="http://en.wikipedia.org/wiki/Convolution">Convolution</a> is a
mathematical process which can be applied to an audio signal to achieve many
interesting high-quality linear effects. Very often, the effect is used to
simulate an acoustic space such as a concert hall, cathedral, or outdoor
amphitheater. It can also be used for complex filter effects, like a muffled
sound coming from inside a closet, sound underwater, sound coming through a
telephone, or playing through a vintage speaker cabinet. This technique is very
commonly used in major motion picture and music production and is considered to
be extremely versatile and of high quality. </p>
<p>Each unique effect is defined by an <code>impulse response</code>. An
impulse response can be represented as an audio file and <a
href="#recording-impulse-responses">can be recorded</a> from a real acoustic
space such as a cave, or can be synthetically generated through a great variety
of techniques. </p>
<h3 id="Convolution-motivation">Motivation for use as a Standard</h3>
<p>A key feature of many game audio engines (OpenAL, FMOD, Creative's EAX,
Microsoft's XACT Audio, etc.) is a reverberation effect for simulating the
sound of being in an acoustic space. But the code used to generate the effect
has generally been custom and algorithmic (generally using a hand-tweaked set
of delay lines and allpass filters which feedback into each other). In nearly
all cases, not only is the implementation custom, but the code is proprietary
and closed-source, each company adding its own "black magic" to achieve its
unique quality. Each implementation being custom with a different set of
parameters makes it impossible to achieve a uniform desired effect. And the
code being proprietary makes it impossible to adopt a single one of the
implementations as a standard. Additionally, algorithmic reverberation effects
are limited to a relatively narrow range of different effects, regardless of
how the parameters are tweaked. </p>
<p>A convolution effect solves these problems by using a very precisely defined
mathematical algorithm as the basis of its processing. An impulse response
represents an exact sound effect to be applied to an audio stream and is easily
represented by an audio file which can be referenced by URL. The range of
possible effects is enormous. </p>
<h3 id="Convolution-implementation-guide">Implementation Guide</h3>
<p>
Linear convolution can be implemented efficiently.
Here are some <a href="https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/convolution.html">notes</a>
describing how it can be practically implemented.
</p>
<h3 id="Convolution-reverb-effect">Reverb Effect (with matrixing)</h3>
<p class="norm">This section is normative.</p>
<p>
In the general case the source
has N input channels, the impulse response has K channels, and the playback
system has M output channels. Thus it's a matter of how to matrix these
channels to achieve the final result.
</p>
<p>
The subset of N, M, K below must be implemented (note that the first image in the diagram is just illustrating
the general case and is not normative, while the following images are normative).
Without loss of generality, developers desiring more complex and arbitrary matrixing can use multiple <code>ConvolverNode</code>
objects in conjunction with an <code>ChannelMergerNode</code>.
</p>
<p>Single channel convolution operates on a mono audio input, using a mono
impulse response, and generating a mono output. But to achieve a more spacious sound, 2 channel audio
inputs and 1, 2, or 4 channel impulse responses will be considered. The following diagram, illustrates the
common cases for stereo playback where N and M are 1 or 2 and K is 1, 2, or 4.
</p>
<img alt="reverb matrixing" src="images/reverb-matrixing.png" />
<h3 id="recording-impulse-responses">Recording Impulse Responses</h3>
<p class="norm">This section is informative.</p>
<img alt="impulse response" src="images/impulse-response.png" /> <br />
<br />
<p>The most <a
href="http://pcfarina.eng.unipr.it/Public/Papers/226-AES122.pdf">modern</a> and
accurate way to record the impulse response of a real acoustic space is to use
a long exponential sine sweep. The test-tone can be as long as 20 or 30
seconds, or longer. <br />
Several recordings of the test tone played through a speaker can be made with
microphones placed and oriented at various positions in the room. It's
important to document speaker placement/orientation, the types of microphones,
their settings, placement, and orientations for each recording taken. </p>
<p>Post-processing is required for each of these recordings by performing an
inverse-convolution with the test tone, yielding the impulse response of the
room with the corresponding microphone placement. These impulse responses are
then ready to be loaded into the convolution reverb engine to re-create the
sound of being in the room. </p>
<h3 id="tools">Tools</h3>
<p>Two command-line tools have been written: <br />
<code>generate_testtones</code> generates an exponential sine-sweep test-tone
and its inverse. Another tool <code>convolve</code> was written for
post-processing. With these tools, anybody with recording equipment can record
their own impulse responses. To test the tools in practice, several recordings
were made in a warehouse space with interesting acoustics. These were later
post-processed with the command-line tools. </p>
<pre>% generate_testtones -h
Usage: generate_testtone
[-o /Path/To/File/To/Create] Two files will be created: .tone and .inverse
[-rate <sample rate>] sample rate of the generated test tones
[-duration <duration>] The duration, in seconds, of the generated files
[-min_freq <min_freq>] The minimum frequency, in hertz, for the sine sweep
% convolve -h
Usage: convolve input_file impulse_response_file output_file</pre>
<br />
<h3 id="recording-setup">Recording Setup</h3>
<img alt="recording setup" src="images/recording-setup.png" /> <br />
<br />
Audio Interface: Metric Halo Mobile I/O 2882 <br />
<br />
<br />
<br />
<img alt="microphones speaker" src="images/microphones-speaker.png" /> <br />
<br />
<img alt="microphone" src="images/microphone.png" /> <img alt="speaker"
src="images/speaker.png" /> <br />
<br />
Microphones: AKG 414s, Speaker: Mackie HR824 <br />
<br />
<br />
<h3 id="warehouse">The Warehouse Space</h3>
<img alt="warehouse" src="images/warehouse.png" /> <br />
<br />
</div>
<div id="JavaScriptProcessing-section" class="section">
<h2 id="JavaScriptProcessing">13. JavaScript Synthesis and Processing</h2>
<p class="norm">This section is informative.</p>
<p>The Mozilla project has conducted <a
href="https://wiki.mozilla.org/Audio_Data_API">Experiments</a> to synthesize
and process audio directly in JavaScript. This approach is interesting for a
certain class of audio processing and they have produced a number of impressive
demos. This specification includes a means of synthesizing and processing
directly using JavaScript by using a special subtype of <a
href="#AudioNode-section"><code>AudioNode</code></a> called <a
href="#ScriptProcessorNode-section"><code>ScriptProcessorNode</code></a>. </p>
<p>Here are some interesting examples where direct JavaScript processing can be
useful: </p>
<h3 id="custom-DSP-effects">Custom DSP Effects</h3>
<p>Unusual and interesting custom audio processing can be done directly in JS.
It's also a good test-bed for prototyping new algorithms. This is an extremely
rich area. </p>
<h3 id="educational-applications">Educational Applications</h3>
<p>JS processing is ideal for illustrating concepts in computer music synthesis
and processing, such as showing the de-composition of a square wave into its
harmonic components, FM synthesis techniques, etc. </p>
<h3 id="javaScript-performance">JavaScript Performance</h3>
<p>JavaScript has a variety of <a
href="#JavaScriptPerformance-section">performance issues</a> so it is not
suitable for all types of audio processing. The approach proposed in this
document includes the ability to perform computationally intensive aspects of
the audio processing (too expensive for JavaScript to compute in real-time)
such as multi-source 3D spatialization and convolution in optimized C++ code.
Both direct JavaScript processing and C++ optimized code can be combined due to
the APIs <a href="#ModularRouting-section">modular approach</a>. </p>
<div id="Performance-section" class="section">
<h2 id="Performance">15. Performance Considerations</h2>
<div id="Latency-section" class="section">
<h3 id="Latency">15.1. Latency: What it is and Why it's Important</h3>
</div>
<img alt="latency" src="images/latency.png" />
<p>For web applications, the time delay between mouse and keyboard events
(keydown, mousedown, etc.) and a sound being heard is important. </p>
<p>This time delay is called latency and is caused by several factors (input
device latency, internal buffering latency, DSP processing latency, output
device latency, distance of user's ears from speakers, etc.), and is
cummulative. The larger this latency is, the less satisfying the user's
experience is going to be. In the extreme, it can make musical production or
game-play impossible. At moderate levels it can affect timing and give the
impression of sounds lagging behind or the game being non-responsive. For
musical applications the timing problems affect rhythm. For gaming, the timing
problems affect precision of gameplay. For interactive applications, it
generally cheapens the users experience much in the same way that very low
animation frame-rates do. Depending on the application, a reasonable latency
can be from as low as 3-6 milliseconds to 25-50 milliseconds. </p>
<div id="Glitching-section" class="section">
<h3 id="audio-glitching">15.2. Audio Glitching</h3>
</div>
<p>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. </p>
<h3 id="hardware-scalability">15.3. Hardware Scalability</h3>
<p>The system should gracefully degrade to allow audio processing under
resource constrained conditions without dropping audio frames. </p>
<p>First of all, it should be clear that regardless of the platform, the audio
processing load should never be enough to completely lock up the machine.
Second, the audio rendering needs to produce a clean, un-interrupted audio
stream without audible <a href="#Glitching-section">glitches</a>. </p>
<p>The system should be able to run on a range of hardware, from mobile phones
and tablet devices to laptop and desktop computers. But the more limited
compute resources on a phone device make it necessary to consider techniques to
scale back and reduce the complexity of the audio rendering. For example,
voice-dropping algorithms can be implemented to reduce the total number of
notes playing at any given time. </p>
<p>Here's a list of some techniques which can be used to limit CPU usage: </p>
<h4 id="CPU-monitoring">15.3.1. CPU monitoring</h4>
<p>In order to avoid audio breakup, CPU usage must remain below 100%. </p>
<p>The relative CPU usage can be dynamically measured for each AudioNode (and
chains of connected nodes) as a percentage of the rendering time quantum. In a
single-threaded implementation, overall CPU usage must remain below 100%. The
measured usage may be used internally in the implementation for dynamic
adjustments to the rendering. It may also be exposed through a
<code>cpuUsage</code> attribute of <code>AudioNode</code> for use by
JavaScript. </p>
<p>In cases where the measured CPU usage is near 100% (or whatever threshold is
considered too high), then an attempt to add additional <code>AudioNodes</code>
into the rendering graph can trigger voice-dropping. </p>
<h4 id="Voice-dropping">15.3.2. Voice Dropping</h4>
<p>Voice-dropping is a technique which limits the number of voices (notes)
playing at the same time to keep CPU usage within a reasonable range. There can
either be an upper threshold on the total number of voices allowed at any given
time, or CPU usage can be dynamically monitored and voices dropped when CPU
usage exceeds a threshold. Or a combination of these two techniques can be
applied. When CPU usage is monitored for each voice, it can be measured all the
way from a source node through any effect processing nodes which apply
uniquely to that voice. </p>
<p>When a voice is "dropped", it needs to happen in such a way that it doesn't
introduce audible clicks or pops into the rendered audio stream. One way to
achieve this is to quickly fade-out the rendered audio for that voice before
completely removing it from the rendering graph. </p>
<p>When it is determined that one or more voices must be dropped, there are
various strategies for picking which voice(s) to drop out of the total ensemble
of voices currently playing. Here are some of the factors which can be used in
combination to help with this decision: </p>
<ul>
<li>Older voices, which have been playing the longest can be dropped instead
of more recent voices. </li>
<li>Quieter voices, which are contributing less to the overall mix may be
dropped instead of louder ones. </li>
<li>Voices which are consuming relatively more CPU resources may be dropped
instead of less "expensive" voices.</li>
<li>An AudioNode can have a <code>priority</code> attribute to help determine
the relative importance of the voices.</li>
</ul>
<h4 id="Simplification-of-Effects-Processing">15.3.3. Simplification of Effects
Processing</h4>
<p>Most of the effects described in this document are relatively inexpensive
and will likely be able to run even on the slower mobile devices. However, the
<a href="#ConvolverNode-section">convolution effect</a> can be configured with
a variety of impulse responses, some of which will likely be too heavy for
mobile devices. Generally speaking, CPU usage scales with the length of the
impulse response and the number of channels it has. Thus, it is reasonable to
consider that impulse responses which exceed a certain length will not be
allowed to run. The exact limit can be determined based on the speed of the
device. Instead of outright rejecting convolution with these long responses, it
may be interesting to consider truncating the impulse responses to the maximum
allowed length and/or reducing the number of channels of the impulse response.
</p>
<p>In addition to the convolution effect. The <a
href="#PannerNode-section"><code>PannerNode</code></a> may also be
expensive if using the HRTF panning model. For slower devices, a cheaper
algorithm such as EQUALPOWER can be used to conserve compute resources. </p>
<h4 id="Sample-rate">15.3.4. Sample Rate</h4>
<p>For very slow devices, it may be worth considering running the rendering at
a lower sample-rate than normal. For example, the sample-rate can be reduced
from 44.1KHz to 22.05KHz. This decision must be made when the
<code>AudioContext</code> is created, because changing the sample-rate
on-the-fly can be difficult to implement and will result in audible glitching
when the transition is made. </p>
<h4 id="pre-flighting">15.3.5. Pre-flighting</h4>
<p>It should be possible to invoke some kind of "pre-flighting" code (through
JavaScript) to roughly determine the power of the machine. The JavaScript code
can then use this information to scale back any more intensive processing it
may normally run on a more powerful machine. Also, the underlying
implementation may be able to factor in this information in the voice-dropping
algorithm. </p>
<p><span class="ednote">TODO: add specification and more detail here </span></p>
<h4 id="Authoring-for-different-user-agents">15.3.6. Authoring for different
user agents</h4>
JavaScript code can use information about user-agent to scale back any more
intensive processing it may normally run on a more powerful machine.
<h4 id="Scalability-of-Direct-JavaScript-Synthesis">15.3.7. Scalability of
Direct JavaScript Synthesis / Processing</h4>
<p>Any audio DSP / processing code done directly in JavaScript should also be
concerned about scalability. To the extent possible, the JavaScript code itself
needs to monitor CPU usage and scale back any more ambitious processing when
run on less powerful devices. If it's an "all or nothing" type of processing,
then user-agent check or pre-flighting should be done to avoid generating an
audio stream with audio breakup. </p>
<div id="JavaScriptPerformance-section" class="section">
<h3 id="JavaScriptPerformance">15.4. JavaScript Issues with real-time
Processing and Synthesis: </h3>
</div>
While processing audio in JavaScript, it is extremely challenging to get
reliable, glitch-free audio while achieving a reasonably low-latency,
especially under heavy processor load.
<ul>
<li>JavaScript is very much slower than heavily optimized C++ code and is not
able to take advantage of SSE optimizations and multi-threading which is
critical for getting good performance on today's processors. Optimized
native code can be on the order of twenty times faster for processing FFTs
as compared with JavaScript. It is not efficient enough for heavy-duty
processing of audio such as convolution and 3D spatialization of large
numbers of audio sources. </li>
<li>setInterval() and XHR handling will steal time from the audio processing.
In a reasonably complex game, some JavaScript resources will be needed for
game physics and graphics. This creates challenges because audio rendering
is deadline driven (to avoid glitches and get low enough latency).</li>
<li>JavaScript does not run in a real-time processing thread and thus can be
pre-empted by many other threads running on the system.</li>
<li>Garbage Collection (and autorelease pools on Mac OS X) can cause
unpredictable delay on a JavaScript thread. </li>
<li>Multiple JavaScript contexts can be running on the main thread, stealing
time from the context doing the processing. </li>
<li>Other code (other than JavaScript) such as page rendering runs on the
main thread. </li>
<li>Locks can be taken and memory is allocated on the JavaScript thread. This
can cause additional thread preemption. </li>
</ul>
The problems are even more difficult with today's generation of mobile devices
which have processors with relatively poor performance and power consumption /
battery-life issues. <br />
<br />
<div id="ExampleApplications-section" class="section">
<h2 id="ExampleApplications">16. Example Applications</h2>
<p class="norm">This section is informative.</p>
<p>Please see the <a
href="http://chromium.googlecode.com/svn/trunk/samples/audio/index.html">demo</a>
page for working examples. </p>
<p>Here are some of the types of applications a web audio system should be able
to support: </p>
<h3 id="basic-sound-playback">Basic Sound Playback</h3>
<p>Simple and <a href="#Latency-section"><strong>low-latency</strong></a>
playback of sound effects in response to simple user actions such as mouse
click, roll-over, key press. </p>
<br />
<h3 id="threeD-environmentse-and-games">3D Environments and Games</h3>
<img alt="quake" src="http://payload48.cargocollective.com/1/2/66805/3278334/redteam_680.jpg" />
<br />
<br />
<p>Electronic Arts has produced an impressive immersive game called
<a href="http://sophie-lu.com/Strike-Fortress-EA">Strike Fortress</a>,
taking advantage of 3D spatialization and convolution for room simulation.</p>
<img alt="beach demo" src="images/beach-demo.png" />
<p>3D environments with audio are common in games made for desktop applications
and game consoles. Imagine a 3D island environment with spatialized audio,
seagulls flying overhead, the waves crashing against the shore, the crackling
of the fire, the creaking of the bridge, and the rustling of the trees in the
wind. The sounds can be positioned naturally as one moves through the scene.
Even going underwater, low-pass filters can be tweaked for just the right
underwater sound. </p>
<br />
<br />
<img alt="box2d" src="images/box2d.png" /> <img alt="8-ball"
src="images/8-ball.png" /> <br />
<br />
<p><a href="http://box2d.org/">Box2D</a> is an interesting open-source
library for 2D game physics. It has various implementations, including one
based on Canvas 2D. A demo has been created with dynamic sound effects for each
of the object collisions, taking into account the velocities vectors and
positions to spatialize the sound events, and modulate audio effect parameters
such as filter cutoff. </p>
<p>A virtual pool game with multi-sampled sound effects has also been created.
</p>
<br />
<h3 id="musical-applications">Musical Applications</h3>
<img alt="garageband" src="images/garage-band.png" /> <img
alt="shiny drum machine" src="images/shiny-drum-machine.png" /> <img
alt="tonecraft" src="images/tonecraft.png" /> <br />
<br />
Many music composition and production applications are possible. Applications
requiring tight scheduling of audio events can be implemented and can be both
educational and entertaining. Drum machines, digital DJ applications, and even
timeline-based digital music production software with some of the features of
<a href="http://en.wikipedia.org/wiki/GarageBand">GarageBand</a> can be
written. <br />
<br />
<h3 id="music-visualizers">Music Visualizers</h3>
<img alt="music visualizer" src="images/music-visualizer.png" /> <br />
<br />
When combined with WebGL GLSL shaders, realtime analysis data can be presented
in entertaining ways. These can be as advanced as any found in iTunes. <br />
<br />
<h3 id="educational-applications_2">Educational Applications</h3>
<img alt="javascript processing" src="images/javascript-processing.png" />
<p>A variety of educational applications can be written, illustrating concepts
in music theory and computer music synthesis and processing. </p>
<br />
<h3 id="artistic-audio-exploration">Artistic Audio Exploration</h3>
<p>There are many creative possibilites for artistic sonic environments for
installation pieces. </p>
<br />
</div>
<div id="SecurityConsiderations-section" class="section">
<h2 id="SecurityConsiderations">17. Security Considerations</h2>
<p>This section is <em>informative.</em> </p>
</div>
<div id="PrivacyConsiderations-section" class="section">
<h2 id="PrivacyConsiderations">18. Privacy Considerations</h2>
<p>This section is <em>informative</em>. 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 RealtimeAnalyzerNode or
ScriptProcessorNode interface. The information could subsequently be used to
create a fingerprint of the client. </p>
<p>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
<a href="http://developers.whatwg.org/">getUserMedia()
API</a>. </p>
</div>
<div id="requirements-section" class="section">
<h2 id="requirements">19. Requirements and Use Cases</h2>
<p>Please see <a href="#ExampleApplications-section">Example Applications</a>
</p>
</div>
<div id="oldnames-section" class="section">
<h2 id="OldNames">20. Old Names</h2>
<p class="norm">This section is informative.</p>
<p>Some method and attribute names have been improved during API review.
The new names are described in the main body of this specification in the
description for each node type, etc. Here's a description of the older names
to help content authors migrate to the latest spec. Note that the partial
interfaces are not normative and are only descriptive:
</p>
<blockquote>
<pre>
partial interface <dfn>AudioBufferSourceNode</dfn> {
// Same as start()
void noteOn(double when);
void noteGrainOn(double when, double grainOffset, double grainDuration);
// Same as stop()
void noteOff(double when);
};
partial interface <dfn>AudioContext</dfn> {
// Same as createGain()
GainNode createGainNode();
// Same as createDelay()
DelayNode createDelayNode(optional double maxDelayTime = 1.0);
// Same as createScriptProcessor()
ScriptProcessorNode createJavaScriptNode(optional unsigned long bufferSize = 0,
optional unsigned long numberOfInputChannels = 2,
optional unsigned long numberOfOutputChannels = 2);
};
partial interface <dfn>OscillatorNode</dfn> {
// Same as start()
void noteOn(double when);
// Same as stop()
void noteOff(double when);
};
partial interface <dfn>AudioParam</dfn> {
// Same as setTargetAtTime()
void setTargetValueAtTime(float target, double startTime, double timeConstant);
};
</pre>
</blockquote>
<p>Some attributes taking constant values have changed during API review.
The old way used integer values, while the new way uses Web IDL string values.
</p>
<blockquote>
<pre>
// PannerNode constants for the .panningModel attribute
// Old way
const unsigned short EQUALPOWER = 0;
const unsigned short HRTF = 1;
// New way
enum <dfn>PanningModelType</dfn> {
"equalpower",
"HRTF"
};
</pre>
</blockquote>
<blockquote>
<pre>
// PannerNode constants for the .distanceModel attribute
// Old way
const unsigned short LINEAR_DISTANCE = 0;
const unsigned short INVERSE_DISTANCE = 1;
const unsigned short EXPONENTIAL_DISTANCE = 2;
// New way
enum <dfn>DistanceModelType</dfn> {
"linear",
"inverse",
"exponential"
};
</pre>
</blockquote>
<blockquote>
<pre>
// BiquadFilterNode constants for the .type attribute
// Old way
const unsigned short LOWPASS = 0;
const unsigned short HIGHPASS = 1;
const unsigned short BANDPASS = 2;
const unsigned short LOWSHELF = 3;
const unsigned short HIGHSHELF = 4;
const unsigned short PEAKING = 5;
const unsigned short NOTCH = 6;
const unsigned short ALLPASS = 7;
// New way
enum <dfn>BiquadFilterType</dfn> {
"lowpass",
"highpass",
"bandpass",
"lowshelf",
"highshelf",
"peaking",
"notch",
"allpass"
};
</pre>
</blockquote>
<blockquote>
<pre>
// OscillatorNode constants for the .type attribute
// Old way
const unsigned short SINE = 0;
const unsigned short SQUARE = 1;
const unsigned short SAWTOOTH = 2;
const unsigned short TRIANGLE = 3;
const unsigned short CUSTOM = 4;
// New way
enum <dfn>OscillatorType</dfn> {
"sine",
"square",
"sawtooth",
"triangle",
"custom"
};
</pre>
</blockquote>
</div>
</div>
</div>
<div class="appendix section" id="references">
<h2 id="L17310">A.References</h2>
<div class="section" id="normative-references">
<h3 id="Normative-references">A.1 Normative references</h3>
<dl>
<dt id="DOM">[DOM] </dt>
<dd><a href="http://dom.spec.whatwg.org/">DOM</a>,
A. van Kesteren, A. Gregor, Ms2ger. WHATWG.</dd>
<dt id="HTML">[HTML] </dt>
<dd><a href="http://www.whatwg.org/specs/web-apps/current-work/multipage/">HTML</a>,
I. Hickson. WHATWG.</dd>
<dt id="RFC2119">[RFC2119] </dt>
<dd>S. Bradner. <a
href="http://www.ietf.org/rfc/rfc2119.txt"><cite><span>Key words for use
in RFCs to Indicate Requirement Levels.</span></cite></a> Internet RFC
2119. URL: <a
href="http://www.ietf.org/rfc/rfc2119.txt">http://www.ietf.org/rfc/rfc2119.txt</a>
</dd>
</dl>
</div>
<div class="section" id="informative-references">
<h3 id="Informative-references">A.2 Informative references</h3>
<p>No informative references.</p>
</div>
</div>
<div class="section" id="acknowledgements">
<h2 id="L17335">B.Acknowledgements</h2>
<p>Special thanks to the W3C <a href="http://www.w3.org/2011/audio/">Audio
Working Group</a>. Members of the Working Group are (at the time of writing,
and by alphabetical order): <br />
Berkovitz, Joe (public Invited expert);Cardoso, Gabriel (INRIA);Carlson, Eric
(Apple, Inc.);Gregan, Matthew (Mozilla Foundation);Jägenstedt, Philip (Opera
Software);Kalliokoski, Jussi (public Invited expert);Lowis, Chris (British
Broadcasting Corporation);MacDonald, Alistair (W3C Invited Experts);Michel,
Thierry (W3C/ERCIM);Noble, Jer (Apple, Inc.);O'Callahan, Robert(Mozilla
Foundation);Paradis, Matthew (British Broadcasting Corporation);Raman, T.V.
(Google, Inc.);Rogers, Chris (Google, Inc.);Schepers, Doug (W3C/MIT);Shires,
Glen (Google, Inc.);Smith, Michael (W3C/Keio);Thereaux, Olivier (British
Broadcasting Corporation);Wei, James (Intel Corporation);Wilson, Chris (Google,
Inc.); </p>
</div>
<div class="section" id="ChangeLog-section">
<h2 id="ChangeLog">C. Web Audio API Change Log</h2>
<pre>
user: crogers
date: Sun Dec 09 17:13:56 2012 -0800
summary: Basic description of OfflineAudioContext
user: crogers
date: Tue Dec 04 15:59:30 2012 -0800
summary: minor correction to wording for minValue and maxValue
user: crogers
date: Tue Dec 04 15:49:29 2012 -0800
summary: Bug 20161: Make decodeAudioData neuter its array buffer argument when it begins decoding a buffer, and bring it back to normal when the decoding is finished
user: crogers
date: Tue Dec 04 15:35:17 2012 -0800
summary: Bug 20039: Refine description of audio decoding
user: crogers
date: Tue Dec 04 15:23:07 2012 -0800
summary: elaborate on decoding steps for AudioContext createBuffer() and decodeAudioData()
user: crogers
date: Tue Dec 04 14:56:19 2012 -0800
summary: Bug 19770: Note that if the last event for an AudioParam is a setCurveValue event, the computed value after that event will be equal to the latest curve value
user: crogers
date: Tue Dec 04 14:48:04 2012 -0800
summary: Bug 19769: Note that before the first automation event, the computed AudioParam value will be AudioParam.value
user: crogers
date: Tue Dec 04 14:40:51 2012 -0800
summary: Bug 19768: Explicitly mention that the initial value of AudioParam.value will be defaultValue
user: crogers
date: Tue Dec 04 14:35:59 2012 -0800
summary: Bug 19767: Explicitly mention that the 2nd component of AudioParam.computedValue will be 0 if there are no AudioNodes connected to it
user: crogers
date: Tue Dec 04 14:30:08 2012 -0800
summary: Bug 19764: Note in the spec that AudioParam.minValue/maxValue are merely informational
user: crogers
date: Mon Dec 03 18:03:13 2012 -0800
summary: Convert integer constants to Web IDL enum string constants
user: crogers
date: Mon Dec 03 15:19:22 2012 -0800
summary: Bug 17411: (AudioPannerNodeUnits): AudioPannerNode units are underspecified
user: Ehsan Akhgari (Mozilla)
date: Thu Nov 29 15:59:38 2012 -0500
summary: Change the Web IDL description of decodeAudioData arguments
user: crogers
date: Wed Nov 14 13:24:01 2012 -0800
summary: Bug 17393: (UseDoubles): float/double inconsistency
user: crogers
date: Wed Nov 14 13:16:57 2012 -0800
summary: Bug 17356: (AudioListenerOrientation): AudioListener.setOrientation vectors
user: crogers
date: Wed Nov 14 12:56:06 2012 -0800
summary: Bug 19957: PannerNode.coneGain is unused
user: crogers
date: Wed Nov 14 12:40:46 2012 -0800
summary: Bug 17412: AudioPannerNodeVectorNormalization): AudioPannerNode orientation normalization unspecified
user: crogers
date: Wed Nov 14 12:16:41 2012 -0800
summary: Bug 17411: (AudioPannerNodeUnits): AudioPannerNode units are underspecified
user: crogers
date: Tue Nov 13 16:14:22 2012 -0800
summary: be more explicit about maxDelayTime units
user: crogers
date: Tue Nov 13 16:02:50 2012 -0800
summary: Bug 19766: Clarify that reading AudioParam.computedValue will return the latest computed value for the latest audio quantum
user: crogers
date: Tue Nov 13 15:47:25 2012 -0800
summary: Bug 19872: Should specify the defaults for PannerNode's position, ...
user: crogers
date: Tue Nov 13 15:27:53 2012 -0800
summary: Bug 17390: (Joe Berkovitz): Loop start/stop points
user: croger
date: Tue Nov 13 14:49:20 2012 -0800
summary: Bug 19765: Note that setting AudioParam.value will be ignored when any automation events have been set on the object
user: crogers
date: Tue Nov 13 14:39:07 2012 -0800
summary: Bug 19873: Clarify PannerNode.listener
user: crogers
date: Tue Nov 13 13:35:21 2012 -0800
summary: Bug 19900: Clarify the default values for the AudioParam attributes of BiquadFilterNode
user: crogers
date: Tue Nov 13 13:06:38 2012 -0800
summary: Bug 19884: Specify the default value and ranges for the DynamicsCompressorNode AudioParam members
user: crogers
date: Tue Nov 13 12:57:02 2012 -0800
summary: Bug 19910: Disallow AudioContext.createDelay(max) where max <= 0
user: crogers
date: Mon Nov 12 12:02:18 2012 -0800
summary: Add example code for more complex example
user: Ehsan Akhgari (Mozilla)
date: Thu Nov 01 11:32:39 2012 -0400
summary: Specify the default value for the AudioContext.createDelay() optional argument in Web IDL
user: Ehsan Akhgari (Mozilla)
date: Tue Oct 30 20:29:48 2012 -0400
summary: Mark the AudioParam members as readonly
user: Ehsan Akhgari (Mozilla)
date: Tue Oct 30 20:24:52 2012 -0400
summary: Make GainNode and DelayNode valid Web IDL
user: crogers
date: Mon Oct 29 14:29:23 2012 -0700
summary: consolidate AudioBufferSourceNode start() method
user: crogers
date: Fri Oct 19 15:15:28 2012 -0700
summary: Bug 18332: Node creation method naming inconsistencies
user: crogers
date: Mon Oct 15 17:22:54 2012 -0700
summary: Bug 17407: Interface naming inconsistency
user: crogers
date: Tue Oct 09 17:21:19 2012 -0700
summary: Bug 17369: Oscillator.detune attribute not defined
user: crogers
date: Tue Oct 09 16:08:50 2012 -0700
summary: Bug 17346: HTMLMediaElement integration
user: crogers
date: Tue Oct 09 15:20:50 2012 -0700
summary: Bug 17354: AudioListener default position, orientation and velocity
user: crogers
date: Tue Oct 09 15:02:04 2012 -0700
summary: Bug 17795: Behavior of multiple connections to same node needs to be explicitly defined
user: crogers
date: Mon Oct 08 13:18:45 2012 -0700
summary: Add missing AudioContext.createWaveShaper() method
user: crogers
date: Fri Oct 05 18:13:44 2012 -0700
summary: Bug 17399: AudioParam sampling is undefined
user: crogers
date: Fri Oct 05 17:41:52 2012 -0700
summary: Bug 17386: Realtime Analysis empty section
user: crogers
date: Fri Oct 05 17:38:14 2012 -0700
summary: minor tweak to down-mix section
user: crogers
date: Fri Oct 05 17:35:05 2012 -0700
summary: Bug 17380: Channel down mixing incomplete
user: crogers
date: Fri Oct 05 15:40:57 2012 -0700
summary: Bug 17375: MixerGainStructure should be marked as informative
user: crogers
date: Fri Oct 05 14:29:20 2012 -0700
summary: Bug 17381: (EventScheduling): Event Scheduling ('Need more detail here')
user: crogers
date: Fri Oct 05 13:12:46 2012 -0700
summary: Fix 18663: Need a method to get a readonly reading of the combined value when using AudioParam automation curve
user: crogers
date: Fri Oct 05 12:48:36 2012 -0700
summary: Fix 18662: Setting audioparam value while there is an automation curve will cancel that automation curve and set value immediately
user: crogers
date: Fri Oct 05 12:26:28 2012 -0700
summary: Fix 18661: Use startTime / endTime parameter names for AudioParam automation methods
user: crogers
date: Wed Oct 03 12:26:39 2012 -0700
summary: Specify default value for .distanceModel
user: crogers
date: Tue Oct 02 12:33:36 2012 -0700
summary: Fix Issues 17338 and 17337: AudioGain interface is not needed (Part 2)
user: crogers
date: Tue Oct 02 12:28:55 2012 -0700
summary: Fix Issues 17338 and 17337: AudioGain interface is not needed
user: Ehsan Akhgari (Mozilla)
date: Wed Sep 26 18:22:36 2012 -0400
summary: Make AudioBufferSourceNode.buffer nullable
user: crogers
date: Tue Sep 25 12:56:14 2012 -0700
summary: noteOn/noteOff changed to start/stop -- added deprecation notes
user: Ehsan Akhgari (Mozilla)
date: Fri Aug 24 18:27:29 2012 -0400
summary: Make the AudioContext object have a constructor
user: Ehsan Akhgari (Mozilla)
date: Fri Aug 24 15:54:10 2012 -0400
summary: Denote IDL definitions as Web IDL
user: Ehsan Akhgari (Mozilla)
date: Fri Aug 24 15:04:37 2012 -0400
summary: Use `long` instead of `int`, since the int type doesn't exist in Web IDL
user: Ehsan Akhgari (Mozilla)
date: Fri Aug 24 15:02:43 2012 -0400
summary: Add a missing attribute keyword in AudioProcessingEvent
user: Ehsan Akhgari (Mozilla)
date: Tue Aug 21 15:36:48 2012 -0400
summary: Remove the 'raises' notation from the IDLs
user: crogers
date: Thu Aug 16 16:30:55 2012 -0700
summary: Issue 17398: Add more detailed information about how AudioParam value is calculated
user: crogers
date: Thu Aug 16 15:21:38 2012 -0700
summary: another try with the style sheet
user: crogers
date: Thu Aug 16 14:53:54 2012 -0700
summary: use local style sheet to avoid https errors
user: Ehsan Akhgari (Mozilla)
date: Wed Aug 15 23:05:49 2012 -0400
summary: Replace the white-space based indentation of Web IDL code with a CSS-based one
user: Ehsan Akhgari (Mozilla)
date: Wed Aug 15 22:56:03 2012 -0400
summary: Remove more useless trailing whitespaces
user: Ehsan Akhgari (Mozilla)
date: Wed Aug 15 22:47:21 2012 -0400
summary: Remove the optional 'in' keyword from the Web IDL method declarations
user: Ehsan Akhgari (Mozilla)
date: Wed Aug 15 22:42:03 2012 -0400
summary: Add trailing semicolons for Web IDL interface declarations
user: Ehsan Akhgari (Mozilla)
date: Wed Aug 15 22:37:32 2012 -0400
summary: Remove useless trailing spaces
user: Ehsan Akhgari (Mozilla)
date: Wed Aug 15 22:35:33 2012 -0400
summary: Use the correct Web IDL notation for the AudioBufferCallback callback type
user: Ehsan Akhgari (Mozilla)
date: Wed Aug 15 22:28:37 2012 -0400
summary: Remove the extra semicolon in the IDL file for AudioContext
user: Ehsan Akhgari (Mozilla)
date: Wed Aug 15 22:24:02 2012 -0400
summary: Replace the old [Optional] IDL tag with the Web IDL optional keyword
user: Ehsan Akhgari (Mozilla)
date: Tue Aug 14 10:18:19 2012 -0400
summary: Empty changeset to test my commit access
date: Mon Aug 13 13:26:52 2012 -0700
* Integrate Thierry Michel's 3rd public working draft edits
date: Tue Jun 26 15:56:31 2012 -0700
* add MediaStreamAudioSourceNode
date: Mon Jun 18 13:26:21 2012 -0700
* minor formatting fix
date: Mon Jun 18 13:19:34 2012 -0700
* Add details for azimuth/elevation calculation
date: Fri Jun 15 17:35:27 2012 -0700
* Add equal-power-panning details
date: Thu Jun 14 17:31:16 2012 -0700
* Add equations for distance models
date: Wed Jun 13 17:40:49 2012 -0700
* Bug 17334: Add precise equations for AudioParam.setTargetValueAtTime()
date: Fri Jun 08 17:44:26 2012 -0700
* fix small typo
date: Fri Jun 08 16:54:04 2012 -0700
* Bug 17413: AudioBuffers' relationship to AudioContext
date: Fri Jun 08 16:05:45 2012 -0700
* Bug 17359: Add much more detail about ConvolverNode
date: Fri Jun 08 12:59:29 2012 -0700
* minor formatting fix
date: Fri Jun 08 12:57:11 2012 -0700
* Bug 17335: Add much more technical detail to setValueCurveAtTime()
date: Wed Jun 06 16:34:43 2012 -0700
*Add much more detail about parameter automation, including an example
date: Mon Jun 04 17:25:08 2012 -0700
* ISSUE-85: OscillatorNode folding considerations
date: Mon Jun 04 17:02:20 2012 -0700
* ISSUE-45: AudioGain scale underdefined
date: Mon Jun 04 16:40:43 2012 -0700
* ISSUE-41: AudioNode as input to AudioParam underdefined
date: Mon Jun 04 16:14:48 2012 -0700
* ISSUE-20: Relationship to currentTime
date: Mon Jun 04 15:48:49 2012 -0700
* ISSUE-94: Dynamic Lifetime
date: Mon Jun 04 13:59:31 2012 -0700
* ISSUE-42: add more detail about AudioParam sampling and block processing
date: Mon Jun 04 12:28:48 2012 -0700
* fix typo - minor edits
date: Thu May 24 18:01:20 2012 -0700
* ISSUE-69: add implementors guide for linear convolution
date: Thu May 24 17:35:45 2012 -0700
* ISSUE-49: better define AudioBuffer audio data access
date: Thu May 24 17:15:29 2012 -0700
* fix small typo
date: Thu May 24 17:13:34 2012 -0700
* ISSUE-24: define circular routing behavior
date: Thu May 24 16:35:24 2012 -0700
* ISSUE-42: specify a-rate or k-rate for each AudioParam
date: Fri May 18 17:01:36 2012 -0700
* ISSUE-53: noteOn and noteOff interaction
date: Fri May 18 16:33:29 2012 -0700
* ISSUE-34: Remove .name attribute from AudioParam
date: Fri May 18 16:27:19 2012 -0700
* ISSUE-33: Add maxNumberOfChannels attribute to AudioDestinationNode
date: Fri May 18 15:50:08 2012 -0700
* ISSUE-19: added more info about AudioBuffer - IEEE 32-bit
date: Fri May 18 15:37:27 2012 -0700
* ISSUE-29: remove reference to webkitAudioContext
date: Fri Apr 27 12:36:54 2012 -0700
* fix two small typos reported by James Wei
date: Tue Apr 24 12:27:11 2012 -0700
* small cleanup to ChannelSplitterNode and ChannelMergerNode
date: Tue Apr 17 11:35:56 2012 -0700
* small fix to createWaveTable()
date: Tue Apr 13 2012
* Cleanup AudioNode connect() and disconnect() method descriptions.
* Add AudioNode connect() to AudioParam method.
date: Tue Apr 13 2012
* Add OscillatorNode and WaveTable
* Define default values for optional arguments in createJavaScriptNode(), createChannelSplitter(), createChannelMerger()
* Define default filter type for BiquadFilterNode as LOWPASS
date: Tue Apr 11 2012
* add AudioContext .activeSourceCount attribute
* createBuffer() methods can throw exceptions
* add AudioContext method createMediaElementSource()
* update AudioContext methods createJavaScriptNode() (clean up description of parameters)
* update AudioContext method createChannelSplitter() (add numberOfOutputs parameter)
* update AudioContext method createChannelMerger() (add numberOfInputs parameter)
* update description of out-of-bounds AudioParam values (exception will not be thrown)
* remove AudioBuffer .gain attribute
* remove AudioBufferSourceNode .gain attribute
* remove AudioListener .gain attribute
* add AudioBufferSourceNode .playbackState attribute and state constants
* AnalyserNode no longer requires its output be connected to anything
* update ChannelMergerNode section describing numberOfOutputs (defaults to 6 but settable in constructor)
* update ChannelSplitterNode section describing numberOfInputs (defaults to 6 but settable in constructor)
* add note in Spatialization sections about potential to get arbitrary convolution matrixing
date: Tue Apr 10 2012
* Rebased editor's draft document based on edits from Thierry Michel (from 2nd public working draft).
date: Tue Mar 13 12:13:41 2012 -0100
* fixed all the HTML errors
* added ids to all Headings
* added alt attribute to all img
* fix broken anchors
* added a new status of this document section
* added mandatory spec headers
* generated a new table of content
* added a Reference section
* added an Acknowledgments section
* added a Web Audio API Change Log
date: Fri Mar 09 15:12:42 2012 -0800
* add optional maxDelayTime argument to createDelay()
* add more detail about playback state to AudioBufferSourceNode
* upgrade noteOn(), noteGrainOn(), noteOff() times to double from float
date: Mon Feb 06 16:52:39 2012 -0800
* Cleanup ScriptProcessorNode section
* Add distance model constants for PannerNode according to the OpenAL spec
* Add .normalize attribute to ConvolverNode
* Add getFrequencyResponse() method to BiquadFilterNode
* Tighten up the up-mix equations
date: Fri Nov 04 15:40:58 2011 -0700
summary: Add more technical detail to BiquadFilterNode description (contributed by Raymond Toy)
date: Sat Oct 15 19:08:15 2011 -0700
summary: small edits to the introduction
date: Sat Oct 15 19:00:15 2011 -0700
summary: initial commit
date: Tue Sep 13 12:49:11 2011 -0700
summary: add convolution reverb design document
date: Mon Aug 29 17:05:58 2011 -0700
summary: document the decodeAudioData() method
date: Mon Aug 22 14:36:33 2011 -0700
summary: fix broken MediaElementAudioSourceNode link
date: Mon Aug 22 14:33:57 2011 -0700
summary: refine section describing integration with HTMLMediaElement
date: Mon Aug 01 12:05:53 2011 -0700
summary: add Privacy section
date: Mon Jul 18 17:53:50 2011 -0700
summary: small update - tweak musical applications thumbnail images
date: Mon Jul 18 17:23:00 2011 -0700
summary: initial commit of Web Audio API specification</pre>
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