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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
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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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+ -->
+ <!--[if IE]>
+ <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 &lt;crogers@google.com&gt;</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 &lt;<a
+href="mailto:public-audio@w3.org">public-audio@w3.org</a>&gt; (<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 &lt;= t &lt; 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 &lt;= t &lt; 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 &lt;= t &lt; 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> &lt;= t &lt; <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> &lt;= t &lt; <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 &lt; 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 -&gt; +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) &amp;&amp; loopStart >= 0 &amp;&amp; loopEnd > 0 &amp;&amp; loopStart &lt; 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 &lt; 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 &lt; 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 -&gt; 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 -&gt; +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 &amp; 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 -&gt; 2 : up-mix from mono to stereo
+ output.L = input;
+ output.R = input;
+
+ 1 -&gt; 4 : up-mix from mono to quad
+ output.L = input;
+ output.R = input;
+ output.SL = 0;
+ output.SR = 0;
+
+ 1 -&gt; 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 -&gt; 4 : up-mix from stereo to quad
+ output.L = input.L;
+ output.R = input.R;
+ output.SL = 0;
+ output.SR = 0;
+
+ 2 -&gt; 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 -&gt; 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 -&gt; 1 : stereo to mono
+ output = 0.5 * (input.L + input.R);
+
+ 4 -&gt; 1 : quad to mono
+ output = 0.25 * (input.L + input.R + input.SL + input.SR);
+
+ 5.1 -&gt; 1 : 5.1 to mono
+ output = 0.7071 * (input.L + input.R) + input.C + 0.5 * (input.SL + input.SR)
+
+
+Stereo down-mix:
+
+ 4 -&gt; 2 : quad to stereo
+ output.L = 0.5 * (input.L + input.SL);
+ output.R = 0.5 * (input.R + input.SR);
+
+ 5.1 -&gt; 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 -&gt; 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 &lt; 0)
+ azimuth = 360 - azimuth;
+
+// Make azimuth relative to "front" and not "right" listener vector.
+if ((azimuth >= 0) &amp;&amp; (azimuth &lt;= 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 &lt; -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 &lt;= <em>azimuth</em> &lt;= +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 &lt; -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 &lt;= 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 &lt;= 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) &amp;&amp; (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 &lt;= absInnerAngle)
+ // No attenuation
+ gain = 1;
+else if (absAngle &gt;= 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 &lt;sample rate&gt;] sample rate of the generated test tones
+ [-duration &lt;duration&gt;] The duration, in seconds, of the generated files
+ [-min_freq &lt;min_freq&gt;] 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 &lt;= 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>
+</div>
+</body>
+</html>
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