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diff --git a/testing/web-platform/tests/webaudio/specification.html b/testing/web-platform/tests/webaudio/specification.html new file mode 100644 index 000000000..3178c5e10 --- /dev/null +++ b/testing/web-platform/tests/webaudio/specification.html @@ -0,0 +1,5911 @@ +<?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> + <script src="dfn.js" type="application/ecmascript"></script> + --> + <!--[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 <crogers@google.com></dd> +</dl> + +<p class="copyright"><a +href="http://www.w3.org/Consortium/Legal/ipr-notice#Copyright">Copyright</a> © +2012 <a href="http://www.w3.org/"><acronym +title="World Wide Web Consortium">W3C</acronym></a><sup>®</sup> (<a +href="http://www.csail.mit.edu/"><acronym +title="Massachusetts Institute of Technology">MIT</acronym></a>, <a +href="http://www.ercim.eu/"><acronym +title="European Research Consortium for Informatics and Mathematics">ERCIM</acronym></a>, +<a href="http://www.keio.ac.jp/">Keio</a>), All Rights Reserved. W3C <a +href="http://www.w3.org/Consortium/Legal/ipr-notice#Legal_Disclaimer">liability</a>, +<a +href="http://www.w3.org/Consortium/Legal/ipr-notice#W3C_Trademarks">trademark</a> +and <a href="http://www.w3.org/Consortium/Legal/copyright-documents">document +use</a> rules apply.</p> +<hr /> +</div> + +<div id="abstract-section" class="section"> +<h2 id="abstract">Abstract</h2> + +<p>This specification describes a high-level JavaScript <acronym +title="Application Programming Interface">API</acronym> for processing and +synthesizing audio in web applications. The primary paradigm is of an audio +routing graph, where a number of <a +href="#AudioNode-section"><code>AudioNode</code></a> objects are connected +together to define the overall audio rendering. The actual processing will +primarily take place in the underlying implementation (typically optimized +Assembly / C / C++ code), but <a href="#JavaScriptProcessing-section">direct +JavaScript processing and synthesis</a> is also supported. </p> + +<p>The <a href="#introduction">introductory</a> section covers the motivation +behind this specification.</p> + +<p>This API is designed to be used in conjunction with other APIs and elements +on the web platform, notably: XMLHttpRequest +(using the <code>responseType</code> and <code>response</code> attributes). For +games and interactive applications, it is anticipated to be used with the +<code>canvas</code> 2D and WebGL 3D graphics APIs. </p> +</div> + +<div id="sotd-section" class="section"> +<h2 id="sotd">Status of this Document</h2> + + +<p><em>This section describes the status of this document at the time of its +publication. Other documents may supersede this document. A list of current W3C +publications and the latest revision of this technical report can be found in +the <a href="http://www.w3.org/TR/">W3C technical reports index</a> at +http://www.w3.org/TR/. </em></p> + +<p>This is the Editor's Draft of the <cite>Web Audio API</cite> +specification. It has been produced by the <a +href="http://www.w3.org/2011/audio/"><b>W3C Audio Working Group</b></a> , which +is part of the W3C WebApps Activity.</p> + +<p></p> + +<p>Please send comments about this document to <<a +href="mailto:public-audio@w3.org">public-audio@w3.org</a>> (<a +href="http://lists.w3.org/Archives/Public/public-audio/">public archives</a> of +the W3C audio mailing list). Web content and browser developers are encouraged +to review this draft. </p> + +<p>Publication as a Working Draft does not imply endorsement by the W3C +Membership. This is a draft document and may be updated, replaced or obsoleted +by other documents at any time. It is inappropriate to cite this document as +other than work in progress.</p> + +<p> This document was produced by a group operating under the <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/">5 February 2004 W3C Patent Policy</a>. W3C maintains a <a rel="disclosure" href="http://www.w3.org/2004/01/pp-impl/46884/status">public list of any patent disclosures</a> made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes contains <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/#def-essential">Essential Claim(s)</a> must disclose the information in accordance with <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/#sec-Disclosure">section 6 of the W3C Patent Policy</a>. </p> +</div> + +<div id="toc"> +<h2 id="L13522">Table of Contents</h2> + +<div class="toc"> +<ul> + <li><a href="#introduction">1. Introduction</a> + <ul> + <li><a href="#Features">1.1. Features</a></li> + <li><a href="#ModularRouting">1.2. Modular Routing</a></li> + <li><a href="#APIOverview">1.3. API Overview</a></li> + </ul> + </li> + <li><a href="#conformance">2. Conformance</a></li> + <li><a href="#API-section">4. The Audio API</a> + <ul> + <li><a href="#AudioContext-section">4.1. The AudioContext Interface</a> + <ul> + <li><a href="#attributes-AudioContext">4.1.1. Attributes</a></li> + <li><a href="#methodsandparams-AudioContext">4.1.2. Methods and + Parameters</a></li> + <li><a href="#lifetime-AudioContext">4.1.3. Lifetime</a></li> + </ul> + </li> + <li><a href="#OfflineAudioContext-section">4.1b. The OfflineAudioContext Interface</a> + </li> + + <li><a href="#AudioNode-section">4.2. The AudioNode Interface</a> + <ul> + <li><a href="#attributes-AudioNode">4.2.1. Attributes</a></li> + <li><a href="#methodsandparams-AudioNode">4.2.2. Methods and + Parameters</a></li> + <li><a href="#lifetime-AudioNode">4.2.3. Lifetime</a></li> + </ul> + </li> + <li><a href="#AudioDestinationNode">4.4. The AudioDestinationNode + Interface</a> + <ul> + <li><a href="#attributes-AudioDestinationNode">4.4.1. Attributes</a></li> + </ul> + </li> + <li><a href="#AudioParam">4.5. The AudioParam Interface</a> + <ul> + <li><a href="#attributes-AudioParam">4.5.1. Attributes</a></li> + <li><a href="#methodsandparams-AudioParam">4.5.2. Methods and + Parameters</a></li> + <li><a href="#computedValue-AudioParam-section">4.5.3. Computation of Value</a></li> + <li><a href="#example1-AudioParam-section">4.5.4. AudioParam Automation Example</a></li> + </ul> + </li> + <li><a href="#GainNode">4.7. The GainNode Interface</a> + <ul> + <li><a href="#attributes-GainNode">4.7.1. Attributes</a></li> + </ul> + </li> + <li><a href="#DelayNode">4.8. The DelayNode Interface</a> + <ul> + <li><a href="#attributes-GainNode_2">4.8.1. Attributes</a></li> + </ul> + </li> + <li><a href="#AudioBuffer">4.9. The AudioBuffer Interface</a> + <ul> + <li><a href="#attributes-AudioBuffer">4.9.1. Attributes</a></li> + <li><a href="#methodsandparams-AudioBuffer">4.9.2. Methods and + Parameters</a></li> + </ul> + </li> + <li><a href="#AudioBufferSourceNode">4.10. The AudioBufferSourceNode + Interface</a> + <ul> + <li><a href="#attributes-AudioBufferSourceNode">4.10.1. + Attributes</a></li> + <li><a href="#methodsandparams-AudioBufferSourceNode">4.10.2. Methods and + Parameters</a></li> + </ul> + </li> + <li><a href="#MediaElementAudioSourceNode">4.11. The + MediaElementAudioSourceNode Interface</a></li> + <li><a href="#ScriptProcessorNode">4.12. The ScriptProcessorNode + Interface</a> + <ul> + <li><a href="#attributes-ScriptProcessorNode">4.12.1. Attributes</a></li> + </ul> + </li> + <li><a href="#AudioProcessingEvent">4.13. The AudioProcessingEvent + Interface</a> + <ul> + <li><a href="#attributes-AudioProcessingEvent">4.13.1. Attributes</a></li> + </ul> + </li> + <li><a href="#PannerNode">4.14. The PannerNode Interface</a> + <ul> + <li><a href="#attributes-PannerNode_attributes">4.14.2. + Attributes</a></li> + <li><a href="#Methods_and_Parameters">4.14.3. Methods and + Parameters</a></li> + </ul> + </li> + <li><a href="#AudioListener">4.15. The AudioListener Interface</a> + <ul> + <li><a href="#attributes-AudioListener">4.15.1. Attributes</a></li> + <li><a href="#L15842">4.15.2. Methods and Parameters</a></li> + </ul> + </li> + <li><a href="#ConvolverNode">4.16. The ConvolverNode Interface</a> + <ul> + <li><a href="#attributes-ConvolverNode">4.16.1. Attributes</a></li> + </ul> + </li> + <li><a href="#AnalyserNode">4.17. The AnalyserNode + Interface</a> + <ul> + <li><a href="#attributes-ConvolverNode_2">4.17.1. Attributes</a></li> + <li><a href="#methods-and-parameters">4.17.2. Methods and + Parameters</a></li> + </ul> + </li> + <li><a href="#ChannelSplitterNode">4.18. The ChannelSplitterNode + Interface</a> + <ul> + <li><a href="#example-1">Example:</a></li> + </ul> + </li> + <li><a href="#ChannelMergerNode">4.19. The ChannelMergerNode Interface</a> + <ul> + <li><a href="#example-2">Example:</a></li> + </ul> + </li> + <li><a href="#DynamicsCompressorNode">4.20. The DynamicsCompressorNode + Interface</a> + <ul> + <li><a href="#attributes-DynamicsCompressorNode">4.20.1. + Attributes</a></li> + </ul> + </li> + <li><a href="#BiquadFilterNode">4.21. The BiquadFilterNode Interface</a> + <ul> + <li><a href="#BiquadFilterNode-description">4.21.1 Lowpass</a></li> + <li><a href="#HIGHPASS">4.21.2 Highpass</a></li> + <li><a href="#BANDPASS">4.21.3 Bandpass</a></li> + <li><a href="#LOWSHELF">4.21.4 Lowshelf</a></li> + <li><a href="#L16352">4.21.5 Highshelf</a></li> + <li><a href="#PEAKING">4.21.6 Peaking</a></li> + <li><a href="#NOTCH">4.21.7 Notch</a></li> + <li><a href="#ALLPASS">4.21.8 Allpass</a></li> + <li><a href="#Methods">4.21.9. Methods</a></li> + </ul> + </li> + <li><a href="#WaveShaperNode">4.22. The WaveShaperNode Interface</a> + <ul> + <li><a href="#attributes-WaveShaperNode">4.22.1. + Attributes</a></li> + </ul> + </li> + <li><a href="#OscillatorNode">4.23. The OscillatorNode Interface</a> + <ul> + <li><a href="#attributes-OscillatorNode">4.23.1. + Attributes</a></li> + <li><a href="#methodsandparams-OscillatorNode-section">4.23.2. Methods and + Parameters</a></li> + </ul> + </li> + <li><a href="#PeriodicWave">4.24. The PeriodicWave Interface</a> + </li> + <li><a href="#MediaStreamAudioSourceNode">4.25. The + MediaStreamAudioSourceNode Interface</a></li> + <li><a href="#MediaStreamAudioDestinationNode">4.26. The + MediaStreamAudioDestinationNode Interface</a></li> + </ul> + </li> + <li><a href="#MixerGainStructure">6. Mixer Gain Structure</a> + <ul> + <li><a href="#background">Background</a></li> + <li><a href="#SummingJunction">Summing Inputs</a></li> + <li><a href="#gain-Control">Gain Control</a></li> + <li><a href="#Example-mixer-with-send-busses">Example: Mixer with Send + Busses</a></li> + </ul> + </li> + <li><a href="#DynamicLifetime">7. Dynamic Lifetime</a> + <ul> + <li><a href="#DynamicLifetime-background">Background</a></li> + <li><a href="#Example-DynamicLifetime">Example</a></li> + </ul> + </li> + <li><a href="#UpMix">9. Channel up-mixing and down-mixing</a> + <ul> + <li><a href="#ChannelLayouts">9.1. Speaker Channel Layouts</a> + <ul> + <li><a href="#ChannelOrdering">9.1.1. Channel Ordering</a></li> + <li><a href="#UpMix-sub">9.1.2. Up Mixing</a></li> + <li><a href="#down-mix">9.1.3. Down Mixing</a></li> + </ul> + </li> + + <li><a href="#ChannelRules-section">9.2. Channel Rules Examples</a> + + </ul> + </li> + <li><a href="#Spatialization">11. Spatialization / Panning </a> + <ul> + <li><a href="#Spatialization-background">Background</a></li> + <li><a href="#Spatialization-panning-algorithm">Panning Algorithm</a></li> + <li><a href="#Spatialization-distance-effects">Distance Effects</a></li> + <li><a href="#Spatialization-sound-cones">Sound Cones</a></li> + <li><a href="#Spatialization-doppler-shift">Doppler Shift</a></li> + </ul> + </li> + <li><a href="#Convolution">12. Linear Effects using Convolution</a> + <ul> + <li><a href="#Convolution-background">Background</a></li> + <li><a href="#Convolution-motivation">Motivation for use as a + Standard</a></li> + <li><a href="#Convolution-implementation-guide">Implementation Guide</a></li> + <li><a href="#Convolution-reverb-effect">Reverb Effect (with + matrixing)</a></li> + <li><a href="#recording-impulse-responses">Recording Impulse + Responses</a></li> + <li><a href="#tools">Tools</a></li> + <li><a href="#recording-setup">Recording Setup</a></li> + <li><a href="#warehouse">The Warehouse Space</a></li> + </ul> + </li> + <li><a href="#JavaScriptProcessing">13. JavaScript Synthesis and + Processing</a> + <ul> + <li><a href="#custom-DSP-effects">Custom DSP Effects</a></li> + <li><a href="#educational-applications">Educational Applications</a></li> + <li><a href="#javaScript-performance">JavaScript Performance</a></li> + </ul> + </li> + <li><a href="#Performance">15. Performance Considerations</a> + <ul> + <li><a href="#Latency">15.1. Latency: What it is and Why it's + Important</a></li> + <li><a href="#audio-glitching">15.2. Audio Glitching</a></li> + <li><a href="#hardware-scalability">15.3. Hardware Scalability</a> + <ul> + <li><a href="#CPU-monitoring">15.3.1. CPU monitoring</a></li> + <li><a href="#Voice-dropping">15.3.2. Voice Dropping</a></li> + <li><a href="#Simplification-of-Effects-Processing">15.3.3. + Simplification of Effects Processing</a></li> + <li><a href="#Sample-rate">15.3.4. Sample Rate</a></li> + <li><a href="#pre-flighting">15.3.5. Pre-flighting</a></li> + <li><a href="#Authoring-for-different-user-agents">15.3.6. Authoring + for different user agents</a></li> + <li><a href="#Scalability-of-Direct-JavaScript-Synthesis">15.3.7. + Scalability of Direct JavaScript Synthesis / Processing</a></li> + </ul> + </li> + <li><a href="#JavaScriptPerformance">15.4. JavaScript Issues with + real-time Processing and Synthesis: </a></li> + </ul> + </li> + <li><a href="#ExampleApplications">16. Example Applications</a> + <ul> + <li><a href="#basic-sound-playback">Basic Sound Playback</a></li> + <li><a href="#threeD-environmentse-and-games">3D Environments and + Games</a></li> + <li><a href="#musical-applications">Musical Applications</a></li> + <li><a href="#music-visualizers">Music Visualizers</a></li> + <li><a href="#educational-applications_2">Educational + Applications</a></li> + <li><a href="#artistic-audio-exploration">Artistic Audio + Exploration</a></li> + </ul> + </li> + <li><a href="#SecurityConsiderations">17. Security Considerations</a></li> + <li><a href="#PrivacyConsiderations">18. Privacy Considerations</a></li> + <li><a href="#requirements">19. Requirements and Use Cases</a></li> + <li><a href="#OldNames">20. Old Names</a></li> + <li><a href="#L17310">A.References</a> + <ul> + <li><a href="#Normative-references">A.1 Normative references</a></li> + <li><a href="#Informative-references">A.2 Informative references</a></li> + </ul> + </li> + <li><a href="#L17335">B.Acknowledgements</a></li> + <li><a href="#ChangeLog">C. Web Audio API Change Log</a></li> +</ul> +</div> +</div> + +<div id="sections"> + +<div id="div-introduction" class="section"> +<h2 id="introduction">1. Introduction</h2> + +<p class="norm">This section is informative.</p> + +<p>Audio on the web has been fairly primitive up to this point and until very +recently has had to be delivered through plugins such as Flash and QuickTime. +The introduction of the <code>audio</code> element in HTML5 is very important, +allowing for basic streaming audio playback. But, it is not powerful enough to +handle more complex audio applications. For sophisticated web-based games or +interactive applications, another solution is required. It is a goal of this +specification to include the capabilities found in modern game audio engines as +well as some of the mixing, processing, and filtering tasks that are found in +modern desktop audio production applications. </p> + +<p>The APIs have been designed with a wide variety of <a +href="#ExampleApplications-section">use cases</a> in mind. Ideally, it should +be able to support <i>any</i> use case which could reasonably be implemented +with an optimized C++ engine controlled via JavaScript and run in a browser. +That said, modern desktop audio software can have very advanced capabilities, +some of which would be difficult or impossible to build with this system. +Apple's Logic Audio is one such application which has support for external MIDI +controllers, arbitrary plugin audio effects and synthesizers, highly optimized +direct-to-disk audio file reading/writing, tightly integrated time-stretching, +and so on. Nevertheless, the proposed system will be quite capable of +supporting a large range of reasonably complex games and interactive +applications, including musical ones. And it can be a very good complement to +the more advanced graphics features offered by WebGL. The API has been designed +so that more advanced capabilities can be added at a later time. </p> + +<div id="Features-section" class="section"> +<h2 id="Features">1.1. Features</h2> +</div> + +<p>The API supports these primary features: </p> +<ul> + <li><a href="#ModularRouting-section">Modular routing</a> for simple or + complex mixing/effect architectures, including <a + href="#MixerGainStructure-section">multiple sends and submixes</a>.</li> + <li><a href="#AudioParam">Sample-accurate scheduled sound + playback</a> with low <a href="#Latency-section">latency</a> for musical + applications requiring a very high degree of rhythmic precision such as + drum machines and sequencers. This also includes the possibility of <a + href="#DynamicLifetime-section">dynamic creation</a> of effects. </li> + <li>Automation of audio parameters for envelopes, fade-ins / fade-outs, + granular effects, filter sweeps, LFOs etc. </li> + <li>Flexible handling of channels in an audio stream, allowing them to be split and merged.</li> + + <li>Processing of audio sources from an <code>audio</code> or + <code>video</code> <a href="#MediaElementAudioSourceNode">media + element</a>. </li> + + <li>Processing live audio input using a <a href="#MediaStreamAudioSourceNode">MediaStream</a> + from getUserMedia(). + </li> + + <li>Integration with WebRTC + <ul> + + + <li>Processing audio received from a remote peer using a <a href="#MediaStreamAudioSourceNode">MediaStream</a>. + </li> + + <li>Sending a generated or processed audio stream to a remote peer using a <a href="#MediaStreamAudioDestinationNode">MediaStream</a>. + </li> + + </ul> + </li> + + <li>Audio stream synthesis and processing <a + href="#JavaScriptProcessing-section">directly in JavaScript</a>. </li> + <li><a href="#Spatialization-section">Spatialized audio</a> supporting a wide + range of 3D games and immersive environments: + <ul> + <li>Panning models: equal-power, HRTF, pass-through </li> + <li>Distance Attenuation </li> + <li>Sound Cones </li> + <li>Obstruction / Occlusion </li> + <li>Doppler Shift </li> + <li>Source / Listener based</li> + </ul> + </li> + <li>A <a href="#Convolution-section">convolution engine</a> for a wide range + of linear effects, especially very high-quality room effects. Here are some + examples of possible effects: + <ul> + <li>Small / large room </li> + <li>Cathedral </li> + <li>Concert hall </li> + <li>Cave </li> + <li>Tunnel </li> + <li>Hallway </li> + <li>Forest </li> + <li>Amphitheater </li> + <li>Sound of a distant room through a doorway </li> + <li>Extreme filters</li> + <li>Strange backwards effects</li> + <li>Extreme comb filter effects </li> + </ul> + </li> + <li>Dynamics compression for overall control and sweetening of the mix </li> + <li>Efficient <a href="#AnalyserNode">real-time time-domain and + frequency analysis / music visualizer support</a></li> + <li>Efficient biquad filters for lowpass, highpass, and other common filters. + </li> + <li>A Waveshaping effect for distortion and other non-linear effects</li> + <li>Oscillators</li> + +</ul> + +<div id="ModularRouting-section"> +<h2 id="ModularRouting">1.2. Modular Routing</h2> + +<p>Modular routing allows arbitrary connections between different <a +href="#AudioNode-section"><code>AudioNode</code></a> objects. Each node can +have <dfn>inputs</dfn> and/or <dfn>outputs</dfn>. A <dfn>source node</dfn> has no inputs +and a single output. A <dfn>destination node</dfn> has +one input and no outputs, the most common example being <a +href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a> the final destination to the audio +hardware. Other nodes such as filters can be placed between the source and destination nodes. +The developer doesn't have to worry about low-level stream format details +when two objects are connected together; <a href="#UpMix-section">the right +thing just happens</a>. For example, if a mono audio stream is connected to a +stereo input it should just mix to left and right channels <a +href="#UpMix-section">appropriately</a>. </p> + +<p>In the simplest case, a single source can be routed directly to the output. +All routing occurs within an <a +href="#AudioContext-section"><code>AudioContext</code></a> containing a single +<a href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a>: +</p> +<img alt="modular routing" src="images/modular-routing1.png" /> + +<p>Illustrating this simple routing, here's a simple example playing a single +sound: </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">ECMAScript</span> </div> + +<div class="blockContent"> +<pre class="code"><code class="es-code"> + +var context = new AudioContext(); + +function playSound() { + var source = context.createBufferSource(); + source.buffer = dogBarkingBuffer; + source.connect(context.destination); + source.start(0); +} + </code></pre> +</div> +</div> + +<p>Here's a more complex example with three sources and a convolution reverb +send with a dynamics compressor at the final output stage: </p> +<img alt="modular routing2" src="images/modular-routing2.png" /> + +<div class="example"> + +<div class="exampleHeader"> +Example</div> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">ECMAScript</span></div> + +<div class="blockContent"> +<pre class="code"><code class="es-code"> + +var context = 0; +var compressor = 0; +var reverb = 0; + +var source1 = 0; +var source2 = 0; +var source3 = 0; + +var lowpassFilter = 0; +var waveShaper = 0; +var panner = 0; + +var dry1 = 0; +var dry2 = 0; +var dry3 = 0; + +var wet1 = 0; +var wet2 = 0; +var wet3 = 0; + +var masterDry = 0; +var masterWet = 0; + +function setupRoutingGraph () { + context = new AudioContext(); + + // Create the effects nodes. + lowpassFilter = context.createBiquadFilter(); + waveShaper = context.createWaveShaper(); + panner = context.createPanner(); + compressor = context.createDynamicsCompressor(); + reverb = context.createConvolver(); + + // Create master wet and dry. + masterDry = context.createGain(); + masterWet = context.createGain(); + + // Connect final compressor to final destination. + compressor.connect(context.destination); + + // Connect master dry and wet to compressor. + masterDry.connect(compressor); + masterWet.connect(compressor); + + // Connect reverb to master wet. + reverb.connect(masterWet); + + // Create a few sources. + source1 = context.createBufferSource(); + source2 = context.createBufferSource(); + source3 = context.createOscillator(); + + source1.buffer = manTalkingBuffer; + source2.buffer = footstepsBuffer; + source3.frequency.value = 440; + + // Connect source1 + dry1 = context.createGain(); + wet1 = context.createGain(); + source1.connect(lowpassFilter); + lowpassFilter.connect(dry1); + lowpassFilter.connect(wet1); + dry1.connect(masterDry); + wet1.connect(reverb); + + // Connect source2 + dry2 = context.createGain(); + wet2 = context.createGain(); + source2.connect(waveShaper); + waveShaper.connect(dry2); + waveShaper.connect(wet2); + dry2.connect(masterDry); + wet2.connect(reverb); + + // Connect source3 + dry3 = context.createGain(); + wet3 = context.createGain(); + source3.connect(panner); + panner.connect(dry3); + panner.connect(wet3); + dry3.connect(masterDry); + wet3.connect(reverb); + + // Start the sources now. + source1.start(0); + source2.start(0); + source3.start(0); +} + </code></pre> +</div> +</div> +</div> +</div> + +</div> + +<div id="APIOverview-section" class="section"> +<h2 id="APIOverview">1.3. API Overview</h2> +</div> + +<p>The interfaces defined are: </p> +<ul> + <li>An <a class="dfnref" href="#AudioContext-section">AudioContext</a> + interface, which contains an audio signal graph representing connections + betweens AudioNodes. </li> + <li>An <a class="dfnref" href="#AudioNode-section">AudioNode</a> interface, + which represents audio sources, audio outputs, and intermediate processing + modules. AudioNodes can be dynamically connected together in a <a + href="#ModularRouting-section">modular fashion</a>. <code>AudioNodes</code> + exist in the context of an <code>AudioContext</code> </li> + <li>An <a class="dfnref" + href="#AudioDestinationNode-section">AudioDestinationNode</a> interface, an + AudioNode subclass representing the final destination for all rendered + audio. </li> + <li>An <a class="dfnref" href="#AudioBuffer-section">AudioBuffer</a> + interface, for working with memory-resident audio assets. These can + represent one-shot sounds, or longer audio clips. </li> + <li>An <a class="dfnref" + href="#AudioBufferSourceNode-section">AudioBufferSourceNode</a> interface, + an AudioNode which generates audio from an AudioBuffer. </li> + <li>A <a class="dfnref" + href="#MediaElementAudioSourceNode-section">MediaElementAudioSourceNode</a> + interface, an AudioNode which is the audio source from an + <code>audio</code>, <code>video</code>, or other media element. </li> + <li>A <a class="dfnref" + href="#MediaStreamAudioSourceNode-section">MediaStreamAudioSourceNode</a> + interface, an AudioNode which is the audio source from a + MediaStream such as live audio input, or from a remote peer. </li> + <li>A <a class="dfnref" + href="#MediaStreamAudioDestinationNode-section">MediaStreamAudioDestinationNode</a> + interface, an AudioNode which is the audio destination to a + MediaStream sent to a remote peer. </li> + <li>A <a class="dfnref" + href="#ScriptProcessorNode-section">ScriptProcessorNode</a> interface, an + AudioNode for generating or processing audio directly in JavaScript. </li> + <li>An <a class="dfnref" + href="#AudioProcessingEvent-section">AudioProcessingEvent</a> interface, + which is an event type used with <code>ScriptProcessorNode</code> objects. + </li> + <li>An <a class="dfnref" href="#AudioParam-section">AudioParam</a> interface, + for controlling an individual aspect of an AudioNode's functioning, such as + volume. </li> + <li>An <a class="dfnref" href="#GainNode-section">GainNode</a> + interface, for explicit gain control. Because inputs to AudioNodes support + multiple connections (as a unity-gain summing junction), mixers can be <a + href="#MixerGainStructure-section">easily built</a> with GainNodes. + </li> + <li>A <a class="dfnref" href="#BiquadFilterNode-section">BiquadFilterNode</a> + interface, an AudioNode for common low-order filters such as: + <ul> + <li>Low Pass</li> + <li>High Pass </li> + <li>Band Pass </li> + <li>Low Shelf </li> + <li>High Shelf </li> + <li>Peaking </li> + <li>Notch </li> + <li>Allpass </li> + </ul> + </li> + <li>A <a class="dfnref" href="#DelayNode-section">DelayNode</a> interface, an + AudioNode which applies a dynamically adjustable variable delay. </li> + <li>An <a class="dfnref" href="#PannerNode-section">PannerNode</a> + interface, for spatializing / positioning audio in 3D space. </li> + <li>An <a class="dfnref" href="#AudioListener-section">AudioListener</a> + interface, which works with an <code>PannerNode</code> for + spatialization. </li> + <li>A <a class="dfnref" href="#ConvolverNode-section">ConvolverNode</a> + interface, an AudioNode for applying a <a + href="#Convolution-section">real-time linear effect</a> (such as the sound + of a concert hall). </li> + <li>A <a class="dfnref" + href="#AnalyserNode-section">AnalyserNode</a> interface, + for use with music visualizers, or other visualization applications. </li> + <li>A <a class="dfnref" + href="#ChannelSplitterNode-section">ChannelSplitterNode</a> interface, + for accessing the individual channels of an audio stream in the routing + graph. </li> + <li>A <a class="dfnref" + href="#ChannelMergerNode-section">ChannelMergerNode</a> interface, for + combining channels from multiple audio streams into a single audio stream. + </li> + <li>A <a + href="#DynamicsCompressorNode-section">DynamicsCompressorNode</a> interface, an + AudioNode for dynamics compression. </li> + <li>A <a class="dfnref" href="#dfn-WaveShaperNode">WaveShaperNode</a> + interface, an AudioNode which applies a non-linear waveshaping effect for + distortion and other more subtle warming effects. </li> + <li>A <a class="dfnref" href="#dfn-OscillatorNode">OscillatorNode</a> + interface, an audio source generating a periodic waveform. </li> +</ul> +</div> + +<div id="conformance-section" class="section"> +<h2 id="conformance">2. Conformance</h2> + +<p>Everything in this specification is normative except for examples and +sections marked as being informative. </p> + +<p>The keywords “<span class="rfc2119">MUST</span>”, “<span +class="rfc2119">MUST NOT</span>”, “<span +class="rfc2119">REQUIRED</span>”, “<span class="rfc2119">SHALL</span>”, +“<span class="rfc2119">SHALL NOT</span>”, “<span +class="rfc2119">RECOMMENDED</span>”, “<span class="rfc2119">MAY</span>” +and “<span class="rfc2119">OPTIONAL</span>” in this document are to be +interpreted as described in <cite><a href="http://www.ietf.org/rfc/rfc2119">Key +words for use in RFCs to Indicate Requirement Levels</a></cite> <a +href="#RFC2119">[RFC2119]</a>. </p> + +<p>The following conformance classes are defined by this specification: </p> +<dl> + <dt><dfn id="dfn-conforming-implementation">conforming + implementation</dfn></dt> + <dd><p>A user agent is considered to be a <a class="dfnref" + href="#dfn-conforming-implementation">conforming implementation</a> if it + satisfies all of the <span class="rfc2119">MUST</span>-, <span + class="rfc2119">REQUIRED</span>- and <span + class="rfc2119">SHALL</span>-level criteria in this specification that + apply to implementations. </p> + </dd> +</dl> +</div> + +<div id="terminology-section" class="section"> + +<div id="API-section-section" class="section"> +<h2 id="API-section">4. The Audio API</h2> +</div> + +<div id="AudioContext-section-section" class="section"> +<h2 id="AudioContext-section">4.1. The AudioContext Interface</h2> + +<p>This interface represents a set of <a +href="#AudioNode-section"><code>AudioNode</code></a> objects and their +connections. It allows for arbitrary routing of signals to the <a +href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a> +(what the user ultimately hears). Nodes are created from the context and are +then <a href="#ModularRouting-section">connected</a> together. In most use +cases, only a single AudioContext is used per document.</p> + +<br> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="audio-context-idl"> + +callback DecodeSuccessCallback = void (AudioBuffer decodedData); +callback DecodeErrorCallback = void (); + +[Constructor] +interface <dfn id="dfn-AudioContext">AudioContext</dfn> : EventTarget { + + readonly attribute AudioDestinationNode destination; + readonly attribute float sampleRate; + readonly attribute double currentTime; + readonly attribute AudioListener listener; + + AudioBuffer createBuffer(unsigned long numberOfChannels, unsigned long length, float sampleRate); + + void decodeAudioData(ArrayBuffer audioData, + DecodeSuccessCallback successCallback, + optional DecodeErrorCallback errorCallback); + + + <span class="comment">// AudioNode creation </span> + AudioBufferSourceNode createBufferSource(); + + MediaElementAudioSourceNode createMediaElementSource(HTMLMediaElement mediaElement); + + MediaStreamAudioSourceNode createMediaStreamSource(MediaStream mediaStream); + MediaStreamAudioDestinationNode createMediaStreamDestination(); + + ScriptProcessorNode createScriptProcessor(optional unsigned long bufferSize = 0, + optional unsigned long numberOfInputChannels = 2, + optional unsigned long numberOfOutputChannels = 2); + + AnalyserNode createAnalyser(); + GainNode createGain(); + DelayNode createDelay(optional double maxDelayTime = 1.0); + BiquadFilterNode createBiquadFilter(); + WaveShaperNode createWaveShaper(); + PannerNode createPanner(); + ConvolverNode createConvolver(); + + ChannelSplitterNode createChannelSplitter(optional unsigned long numberOfOutputs = 6); + ChannelMergerNode createChannelMerger(optional unsigned long numberOfInputs = 6); + + DynamicsCompressorNode createDynamicsCompressor(); + + OscillatorNode createOscillator(); + PeriodicWave createPeriodicWave(Float32Array real, Float32Array imag); + +}; +</code></pre> +</div> +</div> + +<div id="attributes-AudioContext-section" class="section"> +<h3 id="attributes-AudioContext">4.1.1. Attributes</h3> +<dl> + <dt id="dfn-destination"><code>destination</code></dt> + <dd><p>An <a + href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a> + with a single input representing the final destination for all audio. + Usually this will represent the actual audio hardware. + All AudioNodes actively rendering + audio will directly or indirectly connect to <code>destination</code>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-sampleRate"><code>sampleRate</code></dt> + <dd><p>The sample rate (in sample-frames per second) at which the + AudioContext handles audio. It is assumed that all AudioNodes in the + context run at this rate. In making this assumption, sample-rate + converters or "varispeed" processors are not supported in real-time + processing.</p> + </dd> +</dl> +<dl> + <dt id="dfn-currentTime"><code>currentTime</code></dt> + <dd><p>This is a time in seconds which starts at zero when the context is + created and increases in real-time. All scheduled times are relative to + it. This is not a "transport" time which can be started, paused, and + re-positioned. It is always moving forward. A GarageBand-like timeline + transport system can be very easily built on top of this (in JavaScript). + This time corresponds to an ever-increasing hardware timestamp. </p> + </dd> +</dl> +<dl> + <dt id="dfn-listener"><code>listener</code></dt> + <dd><p>An <a href="#AudioListener-section"><code>AudioListener</code></a> + which is used for 3D <a + href="#Spatialization-section">spatialization</a>.</p> + </dd> +</dl> +</div> + +<div id="methodsandparams-AudioContext-section" class="section"> +<h3 id="methodsandparams-AudioContext">4.1.2. Methods and Parameters</h3> +<dl> + <dt id="dfn-createBuffer">The <code>createBuffer</code> method</dt> + <dd><p>Creates an AudioBuffer of the given size. The audio data in the + buffer will be zero-initialized (silent). An NOT_SUPPORTED_ERR exception will be thrown if + the <code>numberOfChannels</code> or <code>sampleRate</code> are out-of-bounds, + or if length is 0.</p> + <p>The <dfn id="dfn-numberOfChannels">numberOfChannels</dfn> parameter + determines how many channels the buffer will have. An implementation must support at least 32 channels. </p> + <p>The <dfn id="dfn-length">length</dfn> parameter determines the size of + the buffer in sample-frames. </p> + <p>The <dfn id="dfn-sampleRate_2">sampleRate</dfn> parameter describes + the sample-rate of the linear PCM audio data in the buffer in + sample-frames per second. An implementation must support sample-rates in at least the range 22050 to 96000.</p> + </dd> +</dl> +<dl> + <dt id="dfn-decodeAudioData">The <code>decodeAudioData</code> method</dt> + <dd><p>Asynchronously decodes the audio file data contained in the + ArrayBuffer. The ArrayBuffer can, for example, be loaded from an XMLHttpRequest's + <code>response</code> attribute after setting the <code>responseType</code> to "arraybuffer". + Audio file data can be in any of the + formats supported by the <code>audio</code> element. </p> + <p><dfn id="dfn-audioData">audioData</dfn> is an ArrayBuffer containing + audio file data.</p> + <p><dfn id="dfn-successCallback">successCallback</dfn> is a callback + function which will be invoked when the decoding is finished. The single + argument to this callback is an AudioBuffer representing the decoded PCM + audio data.</p> + <p><dfn id="dfn-errorCallback">errorCallback</dfn> is a callback function + which will be invoked if there is an error decoding the audio file + data.</p> + + <p> + The following steps must be performed: + </p> + <ol> + + <li>Temporarily neuter the <dfn>audioData</dfn> ArrayBuffer in such a way that JavaScript code may not + access or modify the data.</li> + <li>Queue a decoding operation to be performed on another thread.</li> + <li>The decoding thread will attempt to decode the encoded <dfn>audioData</dfn> into linear PCM. + If a decoding error is encountered due to the audio format not being recognized or supported, or + because of corrupted/unexpected/inconsistent data then the <dfn>audioData</dfn> neutered state + will be restored to normal and the <dfn>errorCallback</dfn> will be + scheduled to run on the main thread's event loop and these steps will be terminated.</li> + <li>The decoding thread will take the result, representing the decoded linear PCM audio data, + and resample it to the sample-rate of the AudioContext if it is different from the sample-rate + of <dfn>audioData</dfn>. The final result (after possibly sample-rate converting) will be stored + in an AudioBuffer. + </li> + <li>The <dfn>audioData</dfn> neutered state will be restored to normal + </li> + <li> + The <dfn>successCallback</dfn> function will be scheduled to run on the main thread's event loop + given the AudioBuffer from step (4) as an argument. + </li> + </ol> + </dd> +</dl> +<dl> + <dt id="dfn-createBufferSource">The <code>createBufferSource</code> + method</dt> + <dd><p>Creates an <a + href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createMediaElementSource">The <code>createMediaElementSource</code> + method</dt> + <dd><p>Creates a <a + href="#MediaElementAudioSourceNode-section"><code>MediaElementAudioSourceNode</code></a> given an HTMLMediaElement. + As a consequence of calling this method, audio playback from the HTMLMediaElement will be re-routed + into the processing graph of the AudioContext.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createMediaStreamSource">The <code>createMediaStreamSource</code> + method</dt> + <dd><p>Creates a <a + href="#MediaStreamAudioSourceNode-section"><code>MediaStreamAudioSourceNode</code></a> given a MediaStream. + As a consequence of calling this method, audio playback from the MediaStream will be re-routed + into the processing graph of the AudioContext.</p> + </dd> +</dl> + +<dl> + <dt id="dfn-createMediaStreamDestination">The <code>createMediaStreamDestination</code> + method</dt> + <dd><p>Creates a <a + href="#MediaStreamAudioDestinationNode-section"><code>MediaStreamAudioDestinationNode</code></a>. + </p> + </dd> +</dl> + +<dl> + <dt id="dfn-createScriptProcessor">The <code>createScriptProcessor</code> + method</dt> + <dd><p>Creates a <a + href="#ScriptProcessorNode"><code>ScriptProcessorNode</code></a> for + direct audio processing using JavaScript. An INDEX_SIZE_ERR exception MUST be thrown if <code>bufferSize</code> or <code>numberOfInputChannels</code> or <code>numberOfOutputChannels</code> + are outside the valid range. </p> + <p>The <dfn id="dfn-bufferSize">bufferSize</dfn> parameter determines the + buffer size in units of sample-frames. If it's not passed in, or if the + value is 0, then the implementation will choose the best buffer size for + the given environment, which will be constant power of 2 throughout the lifetime + of the node. Otherwise if the author explicitly specifies the bufferSize, + it must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, + 16384. This value controls how + frequently the <code>audioprocess</code> event is dispatched and + how many sample-frames need to be processed each call. Lower values for + <code>bufferSize</code> will result in a lower (better) <a + href="#Latency-section">latency</a>. Higher values will be necessary to + avoid audio breakup and <a href="#Glitching-section">glitches</a>. + It is recommended for authors to not specify this buffer size and allow + the implementation to pick a good buffer size to balance between latency + and audio quality. + </p> + <p>The <dfn id="dfn-numberOfInputChannels">numberOfInputChannels</dfn> parameter (defaults to 2) and + determines the number of channels for this node's input. Values of up to 32 must be supported. </p> + <p>The <dfn id="dfn-numberOfOutputChannels">numberOfOutputChannels</dfn> parameter (defaults to 2) and + determines the number of channels for this node's output. Values of up to 32 must be supported.</p> + <p>It is invalid for both <code>numberOfInputChannels</code> and + <code>numberOfOutputChannels</code> to be zero. </p> + </dd> +</dl> +<dl> + <dt id="dfn-createAnalyser">The <code>createAnalyser</code> method</dt> + <dd><p>Creates a <a + href="#AnalyserNode-section"><code>AnalyserNode</code></a>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createGain">The <code>createGain</code> method</dt> + <dd><p>Creates a <a + href="#GainNode-section"><code>GainNode</code></a>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createDelay">The <code>createDelay</code> method</dt> + <dd><p>Creates a <a href="#DelayNode-section"><code>DelayNode</code></a> + representing a variable delay line. The initial default delay time will + be 0 seconds.</p> + <p>The <dfn id="dfn-maxDelayTime">maxDelayTime</dfn> parameter is + optional and specifies the maximum delay time in seconds allowed for the delay line. If specified, this value MUST be + greater than zero and less than three minutes or a NOT_SUPPORTED_ERR exception will be thrown.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createBiquadFilter">The <code>createBiquadFilter</code> + method</dt> + <dd><p>Creates a <a + href="#BiquadFilterNode-section"><code>BiquadFilterNode</code></a> + representing a second order filter which can be configured as one of + several common filter types.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createWaveShaper">The <code>createWaveShaper</code> + method</dt> + <dd><p>Creates a <a + href="#WaveShaperNode-section"><code>WaveShaperNode</code></a> + representing a non-linear distortion.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createPanner">The <code>createPanner</code> method</dt> + <dd><p>Creates an <a + href="#PannerNode-section"><code>PannerNode</code></a>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createConvolver">The <code>createConvolver</code> method</dt> + <dd><p>Creates a <a + href="#ConvolverNode-section"><code>ConvolverNode</code></a>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createChannelSplitter">The <code>createChannelSplitter</code> + method</dt> + <dd><p>Creates an <a + href="#ChannelSplitterNode-section"><code>ChannelSplitterNode</code></a> + representing a channel splitter. An exception will be thrown for invalid parameter values.</p> + <p>The <dfn id="dfn-numberOfOutputs">numberOfOutputs</dfn> parameter + determines the number of outputs. Values of up to 32 must be supported. If not specified, then 6 will be used. </p> + </dd> +</dl> +<dl> + <dt id="dfn-createChannelMerger">The <code>createChannelMerger</code> + method</dt> + <dd><p>Creates an <a + href="#ChannelMergerNode-section"><code>ChannelMergerNode</code></a> + representing a channel merger. An exception will be thrown for invalid parameter values.</p> + <p>The <dfn id="dfn-numberOfInputs">numberOfInputs</dfn> parameter + determines the number of inputs. Values of up to 32 must be supported. If not specified, then 6 will be used. </p> + </dd> +</dl> +<dl> + <dt id="dfn-createDynamicsCompressor">The + <code>createDynamicsCompressor</code> method</dt> + <dd><p>Creates a <a + href="#DynamicsCompressorNode-section"><code>DynamicsCompressorNode</code></a>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createOscillator">The + <code>createOscillator</code> method</dt> + <dd><p>Creates an <a + href="#OscillatorNode-section"><code>OscillatorNode</code></a>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-createPeriodicWave">The + <code>createPeriodicWave</code> method</dt> + <dd><p>Creates a <a + href="#PeriodicWave-section"><code>PeriodicWave</code></a> representing a waveform containing arbitrary harmonic content. + The <code>real</code> and <code>imag</code> parameters must be of type <code>Float32Array</code> of equal + lengths greater than zero and less than or equal to 4096 or an exception will be thrown. + These parameters specify the Fourier coefficients of a + <a href="http://en.wikipedia.org/wiki/Fourier_series">Fourier series</a> representing the partials of a periodic waveform. + The created PeriodicWave will be used with an <a href="#OscillatorNode-section"><code>OscillatorNode</code></a> + and will represent a <em>normalized</em> time-domain waveform having maximum absolute peak value of 1. + Another way of saying this is that the generated waveform of an <a href="#OscillatorNode-section"><code>OscillatorNode</code></a> + will have maximum peak value at 0dBFS. Conveniently, this corresponds to the full-range of the signal values used by the Web Audio API. + Because the PeriodicWave will be normalized on creation, the <code>real</code> and <code>imag</code> parameters + represent <em>relative</em> values. + </p> + <p>The <dfn id="dfn-real">real</dfn> parameter represents an array of <code>cosine</code> terms (traditionally the A terms). + In audio terminology, the first element (index 0) is the DC-offset of the periodic waveform and is usually set to zero. + The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.</p> + <p>The <dfn id="dfn-imag">imag</dfn> parameter represents an array of <code>sine</code> terms (traditionally the B terms). + The first element (index 0) should be set to zero (and will be ignored) since this term does not exist in the Fourier series. + The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.</p> + </dd> +</dl> +</div> +</div> + +<h3 id="lifetime-AudioContext">4.1.3. Lifetime</h3> +<p class="norm">This section is informative.</p> + +<p> +Once created, an <code>AudioContext</code> will continue to play sound until it has no more sound to play, or +the page goes away. +</p> + +<div id="OfflineAudioContext-section-section" class="section"> +<h2 id="OfflineAudioContext-section">4.1b. The OfflineAudioContext Interface</h2> +<p> +OfflineAudioContext is a particular type of AudioContext for rendering/mixing-down (potentially) faster than real-time. +It does not render to the audio hardware, but instead renders as quickly as possible, calling a completion event handler +with the result provided as an AudioBuffer. +</p> + + +<p> +</p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="offline-audio-context-idl"> +[Constructor(unsigned long numberOfChannels, unsigned long length, float sampleRate)] +interface <dfn id="dfn-OfflineAudioContext">OfflineAudioContext</dfn> : AudioContext { + + void startRendering(); + + attribute EventHandler oncomplete; + +}; +</code></pre> +</div> +</div> + + +<div id="attributes-OfflineAudioContext-section" class="section"> +<h3 id="attributes-OfflineAudioContext">4.1b.1. Attributes</h3> +<dl> + <dt id="dfn-oncomplete"><code>oncomplete</code></dt> + <dd><p>An EventHandler of type <a href="#OfflineAudioCompletionEvent-section">OfflineAudioCompletionEvent</a>.</p> + </dd> +</dl> +</div> + + +<div id="methodsandparams-OfflineAudioContext-section" class="section"> +<h3 id="methodsandparams-OfflineAudioContext">4.1b.2. Methods and Parameters</h3> +<dl> + <dt id="dfn-startRendering">The <code>startRendering</code> + method</dt> + <dd><p>Given the current connections and scheduled changes, starts rendering audio. The + <code>oncomplete</code> handler will be called once the rendering has finished. + This method must only be called one time or an exception will be thrown.</p> + </dd> +</dl> +</div> + + +<div id="OfflineAudioCompletionEvent-section" class="section"> +<h2 id="OfflineAudioCompletionEvent">4.1c. The OfflineAudioCompletionEvent Interface</h2> + +<p>This is an <code>Event</code> object which is dispatched to <a +href="#OfflineAudioContext-section"><code>OfflineAudioContext</code></a>. </p> + + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="offline-audio-completion-event-idl"> + +interface <dfn id="dfn-OfflineAudioCompletionEvent">OfflineAudioCompletionEvent</dfn> : Event { + + readonly attribute AudioBuffer renderedBuffer; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-OfflineAudioCompletionEvent-section" class="section"> +<h3 id="attributes-OfflineAudioCompletionEvent">4.1c.1. Attributes</h3> +<dl> + <dt id="dfn-renderedBuffer"><code>renderedBuffer</code></dt> + <dd><p>An AudioBuffer containing the rendered audio data once an OfflineAudioContext has finished rendering. + It will have a number of channels equal to the <code>numberOfChannels</code> parameter + of the OfflineAudioContext constructor.</p> + </dd> +</dl> +</div> +</div> + + +<div id="AudioNode-section-section" class="section"> +<h2 id="AudioNode-section">4.2. The AudioNode Interface</h2> + +<p>AudioNodes are the building blocks of an <a +href="#AudioContext-section"><code>AudioContext</code></a>. This interface +represents audio sources, the audio destination, and intermediate processing +modules. These modules can be connected together to form <a +href="#ModularRouting-section">processing graphs</a> for rendering audio to the +audio hardware. Each node can have <dfn>inputs</dfn> and/or <dfn>outputs</dfn>. +A <dfn>source node</dfn> has no inputs +and a single output. An <a +href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a> has +one input and no outputs and represents the final destination to the audio +hardware. Most processing nodes such as filters will have one input and one +output. Each type of <code>AudioNode</code> differs in the details of how it processes or synthesizes audio. But, in general, <code>AudioNodes</code> +will process its inputs (if it has any), and generate audio for its outputs (if it has any). + </p> + +<p> +Each <dfn>output</dfn> has one or more <dfn>channels</dfn>. The exact number of channels depends on the details of the specific AudioNode. +</p> + +<p> +An output may connect to one or more <code>AudioNode</code> inputs, thus <em>fan-out</em> is supported. An input initially has no connections, +but may be connected from one +or more <code>AudioNode</code> outputs, thus <em>fan-in</em> is supported. When the <code>connect()</code> method is called to connect +an output of an AudioNode to an input of an AudioNode, we call that a <dfn>connection</dfn> to the input. +</p> + +<p> +Each AudioNode <dfn>input</dfn> has a specific number of channels at any given time. This number can change depending on the <dfn>connection(s)</dfn> +made to the input. If the input has no connections then it has one channel which is silent. +</p> + +<p> +For each <dfn>input</dfn>, an <code>AudioNode</code> performs a mixing (usually an up-mixing) of all connections to that input. + +Please see <a href="#MixerGainStructure-section">Mixer Gain Structure</a> for more informative details, and the <a href="#UpMix-section">Channel up-mixing and down-mixing</a> + section for normative requirements. + +</p> + +<p> +For performance reasons, practical implementations will need to use block processing, with each <code>AudioNode</code> processing a +fixed number of sample-frames of size <em>block-size</em>. In order to get uniform behavior across implementations, we will define this +value explicitly. <em>block-size</em> is defined to be 128 sample-frames which corresponds to roughly 3ms at a sample-rate of 44.1KHz. +</p> + +<p> +AudioNodes are <em>EventTarget</em>s, as described in <cite><a href="http://dom.spec.whatwg.org/">DOM</a></cite> +<a href="#DOM">[DOM]</a>. This means that it is possible to dispatch events to AudioNodes the same +way that other EventTargets accept events. +</p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="audio-node-idl"> + +enum <dfn>ChannelCountMode</dfn> { + "max", + "clamped-max", + "explicit" +}; + +enum <dfn>ChannelInterpretation</dfn> { + "speakers", + "discrete" +}; + +interface <dfn id="dfn-AudioNode">AudioNode</dfn> : EventTarget { + + void connect(AudioNode destination, optional unsigned long output = 0, optional unsigned long input = 0); + void connect(AudioParam destination, optional unsigned long output = 0); + void disconnect(optional unsigned long output = 0); + + readonly attribute AudioContext context; + readonly attribute unsigned long numberOfInputs; + readonly attribute unsigned long numberOfOutputs; + + // Channel up-mixing and down-mixing rules for all inputs. + attribute unsigned long channelCount; + attribute ChannelCountMode channelCountMode; + attribute ChannelInterpretation channelInterpretation; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-AudioNode-section" class="section"> +<h3 id="attributes-AudioNode">4.2.1. Attributes</h3> +<dl> + <dt id="dfn-context"><code>context</code></dt> + <dd><p>The AudioContext which owns this AudioNode.</p> + </dd> +</dl> +<dl> + <dt id="dfn-numberOfInputs_2"><code>numberOfInputs</code></dt> + <dd><p>The number of inputs feeding into the AudioNode. For <dfn>source nodes</dfn>, + this will be 0.</p> + </dd> +</dl> +<dl> + <dt id="dfn-numberOfOutputs_2"><code>numberOfOutputs</code></dt> + <dd><p>The number of outputs coming out of the AudioNode. This will be 0 + for an AudioDestinationNode.</p> + </dd> +</dl> +<dl> + <dt id="dfn-channelCount"><code>channelCount</code><dt> + <dd><p>The number of channels used when up-mixing and down-mixing connections to any inputs to the node. The default value is 2 + except for specific nodes where its value is specially determined. + This attribute has no effect for nodes with no inputs. + If this value is set to zero, the implementation MUST raise the + NOT_SUPPORTED_ERR exception.</p> + <p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a> + section for more information on this attribute.</p> + </dd> +</dl> +<dl> + <dt id="dfn-channelCountMode"><code>channelCountMode</code><dt> + <dd><p>Determines how channels will be counted when up-mixing and down-mixing connections to any inputs to the node + . This attribute has no effect for nodes with no inputs.</p> + <p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a> + section for more information on this attribute.</p> + </dd> +</dl> +<dl> + <dt id="dfn-channelInterpretation"><code>channelInterpretation</code><dt> + <dd><p>Determines how individual channels will be treated when up-mixing and down-mixing connections to any inputs to the node. + This attribute has no effect for nodes with no inputs.</p> + <p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a> + section for more information on this attribute.</p> + </dd> +</dl> +</div> + +<div id="methodsandparams-AudioNode-section" class="section"> +<h3 id="methodsandparams-AudioNode">4.2.2. Methods and Parameters</h3> +<dl> + <dt id="dfn-connect-AudioNode">The <code>connect</code> to AudioNode method</dt> + <dd><p>Connects the AudioNode to another AudioNode.</p> + <p>The <dfn id="dfn-destination_2">destination</dfn> parameter is the + AudioNode to connect to.</p> + <p>The <dfn id="dfn-output_2">output</dfn> parameter is an index + describing which output of the AudioNode from which to connect. An + out-of-bound value throws an exception.</p> + <p>The <dfn id="dfn-input_2">input</dfn> parameter is an index describing + which input of the destination AudioNode to connect to. An out-of-bound + value throws an exception. </p> + <p>It is possible to connect an AudioNode output to more than one input + with multiple calls to connect(). Thus, "fan-out" is supported. </p> + <p> + It is possible to connect an AudioNode to another AudioNode which creates a <em>cycle</em>. + In other words, an AudioNode may connect to another AudioNode, which in turn connects back + to the first AudioNode. This is allowed only if there is at least one + <a class="dfnref" href="#DelayNode-section">DelayNode</a> in the <em>cycle</em> or an exception will + be thrown. + </p> + + <p> + There can only be one connection between a given output of one specific node and a given input of another specific node. + Multiple connections with the same termini are ignored. For example: + </p> + + <pre> + nodeA.connect(nodeB); + nodeA.connect(nodeB); + + will have the same effect as + + nodeA.connect(nodeB); + </pre> + + </dd> +</dl> +<dl> + <dt id="dfn-connect-AudioParam">The <code>connect</code> to AudioParam method</dt> + <dd><p>Connects the AudioNode to an AudioParam, controlling the parameter + value with an audio-rate signal. + </p> + + <p>The <dfn id="dfn-destination_3">destination</dfn> parameter is the + AudioParam to connect to.</p> + <p>The <dfn id="dfn-output_3-destination">output</dfn> parameter is an index + describing which output of the AudioNode from which to connect. An + out-of-bound value throws an exception.</p> + + <p>It is possible to connect an AudioNode output to more than one AudioParam + with multiple calls to connect(). Thus, "fan-out" is supported. </p> + <p>It is possible to connect more than one AudioNode output to a single AudioParam + with multiple calls to connect(). Thus, "fan-in" is supported. </p> + <p>An AudioParam will take the rendered audio data from any AudioNode output connected to it and <a href="#down-mix">convert it to mono</a> by down-mixing if it is not + already mono, then mix it together with other such outputs and finally will mix with the <em>intrinsic</em> + parameter value (the value the AudioParam would normally have without any audio connections), including any timeline changes + scheduled for the parameter. </p> + + <p> + There can only be one connection between a given output of one specific node and a specific AudioParam. + Multiple connections with the same termini are ignored. For example: + </p> + + <pre> + nodeA.connect(param); + nodeA.connect(param); + + will have the same effect as + + nodeA.connect(param); + </pre> + + </dd> +</dl> +<dl> + <dt id="dfn-disconnect">The <code>disconnect</code> method</dt> + <dd><p>Disconnects an AudioNode's output.</p> + <p>The <dfn id="dfn-output_3-disconnect">output</dfn> parameter is an index + describing which output of the AudioNode to disconnect. An out-of-bound + value throws an exception.</p> + </dd> +</dl> +</div> +</div> + +<h3 id="lifetime-AudioNode">4.2.3. Lifetime</h3> + +<p class="norm">This section is informative.</p> + +<p>An implementation may choose any method to avoid unnecessary resource usage and unbounded memory growth of unused/finished +nodes. The following is a description to help guide the general expectation of how node lifetime would be managed. +</p> + +<p> +An <code>AudioNode</code> will live as long as there are any references to it. There are several types of references: +</p> + +<ol> +<li>A <em>normal</em> JavaScript reference obeying normal garbage collection rules. </li> +<li>A <em>playing</em> reference for both <code>AudioBufferSourceNodes</code> and <code>OscillatorNodes</code>. +These nodes maintain a <em>playing</em> +reference to themselves while they are currently playing.</li> +<li>A <em>connection</em> reference which occurs if another <code>AudioNode</code> is connected to it. </li> +<li>A <em>tail-time</em> reference which an <code>AudioNode</code> maintains on itself as long as it has +any internal processing state which has not yet been emitted. For example, a <code>ConvolverNode</code> has +a tail which continues to play even after receiving silent input (think about clapping your hands in a large concert + hall and continuing to hear the sound reverberate throughout the hall). Some <code>AudioNodes</code> have this + property. Please see details for specific nodes.</li> +</ol> + +<p> +Any <code>AudioNodes</code> which are connected in a cycle <em>and</em> are directly or indirectly connected to the +<code>AudioDestinationNode</code> of the <code>AudioContext</code> will stay alive as long as the <code>AudioContext</code> is alive. +</p> + +<p> +When an <code>AudioNode</code> has no references it will be deleted. But before it is deleted, it will disconnect itself +from any other <code>AudioNodes</code> which it is connected to. In this way it releases all connection references (3) it has to other nodes. +</p> + +<p> +Regardless of any of the above references, it can be assumed that the <code>AudioNode</code> will be deleted when its <code>AudioContext</code> is deleted. +</p> + + +<div id="AudioDestinationNode-section" class="section"> +<h2 id="AudioDestinationNode">4.4. The AudioDestinationNode Interface</h2> + +<p>This is an <a href="#AudioNode-section"><code>AudioNode</code></a> +representing the final audio destination and is what the user will ultimately +hear. It can often be considered as an audio output device which is connected to +speakers. All rendered audio to be heard will be routed to this node, a +"terminal" node in the AudioContext's routing graph. There is only a single +AudioDestinationNode per AudioContext, provided through the +<code>destination</code> attribute of <a +href="#AudioContext-section"><code>AudioContext</code></a>. </p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 0 + + channelCount = 2; + channelCountMode = "explicit"; + channelInterpretation = "speakers"; +</pre> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="audio-destination-node-idl"> + +interface <dfn id="dfn-AudioDestinationNode">AudioDestinationNode</dfn> : AudioNode { + + readonly attribute unsigned long maxChannelCount; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-AudioDestinationNode-section" class="section"> +<h3 id="attributes-AudioDestinationNode">4.4.1. Attributes</h3> +<dl> + <dt id="dfn-maxChannelCount"><code>maxChannelCount</code></dt> + <dd><p>The maximum number of channels that the <code>channelCount</code> attribute can be set to. + An <code>AudioDestinationNode</code> representing the audio hardware end-point (the normal case) can potentially output more than + 2 channels of audio if the audio hardware is multi-channel. <code>maxChannelCount</code> is the maximum number of channels that + this hardware is capable of supporting. If this value is 0, then this indicates that <code>channelCount</code> may not be + changed. This will be the case for an <code>AudioDestinationNode</code> in an <code>OfflineAudioContext</code> and also for + basic implementations with hardware support for stereo output only.</p> + + <p><code>channelCount</code> defaults to 2 for a destination in a normal AudioContext, and may be set to any non-zero value less than or equal + to <code>maxChannelCount</code>. An exception will be thrown if this value is not within the valid range. Giving a concrete example, if + the audio hardware supports 8-channel output, then we may set <code>numberOfChannels</code> to 8, and render 8-channels of output. + </p> + + <p> + For an AudioDestinationNode in an OfflineAudioContext, the <code>channelCount</code> is determined when the offline context is created and this value + may not be changed. + </p> + + </dd> +</dl> + +</div> +</div> + +<div id="AudioParam-section" class="section"> +<h2 id="AudioParam">4.5. The AudioParam Interface</h2> + +<p>AudioParam controls an individual aspect of an <a +href="#AudioNode-section"><code>AudioNode</code></a>'s functioning, such as +volume. The parameter can be set immediately to a particular value using the +"value" attribute. Or, value changes can be scheduled to happen at +very precise times (in the coordinate system of AudioContext.currentTime), for envelopes, volume fades, LFOs, filter sweeps, grain +windows, etc. In this way, arbitrary timeline-based automation curves can be +set on any AudioParam. Additionally, audio signals from the outputs of <code>AudioNodes</code> can be connected +to an <code>AudioParam</code>, summing with the <em>intrinsic</em> parameter value. +</p> + +<p> +Some synthesis and processing <code>AudioNodes</code> have <code>AudioParams</code> as attributes whose values must + be taken into account on a per-audio-sample basis. +For other <code>AudioParams</code>, sample-accuracy is not important and the value changes can be sampled more coarsely. +Each individual <code>AudioParam</code> will specify that it is either an <em>a-rate</em> parameter +which means that its values must be taken into account on a per-audio-sample basis, or it is a <em>k-rate</em> parameter. +</p> + +<p> +Implementations must use block processing, with each <code>AudioNode</code> +processing 128 sample-frames in each block. +</p> + +<p> +For each 128 sample-frame block, the value of a <em>k-rate</em> parameter must +be sampled at the time of the very first sample-frame, and that value must be +used for the entire block. <em>a-rate</em> parameters must be sampled for each +sample-frame of the block. +</p> + + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="audio-param-idl"> + +interface <dfn id="dfn-AudioParam">AudioParam</dfn> { + + attribute float value; + readonly attribute float defaultValue; + + <span class="comment">// Parameter automation. </span> + void setValueAtTime(float value, double startTime); + void linearRampToValueAtTime(float value, double endTime); + void exponentialRampToValueAtTime(float value, double endTime); + + <span class="comment">// Exponentially approach the target value with a rate having the given time constant. </span> + void setTargetAtTime(float target, double startTime, double timeConstant); + + <span class="comment">// Sets an array of arbitrary parameter values starting at time for the given duration. </span> + <span class="comment">// The number of values will be scaled to fit into the desired duration. </span> + void setValueCurveAtTime(Float32Array values, double startTime, double duration); + + <span class="comment">// Cancels all scheduled parameter changes with times greater than or equal to startTime. </span> + void cancelScheduledValues(double startTime); + +}; +</code></pre> +</div> +</div> + + + +<div id="attributes-AudioParam-section" class="section"> +<h3 id="attributes-AudioParam">4.5.1. Attributes</h3> + +<dl> + <dt id="dfn-value"><code>value</code></dt> + <dd><p>The parameter's floating-point value. This attribute is initialized to the + <code>defaultValue</code>. If a value is set during a time when there are any automation events scheduled then + it will be ignored and no exception will be thrown.</p> + </dd> +</dl> +<dl> + <dt id="dfn-defaultValue"><code>defaultValue</code></dt> + <dd><p>Initial value for the value attribute</p> + </dd> +</dl> +</div> + +<div id="methodsandparams-AudioParam-section" class="section"> +<h3 id="methodsandparams-AudioParam">4.5.2. Methods and Parameters</h3> + +<p> +An <code>AudioParam</code> maintains a time-ordered event list which is initially empty. The times are in +the time coordinate system of AudioContext.currentTime. The events define a mapping from time to value. The following methods +can change the event list by adding a new event into the list of a type specific to the method. Each event +has a time associated with it, and the events will always be kept in time-order in the list. These +methods will be called <em>automation</em> methods:</p> + +<ul> +<li>setValueAtTime() - <em>SetValue</em></li> +<li>linearRampToValueAtTime() - <em>LinearRampToValue</em></li> +<li>exponentialRampToValueAtTime() - <em>ExponentialRampToValue</em></li> +<li>setTargetAtTime() - <em>SetTarget</em></li> +<li>setValueCurveAtTime() - <em>SetValueCurve</em></li> +</ul> + +<p> +The following rules will apply when calling these methods: +</p> +<ul> +<li>If one of these events is added at a time where there is already an event of the exact same type, then the new event will replace the old +one.</li> +<li>If one of these events is added at a time where there is already one or more events of a different type, then it will be +placed in the list after them, but before events whose times are after the event. </li> +<li>If setValueCurveAtTime() is called for time T and duration D and there are any events having a time greater than T, but less than +T + D, then an exception will be thrown. In other words, it's not ok to schedule a value curve during a time period containing other events.</li> +<li>Similarly an exception will be thrown if any <em>automation</em> method is called at a time which is inside of the time interval +of a <em>SetValueCurve</em> event at time T and duration D.</li> +</ul> +<p> +</p> + +<dl> + <dt id="dfn-setValueAtTime">The <code>setValueAtTime</code> method</dt> + <dd><p>Schedules a parameter value change at the given time.</p> + <p>The <dfn id="dfn-value_2">value</dfn> parameter is the value the + parameter will change to at the given time.</p> + <p>The <dfn id="dfn-startTime_2">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p> + <p> + If there are no more events after this <em>SetValue</em> event, then for t >= startTime, v(t) = value. In other words, the value will remain constant. + </p> + <p> + If the next event (having time T1) after this <em>SetValue</em> event is not of type <em>LinearRampToValue</em> or <em>ExponentialRampToValue</em>, + then, for t: startTime <= t < T1, v(t) = value. + In other words, the value will remain constant during this time interval, allowing the creation of "step" functions. + </p> + <p> + If the next event after this <em>SetValue</em> event is of type <em>LinearRampToValue</em> or <em>ExponentialRampToValue</em> then please + see details below. + </p> + </dd> +</dl> +<dl> + <dt id="dfn-linearRampToValueAtTime">The <code>linearRampToValueAtTime</code> + method</dt> + <dd><p>Schedules a linear continuous change in parameter value from the + previous scheduled parameter value to the given value.</p> + <p>The <dfn id="dfn-value_3">value</dfn> parameter is the value the + parameter will linearly ramp to at the given time.</p> + <p>The <dfn id="dfn-endTime_3">endTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p> + + <p> + The value during the time interval T0 <= t < T1 (where T0 is the time of the previous event and T1 is the endTime parameter passed into this method) + will be calculated as: + </p> + <pre> + v(t) = V0 + (V1 - V0) * ((t - T0) / (T1 - T0)) + </pre> + <p> + Where V0 is the value at the time T0 and V1 is the value parameter passed into this method. + </p> + <p> + If there are no more events after this LinearRampToValue event then for t >= T1, v(t) = V1 + </p> + + </dd> +</dl> +<dl> + <dt id="dfn-exponentialRampToValueAtTime">The + <code>exponentialRampToValueAtTime</code> method</dt> + <dd><p>Schedules an exponential continuous change in parameter value from + the previous scheduled parameter value to the given value. Parameters + representing filter frequencies and playback rate are best changed + exponentially because of the way humans perceive sound. </p> + <p>The <dfn id="dfn-value_4">value</dfn> parameter is the value the + parameter will exponentially ramp to at the given time. An exception will be thrown if this value is less than + or equal to 0, or if the value at the time of the previous event is less than or equal to 0.</p> + <p>The <dfn id="dfn-endTime_4">endTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p> + <p> + The value during the time interval T0 <= t < T1 (where T0 is the time of the previous event and T1 is the endTime parameter passed into this method) + will be calculated as: + </p> + <pre> + v(t) = V0 * (V1 / V0) ^ ((t - T0) / (T1 - T0)) + </pre> + <p> + Where V0 is the value at the time T0 and V1 is the value parameter passed into this method. + </p> + <p> + If there are no more events after this ExponentialRampToValue event then for t >= T1, v(t) = V1 + </p> + </dd> +</dl> +<dl> + <dt id="dfn-setTargetAtTime">The <code>setTargetAtTime</code> + method</dt> + <dd><p>Start exponentially approaching the target value at the given time + with a rate having the given time constant. Among other uses, this is + useful for implementing the "decay" and "release" portions of an ADSR + envelope. Please note that the parameter value does not immediately + change to the target value at the given time, but instead gradually + changes to the target value.</p> + <p>The <dfn id="dfn-target">target</dfn> parameter is the value + the parameter will <em>start</em> changing to at the given time.</p> + <p>The <dfn id="dfn-startTime">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p> + <p>The <dfn id="dfn-timeConstant">timeConstant</dfn> parameter is the + time-constant value of first-order filter (exponential) approach to the + target value. The larger this value is, the slower the transition will + be.</p> + <p> + More precisely, <em>timeConstant</em> is the time it takes a first-order linear continuous time-invariant system + to reach the value 1 - 1/e (around 63.2%) given a step input response (transition from 0 to 1 value). + </p> + <p> + During the time interval: <em>T0</em> <= t < <em>T1</em>, where T0 is the <em>startTime</em> parameter and T1 represents the time of the event following this + event (or <em>infinity</em> if there are no following events): + </p> + <pre> + v(t) = V1 + (V0 - V1) * exp(-(t - T0) / <em>timeConstant</em>) + </pre> + <p> + Where V0 is the initial value (the .value attribute) at T0 (the <em>startTime</em> parameter) and V1 is equal to the <em>target</em> + parameter. + </p> + </dd> +</dl> +<dl> + <dt id="dfn-setValueCurveAtTime">The <code>setValueCurveAtTime</code> + method</dt> + <dd><p>Sets an array of arbitrary parameter values starting at the given + time for the given duration. The number of values will be scaled to fit + into the desired duration. </p> + <p>The <dfn id="dfn-values">values</dfn> parameter is a Float32Array + representing a parameter value curve. These values will apply starting at + the given time and lasting for the given duration. </p> + <p>The <dfn id="dfn-startTime_5">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p> + <p>The <dfn id="dfn-duration_5">duration</dfn> parameter is the + amount of time in seconds (after the <em>time</em> parameter) where values will be calculated according to the <em>values</em> parameter..</p> + <p> + During the time interval: <em>startTime</em> <= t < <em>startTime</em> + <em>duration</em>, values will be calculated: + </p> + <pre> + v(t) = values[N * (t - startTime) / duration], where <em>N</em> is the length of the <em>values</em> array. + </pre> + <p> + After the end of the curve time interval (t >= <em>startTime</em> + <em>duration</em>), the value will remain constant at the final curve value, + until there is another automation event (if any). + </p> + </dd> +</dl> +<dl> + <dt id="dfn-cancelScheduledValues">The <code>cancelScheduledValues</code> + method</dt> + <dd><p>Cancels all scheduled parameter changes with times greater than or + equal to startTime.</p> + <p>The <dfn>startTime</dfn> parameter is the starting + time at and after which any previously scheduled parameter changes will + be cancelled. It is a time in the same time coordinate system as AudioContext.currentTime.</p> + </dd> +</dl> +</div> +</div> + + + +<div id="computedValue-AudioParam-section" class="section"> +<h3>4.5.3. Computation of Value</h3> + +<p> +<dfn>computedValue</dfn> is the final value controlling the audio DSP and is computed by the audio rendering thread during each rendering time quantum. + It must be internally computed as follows: +</p> + +<ol> +<li>An <em>intrinsic</em> parameter value will be calculated at each time, which is either the value set directly to the .value attribute, +or, if there are any scheduled parameter changes (automation events) with times before or at this time, +the value as calculated from these events. If the .value attribute +is set after any automation events have been scheduled, then these events will be removed. When read, the .value attribute +always returns the <em>intrinsic</em> value for the current time. If automation events are removed from a given time range, then the +<em>intrinsic</em> value will remain unchanged and stay at its previous value until either the .value attribute is directly set, or automation events are added +for the time range. +</li> + +<li> +An AudioParam will take the rendered audio data from any AudioNode output connected to it and <a href="#down-mix">convert it to mono</a> by down-mixing if it is not +already mono, then mix it together with other such outputs. If there are no AudioNodes connected to it, then this value is 0, having no +effect on the <em>computedValue</em>. +</li> + +<li> +The <em>computedValue</em> is the sum of the <em>intrinsic</em> value and the value calculated from (2). +</li> + +</ol> + +</div> + + +<div id="example1-AudioParam-section" class="section"> +<h3 id="example1-AudioParam">4.5.4. AudioParam Automation Example</h3> + + + +<div class="example"> + +<div class="exampleHeader"> +Example</div> +<img alt="AudioParam automation" src="images/audioparam-automation1.png" /> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">ECMAScript</span></div> + +<div class="blockContent"> +<pre class="code"><code class="es-code"> +var t0 = 0; +var t1 = 0.1; +var t2 = 0.2; +var t3 = 0.3; +var t4 = 0.4; +var t5 = 0.6; +var t6 = 0.7; +var t7 = 1.0; + +var curveLength = 44100; +var curve = new Float32Array(curveLength); +for (var i = 0; i < curveLength; ++i) + curve[i] = Math.sin(Math.PI * i / curveLength); + +param.setValueAtTime(0.2, t0); +param.setValueAtTime(0.3, t1); +param.setValueAtTime(0.4, t2); +param.linearRampToValueAtTime(1, t3); +param.linearRampToValueAtTime(0.15, t4); +param.exponentialRampToValueAtTime(0.75, t5); +param.exponentialRampToValueAtTime(0.05, t6); +param.setValueCurveAtTime(curve, t6, t7 - t6); +</code></pre> +</div> +</div> +</div> +</div> + +<div id="GainNode-section" class="section"> +<h2 id="GainNode">4.7. The GainNode Interface</h2> + +<p>Changing the gain of an audio signal is a fundamental operation in audio +applications. The <code>GainNode</code> is one of the building blocks for creating <a +href="#MixerGainStructure-section">mixers</a>. +This interface is an AudioNode with a single input and single +output: </p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 + + channelCountMode = "max"; + channelInterpretation = "speakers"; +</pre> + +<p>It multiplies the input audio signal by the (possibly time-varying) <code>gain</code> attribute, copying the result to the output. + By default, it will take the input and pass it through to the output unchanged, which represents a constant gain change + of 1. +</p> + +<p> +As with other <code>AudioParams</code>, the <code>gain</code> parameter represents a mapping from time +(in the coordinate system of AudioContext.currentTime) to floating-point value. + +Every PCM audio sample in the input is multiplied by the <code>gain</code> parameter's value for the specific time +corresponding to that audio sample. This multiplied value represents the PCM audio sample for the output. +</p> + +<p> +The number of channels of the output will always equal the number of channels of the input, with each channel +of the input being multiplied by the <code>gain</code> values and being copied into the corresponding channel +of the output. +</p> + +<p> + The implementation must make +gain changes to the audio stream smoothly, without introducing noticeable +clicks or glitches. This process is called "de-zippering". </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="gain-node-idl"> + +interface <dfn id="dfn-GainNode">GainNode</dfn> : AudioNode { + + readonly attribute AudioParam gain; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-GainNode-section" class="section"> +<h3 id="attributes-GainNode">4.7.1. Attributes</h3> +<dl> + <dt id="dfn-gain"><code>gain</code></dt> + <dd><p>Represents the amount of gain to apply. Its + default <code>value</code> is 1 (no gain change). The nominal <code>minValue</code> is 0, but may be + set negative for phase inversion. The nominal <code>maxValue</code> is 1, but higher values are allowed (no + exception thrown).This parameter is <em>a-rate</em> </p> + </dd> +</dl> +</div> +</div> + +<div id="DelayNode-section" class="section"> +<h2 id="DelayNode">4.8. The DelayNode Interface</h2> + +<p>A delay-line is a fundamental building block in audio applications. This +interface is an AudioNode with a single input and single output: </p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 + + channelCountMode = "max"; + channelInterpretation = "speakers"; +</pre> + +<p> +The number of channels of the output always equals the number of channels of the input. +</p> + +<p>It delays the incoming audio signal by a certain amount. The default +amount is 0 seconds (no delay). When the delay time is changed, the +implementation must make the transition smoothly, without introducing +noticeable clicks or glitches to the audio stream. </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="delay-node-idl"> + +interface <dfn id="dfn-DelayNode">DelayNode</dfn> : AudioNode { + + readonly attribute AudioParam delayTime; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-GainNode-section_2" class="section"> +<h3 id="attributes-GainNode_2">4.8.1. Attributes</h3> +<dl> + <dt id="dfn-delayTime_2"><code>delayTime</code></dt> + <dd><p>An AudioParam object representing the amount of delay (in seconds) + to apply. The default value (<code>delayTime.value</code>) is 0 (no + delay). The minimum value is 0 and the maximum value is determined by the <em>maxDelayTime</em> + argument to the <code>AudioContext</code> method <code>createDelay</code>. This parameter is <em>a-rate</em></p> + </dd> +</dl> +</div> +</div> + +<div id="AudioBuffer-section" class="section"> +<h2 id="AudioBuffer">4.9. The AudioBuffer Interface</h2> + +<p>This interface represents a memory-resident audio asset (for one-shot sounds +and other short audio clips). Its format is non-interleaved IEEE 32-bit linear PCM with a +nominal range of -1 -> +1. It can contain one or more channels. Typically, it would be expected that the length +of the PCM data would be fairly short (usually somewhat less than a minute). +For longer sounds, such as music soundtracks, streaming should be used with the +<code>audio</code> element and <code>MediaElementAudioSourceNode</code>. </p> + +<p> +An AudioBuffer may be used by one or more AudioContexts. +</p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="audio-buffer-idl"> + +interface <dfn id="dfn-AudioBuffer">AudioBuffer</dfn> { + + readonly attribute float sampleRate; + readonly attribute long length; + + <span class="comment">// in seconds </span> + readonly attribute double duration; + + readonly attribute long numberOfChannels; + + Float32Array getChannelData(unsigned long channel); + +}; +</code></pre> +</div> +</div> + +<div id="attributes-AudioBuffer-section" class="section"> +<h3 id="attributes-AudioBuffer">4.9.1. Attributes</h3> +<dl> + <dt id="dfn-sampleRate_AudioBuffer"><code>sampleRate</code></dt> + <dd><p>The sample-rate for the PCM audio data in samples per second.</p> + </dd> +</dl> +<dl> + <dt id="dfn-length_AudioBuffer"><code>length</code></dt> + <dd><p>Length of the PCM audio data in sample-frames.</p> + </dd> +</dl> +<dl> + <dt id="dfn-duration_AudioBuffer"><code>duration</code></dt> + <dd><p>Duration of the PCM audio data in seconds.</p> + </dd> +</dl> +<dl> + <dt id="dfn-numberOfChannels_AudioBuffer"><code>numberOfChannels</code></dt> + <dd><p>The number of discrete audio channels.</p> + </dd> +</dl> +</div> + +<div id="methodsandparams-AudioBuffer-section" class="section"> +<h3 id="methodsandparams-AudioBuffer">4.9.2. Methods and Parameters</h3> +<dl> + <dt id="dfn-getChannelData">The <code>getChannelData</code> method</dt> + <dd><p>Returns the <code>Float32Array</code> representing the PCM audio data for the specific channel.</p> + <p>The <dfn id="dfn-channel">channel</dfn> parameter is an index + representing the particular channel to get data for. An index value of 0 represents + the first channel. This index value MUST be less than <code>numberOfChannels</code> + or an exception will be thrown.</p> + </dd> +</dl> +</div> +</div> + +<div id="AudioBufferSourceNode-section" class="section"> +<h2 id="AudioBufferSourceNode">4.10. The AudioBufferSourceNode Interface</h2> + +<p>This interface represents an audio source from an in-memory audio asset in +an <code>AudioBuffer</code>. It is useful for playing short audio assets +which require a high degree of scheduling flexibility (can playback in +rhythmically perfect ways). The start() method is used to schedule when +sound playback will happen. The playback will stop automatically when +the buffer's audio data has been completely +played (if the <code>loop</code> attribute is false), or when the stop() +method has been called and the specified time has been reached. Please see more +details in the start() and stop() description. start() and stop() may not be issued +multiple times for a given +AudioBufferSourceNode. </p> +<pre> numberOfInputs : 0 + numberOfOutputs : 1 + </pre> + +<p> +The number of channels of the output always equals the number of channels of the AudioBuffer +assigned to the .buffer attribute, or is one channel of silence if .buffer is NULL. +</p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="audio-buffer-source-node-idl"> + +interface <dfn id="dfn-AudioBufferSourceNode">AudioBufferSourceNode</dfn> : AudioNode { + + attribute AudioBuffer? buffer; + + readonly attribute AudioParam playbackRate; + + attribute boolean loop; + attribute double loopStart; + attribute double loopEnd; + + void start(optional double when = 0, optional double offset = 0, optional double duration); + void stop(optional double when = 0); + + attribute EventHandler onended; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-AudioBufferSourceNode-section" class="section"> +<h3 id="attributes-AudioBufferSourceNode">4.10.1. Attributes</h3> +<dl> + <dt id="dfn-buffer_AudioBufferSourceNode"><code>buffer</code></dt> + <dd><p>Represents the audio asset to be played. </p> + </dd> +</dl> +<dl> + <dt id="dfn-playbackRate_AudioBufferSourceNode"><code>playbackRate</code></dt> + <dd><p>The speed at which to render the audio stream. The default + playbackRate.value is 1. This parameter is <em>a-rate</em> </p> + </dd> +</dl> +<dl> + <dt id="dfn-loop_AudioBufferSourceNode"><code>loop</code></dt> + <dd><p>Indicates if the audio data should play in a loop. The default value is false. </p> + </dd> +</dl> + +<dl> + <dt id="dfn-loopStart_AudioBufferSourceNode"><code>loopStart</code></dt> + <dd><p>An optional value in seconds where looping should begin if the <code>loop</code> attribute is true. + Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer.</p> + </dd> +</dl> +<dl> + <dt id="dfn-loopEnd_AudioBufferSourceNode"><code>loopEnd</code></dt> + <dd><p>An optional value in seconds where looping should end if the <code>loop</code> attribute is true. + Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer.</p> + </dd> +</dl> +<dl> + <dt id="dfn-onended_AudioBufferSourceNode"><code>onended</code></dt> + <dd><p>A property used to set the <code>EventHandler</code> (described in <cite><a + href="http://www.whatwg.org/specs/web-apps/current-work/#eventhandler">HTML</a></cite>) + for the ended event that is dispatched to <a + href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a> + node types. When the playback of the buffer for an <code>AudioBufferSourceNode</code> + is finished, an event of type <code>Event</code> (described in <cite><a + href="http://www.whatwg.org/specs/web-apps/current-work/#event">HTML</a></cite>) + will be dispatched to the event handler. </p> + </dd> +</dl> + + +</div> +</div> + +<div id="methodsandparams-AudioBufferSourceNode-section" class="section"> +<h3 id="methodsandparams-AudioBufferSourceNode">4.10.2. Methods and +Parameters</h3> +<dl> + <dt id="dfn-start">The <code>start</code> method</dt> + <dd><p>Schedules a sound to playback at an exact time.</p> + <p>The <dfn id="dfn-when">when</dfn> parameter describes at what time (in + seconds) the sound should start playing. It is in the same + time coordinate system as AudioContext.currentTime. If 0 is passed in for + this value or if the value is less than <b>currentTime</b>, then the + sound will start playing immediately. <code>start</code> may only be called one time + and must be called before <code>stop</code> is called or an exception will be thrown.</p> + <p>The <dfn id="dfn-offset">offset</dfn> parameter describes + the offset time in the buffer (in seconds) where playback will begin. If 0 is passed + in for this value, then playback will start from the beginning of the buffer.</p> + <p>The <dfn id="dfn-duration">duration</dfn> parameter + describes the duration of the portion (in seconds) to be played. If this parameter is not passed, + the duration will be equal to the total duration of the AudioBuffer minus the <code>offset</code> parameter. + Thus if neither <code>offset</code> nor <code>duration</code> are specified then the implied duration is + the total duration of the AudioBuffer. + </p> + + </dd> +</dl> +<dl> + <dt id="dfn-stop">The <code>stop</code> method</dt> + <dd><p>Schedules a sound to stop playback at an exact time.</p> + <p>The <dfn id="dfn-when_AudioBufferSourceNode_2">when</dfn> parameter + describes at what time (in seconds) the sound should stop playing. + It is in the same time coordinate system as AudioContext.currentTime. + If 0 is passed in for this value or if the value is less than + <b>currentTime</b>, then the sound will stop playing immediately. + <code>stop</code> must only be called one time and only after a call to <code>start</code> or <code>stop</code>, + or an exception will be thrown.</p> + </dd> +</dl> +</div> + +<div id="looping-AudioBufferSourceNode-section" class="section"> +<h3 id="looping-AudioBufferSourceNode">4.10.3. Looping</h3> +<p> +If the <code>loop</code> attribute is true when <code>start()</code> is called, then playback will continue indefinitely +until <code>stop()</code> is called and the stop time is reached. We'll call this "loop" mode. Playback always starts at the point in the buffer indicated +by the <code>offset</code> argument of <code>start()</code>, and in <em>loop</em> mode will continue playing until it reaches the <em>actualLoopEnd</em> position +in the buffer (or the end of the buffer), at which point it will wrap back around to the <em>actualLoopStart</em> position in the buffer, and continue +playing according to this pattern. +</p> + +<p> +In <em>loop</em> mode then the <em>actual</em> loop points are calculated as follows from the <code>loopStart</code> and <code>loopEnd</code> attributes: +</p> + +<blockquote> +<pre> + if ((loopStart || loopEnd) && loopStart >= 0 && loopEnd > 0 && loopStart < loopEnd) { + actualLoopStart = loopStart; + actualLoopEnd = min(loopEnd, buffer.length); + } else { + actualLoopStart = 0; + actualLoopEnd = buffer.length; + } +</pre> +</blockquote> + +<p> +Note that the default values for <code>loopStart</code> and <code>loopEnd</code> are both 0, which indicates that looping should occur from the very start +to the very end of the buffer. +</p> + +<p> +Please note that as a low-level implementation detail, the AudioBuffer is at a specific sample-rate (usually the same as the AudioContext sample-rate), and +that the loop times (in seconds) must be converted to the appropriate sample-frame positions in the buffer according to this sample-rate. +</p> + +</div> + +<div id="MediaElementAudioSourceNode-section" class="section"> +<h2 id="MediaElementAudioSourceNode">4.11. The MediaElementAudioSourceNode +Interface</h2> + +<p>This interface represents an audio source from an <code>audio</code> or +<code>video</code> element. </p> +<pre> numberOfInputs : 0 + numberOfOutputs : 1 + </pre> + +<p> +The number of channels of the output corresponds to the number of channels of the media referenced by the HTMLMediaElement. +Thus, changes to the media element's .src attribute can change the number of channels output by this node. +If the .src attribute is not set, then the number of channels output will be one silent channel. +</p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="media-element-audio-source-node-idl"> + +interface <dfn id="dfn-MediaElementAudioSourceNode">MediaElementAudioSourceNode</dfn> : AudioNode { + +}; +</code></pre> +</div> +</div> +</div> + +<p>A MediaElementAudioSourceNode +is created given an HTMLMediaElement using the AudioContext <a href="#dfn-createMediaElementSource">createMediaElementSource()</a> method. </p> + +<p> +The number of channels of the single output equals the number of channels of the audio referenced by +the HTMLMediaElement passed in as the argument to createMediaElementSource(), or is 1 if the HTMLMediaElement +has no audio. +</p> + +<p> +The HTMLMediaElement must behave in an identical fashion after the MediaElementAudioSourceNode has +been created, <em>except</em> that the rendered audio will no longer be heard directly, but instead will be heard +as a consequence of the MediaElementAudioSourceNode being connected through the routing graph. Thus pausing, seeking, +volume, <code>.src</code> attribute changes, and other aspects of the HTMLMediaElement must behave as they normally would +if <em>not</em> used with a MediaElementAudioSourceNode. +</p> + +<div class="example"> + +<div class="exampleHeader"> +Example</div> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">ECMAScript</span></div> + +<div class="blockContent"> +<pre class="code"><code class="es-code"> +var mediaElement = document.getElementById('mediaElementID'); +var sourceNode = context.createMediaElementSource(mediaElement); +sourceNode.connect(filterNode); + </code></pre> +</div> +</div> +</div> +</div> + + +<div id="ScriptProcessorNode-section" class="section"> +<h2 id="ScriptProcessorNode">4.12. The ScriptProcessorNode Interface</h2> + +<p>This interface is an AudioNode which can generate, process, or analyse audio +directly using JavaScript. </p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 + + channelCount = numberOfInputChannels; + channelCountMode = "explicit"; + channelInterpretation = "speakers"; +</pre> + +<p>The ScriptProcessorNode is constructed with a <code>bufferSize</code> which +must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. +This value controls how frequently the <code>audioprocess</code> event +is dispatched and how many sample-frames need to be processed each call. +Lower numbers for <code>bufferSize</code> will result in a lower (better) <a +href="#Latency-section">latency</a>. Higher numbers will be necessary to avoid +audio breakup and <a href="#Glitching-section">glitches</a>. +This value will be picked by the implementation if the bufferSize argument +to <code>createScriptProcessor</code> is not passed in, or is set to 0.</p> + +<p><code>numberOfInputChannels</code> and <code>numberOfOutputChannels</code> +determine the number of input and output channels. It is invalid for both +<code>numberOfInputChannels</code> and <code>numberOfOutputChannels</code> to +be zero. </p> +<pre> var node = context.createScriptProcessor(bufferSize, numberOfInputChannels, numberOfOutputChannels); + </pre> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="script-processor-node-idl"> + +interface <dfn id="dfn-ScriptProcessorNode">ScriptProcessorNode</dfn> : AudioNode { + + attribute EventHandler onaudioprocess; + + readonly attribute long bufferSize; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-ScriptProcessorNode-section" class="section"> +<h3 id="attributes-ScriptProcessorNode">4.12.1. Attributes</h3> +<dl> + <dt id="dfn-onaudioprocess"><code>onaudioprocess</code></dt> + <dd><p>A property used to set the <code>EventHandler</code> (described in <cite><a + href="http://www.whatwg.org/specs/web-apps/current-work/#eventhandler">HTML</a></cite>) + for the audioprocess event that is dispatched to <a + href="#ScriptProcessorNode-section"><code>ScriptProcessorNode</code></a> + node types. An event of type <a + href="#AudioProcessingEvent-section"><code>AudioProcessingEvent</code></a> + will be dispatched to the event handler. </p> + </dd> +</dl> +<dl> + <dt id="dfn-bufferSize_ScriptProcessorNode"><code>bufferSize</code></dt> + <dd><p>The size of the buffer (in sample-frames) which needs to be + processed each time <code>onprocessaudio</code> is called. Legal values + are (256, 512, 1024, 2048, 4096, 8192, 16384). </p> + </dd> +</dl> +</div> +</div> + +<div id="AudioProcessingEvent-section" class="section"> +<h2 id="AudioProcessingEvent">4.13. The AudioProcessingEvent Interface</h2> + +<p>This is an <code>Event</code> object which is dispatched to <a +href="#ScriptProcessorNode-section"><code>ScriptProcessorNode</code></a> nodes. </p> + +<p>The event handler processes audio from the input (if any) by accessing the +audio data from the <code>inputBuffer</code> attribute. The audio data which is +the result of the processing (or the synthesized data if there are no inputs) +is then placed into the <code>outputBuffer</code>. </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="audio-processing-event-idl"> + +interface <dfn id="dfn-AudioProcessingEvent">AudioProcessingEvent</dfn> : Event { + + readonly attribute double playbackTime; + readonly attribute AudioBuffer inputBuffer; + readonly attribute AudioBuffer outputBuffer; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-AudioProcessingEvent-section" class="section"> +<h3 id="attributes-AudioProcessingEvent">4.13.1. Attributes</h3> +<dl> + <dt id="dfn-playbackTime"><code>playbackTime</code></dt> + <dd><p>The time when the audio will be played in the same time coordinate system as AudioContext.currentTime. + <code>playbackTime</code> allows for very tight synchronization between + processing directly in JavaScript with the other events in the context's + rendering graph. </p> + </dd> +</dl> +<dl> + <dt id="dfn-inputBuffer"><code>inputBuffer</code></dt> + <dd><p>An AudioBuffer containing the input audio data. It will have a number of channels equal to the <code>numberOfInputChannels</code> parameter + of the createScriptProcessor() method. This AudioBuffer is only valid while in the scope of the <code>onaudioprocess</code> + function. Its values will be meaningless outside of this scope.</p> + </dd> +</dl> +<dl> + <dt id="dfn-outputBuffer"><code>outputBuffer</code></dt> + <dd><p>An AudioBuffer where the output audio data should be written. It will have a number of channels equal to the + <code>numberOfOutputChannels</code> parameter of the createScriptProcessor() method. + Script code within the scope of the <code>onaudioprocess</code> function is expected to modify the + <code>Float32Array</code> arrays representing channel data in this AudioBuffer. + Any script modifications to this AudioBuffer outside of this scope will not produce any audible effects.</p> + </dd> +</dl> +</div> +</div> + +<div id="PannerNode-section" class="section"> +<h2 id="PannerNode">4.14. The PannerNode Interface</h2> + +<p>This interface represents a processing node which <a +href="#Spatialization-section">positions / spatializes</a> an incoming audio +stream in three-dimensional space. The spatialization is in relation to the <a +href="#AudioContext-section"><code>AudioContext</code></a>'s <a +href="#AudioListener-section"><code>AudioListener</code></a> +(<code>listener</code> attribute). </p> + +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 + + channelCount = 2; + channelCountMode = "clamped-max"; + channelInterpretation = "speakers"; +</pre> + +<p> +The audio stream from the input will be either mono or stereo, depending on the connection(s) to the input. +</p> + +<p> +The output of this node is hard-coded to stereo (2 channels) and <em>currently</em> cannot be configured. +</p> + + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="panner-node-idl"> + +enum <dfn>PanningModelType</dfn> { + "equalpower", + "HRTF" +}; + +enum <dfn>DistanceModelType</dfn> { + "linear", + "inverse", + "exponential" +}; + +interface <dfn id="dfn-PannerNode">PannerNode</dfn> : AudioNode { + + <span class="comment">// Default for stereo is HRTF </span> + attribute PanningModelType panningModel; + + <span class="comment">// Uses a 3D cartesian coordinate system </span> + void setPosition(double x, double y, double z); + void setOrientation(double x, double y, double z); + void setVelocity(double x, double y, double z); + + <span class="comment">// Distance model and attributes </span> + attribute DistanceModelType distanceModel; + attribute double refDistance; + attribute double maxDistance; + attribute double rolloffFactor; + + <span class="comment">// Directional sound cone </span> + attribute double coneInnerAngle; + attribute double coneOuterAngle; + attribute double coneOuterGain; + +}; +</code></pre> +</div> +</div> +</div> + +<div id="attributes-PannerNode_attributes-section" class="section"> +<h3 id="attributes-PannerNode_attributes">4.14.2. Attributes</h3> +<dl> + <dt id="dfn-panningModel"><code>panningModel</code></dt> + <dd><p>Determines which spatialization algorithm will be used to position + the audio in 3D space. The default is "HRTF". </p> + + <dl> + <dt id="dfn-EQUALPOWER"><code>"equalpower"</code></dt> + <dd><p>A simple and efficient spatialization algorithm using equal-power + panning. </p> + </dd> + </dl> + <dl> + <dt id="dfn-HRTF"><code>"HRTF"</code></dt> + <dd><p>A higher quality spatialization algorithm using a convolution with + measured impulse responses from human subjects. This panning method + renders stereo output. </p> + </dd> + </dl> + </dd> +</dl> +<dl> + <dt id="dfn-distanceModel"><code>distanceModel</code></dt> + <dd><p>Determines which algorithm will be used to reduce the volume of an + audio source as it moves away from the listener. The default is "inverse". +</p> + +<dl> + <dt id="dfn-LINEAR_DISTANCE"><code>"linear"</code></dt> + <dd><p>A linear distance model which calculates <em>distanceGain</em> according to: </p> + <pre> +1 - rolloffFactor * (distance - refDistance) / (maxDistance - refDistance) + </pre> + </dd> +</dl> +<dl> + <dt id="dfn-INVERSE_DISTANCE"><code>"inverse"</code></dt> + <dd><p>An inverse distance model which calculates <em>distanceGain</em> according to: </p> + <pre> +refDistance / (refDistance + rolloffFactor * (distance - refDistance)) + </pre> + </dd> +</dl> +<dl> + <dt id="dfn-EXPONENTIAL_DISTANCE"><code>"exponential"</code></dt> + <dd><p>An exponential distance model which calculates <em>distanceGain</em> according to: </p> + <pre> +pow(distance / refDistance, -rolloffFactor) + </pre> + </dd> +</dl> + + + </dd> +</dl> +<dl> + <dt id="dfn-refDistance"><code>refDistance</code></dt> + <dd><p>A reference distance for reducing volume as source move further from + the listener. The default value is 1. </p> + </dd> +</dl> +<dl> + <dt id="dfn-maxDistance"><code>maxDistance</code></dt> + <dd><p>The maximum distance between source and listener, after which the + volume will not be reduced any further. The default value is 10000. </p> + </dd> +</dl> +<dl> + <dt id="dfn-rolloffFactor"><code>rolloffFactor</code></dt> + <dd><p>Describes how quickly the volume is reduced as source moves away + from listener. The default value is 1. </p> + </dd> +</dl> +<dl> + <dt id="dfn-coneInnerAngle"><code>coneInnerAngle</code></dt> + <dd><p>A parameter for directional audio sources, this is an angle, inside + of which there will be no volume reduction. The default value is 360. </p> + </dd> +</dl> +<dl> + <dt id="dfn-coneOuterAngle"><code>coneOuterAngle</code></dt> + <dd><p>A parameter for directional audio sources, this is an angle, outside + of which the volume will be reduced to a constant value of + <b>coneOuterGain</b>. The default value is 360. </p> + </dd> +</dl> +<dl> + <dt id="dfn-coneOuterGain"><code>coneOuterGain</code></dt> + <dd><p>A parameter for directional audio sources, this is the amount of + volume reduction outside of the <b>coneOuterAngle</b>. The default value is 0. </p> + </dd> +</dl> +</div> + +<h3 id="Methods_and_Parameters">4.14.3. Methods and Parameters</h3> +<dl> + <dt id="dfn-setPosition">The <code>setPosition</code> method</dt> + <dd><p>Sets the position of the audio source relative to the + <b>listener</b> attribute. A 3D cartesian coordinate system is used.</p> + <p>The <dfn id="dfn-x">x, y, z</dfn> parameters represent the coordinates + in 3D space. </p> + <p>The default value is (0,0,0) + </p> + </dd> +</dl> +<dl> + <dt id="dfn-setOrientation">The <code>setOrientation</code> method</dt> + <dd><p>Describes which direction the audio source is pointing in the 3D + cartesian coordinate space. Depending on how directional the sound is + (controlled by the <b>cone</b> attributes), a sound pointing away from + the listener can be very quiet or completely silent.</p> + <p>The <dfn id="dfn-x_2">x, y, z</dfn> parameters represent a direction + vector in 3D space. </p> + <p>The default value is (1,0,0) + </p> + </dd> +</dl> +<dl> + <dt id="dfn-setVelocity">The <code>setVelocity</code> method</dt> + <dd><p>Sets the velocity vector of the audio source. This vector controls + both the direction of travel and the speed in 3D space. This velocity + relative to the listener's velocity is used to determine how much doppler + shift (pitch change) to apply. The units used for this vector is <em>meters / second</em> + and is independent of the units used for position and orientation vectors.</p> + <p>The <dfn id="dfn-x_3">x, y, z</dfn> parameters describe a direction + vector indicating direction of travel and intensity. </p> + <p>The default value is (0,0,0) + </p> + </dd> +</dl> + +<div id="AudioListener-section" class="section"> +<h2 id="AudioListener">4.15. The AudioListener Interface</h2> + +<p>This interface represents the position and orientation of the person +listening to the audio scene. All <a +href="#PannerNode-section"><code>PannerNode</code></a> objects +spatialize in relation to the AudioContext's <code>listener</code>. See <a +href="#Spatialization-section">this</a> section for more details about +spatialization. </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="audio-listener-idl"> + +interface <dfn id="dfn-AudioListener">AudioListener</dfn> { + + attribute double dopplerFactor; + attribute double speedOfSound; + + <span class="comment">// Uses a 3D cartesian coordinate system </span> + void setPosition(double x, double y, double z); + void setOrientation(double x, double y, double z, double xUp, double yUp, double zUp); + void setVelocity(double x, double y, double z); + +}; +</code></pre> +</div> +</div> +</div> + +<div id="attributes-AudioListener-section" class="section"> +<h3 id="attributes-AudioListener">4.15.1. Attributes</h3> +<dl> + <dt id="dfn-dopplerFactor"><code>dopplerFactor</code></dt> + <dd><p>A constant used to determine the amount of pitch shift to use when + rendering a doppler effect. The default value is 1. </p> + </dd> +</dl> +<dl> + <dt id="dfn-speedOfSound"><code>speedOfSound</code></dt> + <dd><p>The speed of sound used for calculating doppler shift. The default + value is 343.3. </p> + </dd> +</dl> +</div> + +<h3 id="L15842">4.15.2. Methods and Parameters</h3> +<dl> + <dt id="dfn-setPosition_2">The <code>setPosition</code> method</dt> + <dd><p>Sets the position of the listener in a 3D cartesian coordinate + space. <code>PannerNode</code> objects use this position relative to + individual audio sources for spatialization.</p> + <p>The <dfn id="dfn-x_AudioListener">x, y, z</dfn> parameters represent + the coordinates in 3D space. </p> + <p>The default value is (0,0,0) + </p> + </dd> +</dl> +<dl> + <dt id="dfn-setOrientation_2">The <code>setOrientation</code> method</dt> + <dd><p>Describes which direction the listener is pointing in the 3D + cartesian coordinate space. Both a <b>front</b> vector and an <b>up</b> + vector are provided. In simple human terms, the <b>front</b> vector represents which + direction the person's nose is pointing. The <b>up</b> vector represents the + direction the top of a person's head is pointing. These values are expected to + be linearly independent (at right angles to each other). For normative requirements + of how these values are to be interpreted, see the + <a href="#Spatialization-section">spatialization section</a>. + </p> + <p>The <dfn id="dfn-x_setOrientation">x, y, z</dfn> parameters represent + a <b>front</b> direction vector in 3D space, with the default value being (0,0,-1) </p> + <p>The <dfn id="dfn-x_setOrientation_2">xUp, yUp, zUp</dfn> parameters + represent an <b>up</b> direction vector in 3D space, with the default value being (0,1,0) </p> + </dd> +</dl> +<dl> + <dt id="dfn-setVelocity_4">The <code>setVelocity</code> method</dt> + <dd><p>Sets the velocity vector of the listener. This vector controls both + the direction of travel and the speed in 3D space. This velocity relative to + an audio source's velocity is used to determine how much doppler shift + (pitch change) to apply. The units used for this vector is <em>meters / second</em> + and is independent of the units used for position and orientation vectors.</p> + <p>The <dfn id="dfn-x_setVelocity_5">x, y, z</dfn> parameters describe a + direction vector indicating direction of travel and intensity. </p> + <p>The default value is (0,0,0) + </p> + </dd> +</dl> + +<div id="ConvolverNode-section" class="section"> +<h2 id="ConvolverNode">4.16. The ConvolverNode Interface</h2> + +<p>This interface represents a processing node which applies a <a +href="#Convolution-section">linear convolution effect</a> given an impulse +response. Normative requirements for multi-channel convolution matrixing are described +<a href="#Convolution-reverb-effect">here</a>. </p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 + + channelCount = 2; + channelCountMode = "clamped-max"; + channelInterpretation = "speakers"; +</pre> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="convolver-node-idl"> + +interface <dfn id="dfn-ConvolverNode">ConvolverNode</dfn> : AudioNode { + + attribute AudioBuffer? buffer; + attribute boolean normalize; + +}; +</code></pre> +</div> +</div> +</div> + +<div id="attributes-ConvolverNode-section" class="section"> +<h3 id="attributes-ConvolverNode">4.16.1. Attributes</h3> +<dl> + <dt id="dfn-buffer_ConvolverNode"><code>buffer</code></dt> + <dd><p>A mono, stereo, or 4-channel <code>AudioBuffer</code> containing the (possibly multi-channel) impulse response + used by the ConvolverNode. This <code>AudioBuffer</code> must be of the same sample-rate as the AudioContext or an exception will + be thrown. At the time when this attribute is set, the <em>buffer</em> and the state of the <em>normalize</em> + attribute will be used to configure the ConvolverNode with this impulse response having the given normalization. + The initial value of this attribute is null.</p> + </dd> +</dl> +<dl> + <dt id="dfn-normalize"><code>normalize</code></dt> + <dd><p>Controls whether the impulse response from the buffer will be scaled + by an equal-power normalization when the <code>buffer</code> atttribute + is set. Its default value is <code>true</code> in order to achieve a more + uniform output level from the convolver when loaded with diverse impulse + responses. If <code>normalize</code> is set to <code>false</code>, then + the convolution will be rendered with no pre-processing/scaling of the + impulse response. Changes to this value do not take effect until the next time + the <em>buffer</em> attribute is set. </p> + + </dd> +</dl> + + <p> + If the <em>normalize</em> attribute is false when the <em>buffer</em> attribute is set then the + ConvolverNode will perform a linear convolution given the exact impulse response contained within the <em>buffer</em>. + </p> + <p> + Otherwise, if the <em>normalize</em> attribute is true when the <em>buffer</em> attribute is set then the + ConvolverNode will first perform a scaled RMS-power analysis of the audio data contained within <em>buffer</em> to calculate a + <em>normalizationScale</em> given this algorithm: + </p> + + + <div class="block"> + + <div class="blockTitleDiv"> + + <div class="blockContent"> + <pre class="code"><code class="es-code"> + +float calculateNormalizationScale(buffer) +{ + const float GainCalibration = 0.00125; + const float GainCalibrationSampleRate = 44100; + const float MinPower = 0.000125; + + // Normalize by RMS power. + size_t numberOfChannels = buffer->numberOfChannels(); + size_t length = buffer->length(); + + float power = 0; + + for (size_t i = 0; i < numberOfChannels; ++i) { + float* sourceP = buffer->channel(i)->data(); + float channelPower = 0; + + int n = length; + while (n--) { + float sample = *sourceP++; + channelPower += sample * sample; + } + + power += channelPower; + } + + power = sqrt(power / (numberOfChannels * length)); + + // Protect against accidental overload. + if (isinf(power) || isnan(power) || power < MinPower) + power = MinPower; + + float scale = 1 / power; + + // Calibrate to make perceived volume same as unprocessed. + scale *= GainCalibration; + + // Scale depends on sample-rate. + if (buffer->sampleRate()) + scale *= GainCalibrationSampleRate / buffer->sampleRate(); + + // True-stereo compensation. + if (buffer->numberOfChannels() == 4) + scale *= 0.5; + + return scale; +} + </code></pre> + + </div> + </div> + </div> + +<p> +During processing, the ConvolverNode will then take this calculated <em>normalizationScale</em> value and multiply it by the result of the linear convolution +resulting from processing the input with the impulse response (represented by the <em>buffer</em>) to produce the +final output. Or any mathematically equivalent operation may be used, such as pre-multiplying the +input by <em>normalizationScale</em>, or pre-multiplying a version of the impulse-response by <em>normalizationScale</em>. +</p> + +</div> + +<div id="AnalyserNode-section" class="section"> +<h2 id="AnalyserNode">4.17. The AnalyserNode Interface</h2> + +<p>This interface represents a node which is able to provide real-time +frequency and time-domain <a href="#AnalyserNode">analysis</a> +information. The audio stream will be passed un-processed from input to output. +</p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 <em>Note that this output may be left unconnected.</em> + + channelCount = 1; + channelCountMode = "explicit"; + channelInterpretation = "speakers"; +</pre> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="analyser-node-idl"> + +interface <dfn id="dfn-AnalyserNode">AnalyserNode</dfn> : AudioNode { + + <span class="comment">// Real-time frequency-domain data </span> + void getFloatFrequencyData(Float32Array array); + void getByteFrequencyData(Uint8Array array); + + <span class="comment">// Real-time waveform data </span> + void getByteTimeDomainData(Uint8Array array); + + attribute unsigned long fftSize; + readonly attribute unsigned long frequencyBinCount; + + attribute double minDecibels; + attribute double maxDecibels; + + attribute double smoothingTimeConstant; + +}; +</code></pre> +</div> +</div> +</div> + +<div id="attributes-ConvolverNode-section_2" class="section"> +<h3 id="attributes-ConvolverNode_2">4.17.1. Attributes</h3> +<dl> + <dt id="dfn-fftSize"><code>fftSize</code></dt> + <dd><p>The size of the FFT used for frequency-domain analysis. This must be + a non-zero power of two in the range 32 to 2048, otherwise an INDEX_SIZE_ERR exception MUST be thrown. + The default value is 2048.</p> + </dd> +</dl> +<dl> + <dt id="dfn-frequencyBinCount"><code>frequencyBinCount</code></dt> + <dd><p>Half the FFT size. </p> + </dd> +</dl> +<dl> + <dt id="dfn-minDecibels"><code>minDecibels</code></dt> + <dd><p>The minimum power value in the scaling range for the FFT analysis + data for conversion to unsigned byte values. + The default value is -100. + If the value of this attribute is set to a value more than or equal to <code>maxDecibels</code>, + an INDEX_SIZE_ERR exception MUST be thrown.</p> + </dd> +</dl> +<dl> + <dt id="dfn-maxDecibels"><code>maxDecibels</code></dt> + <dd><p>The maximum power value in the scaling range for the FFT analysis + data for conversion to unsigned byte values. + The default value is -30. + If the value of this attribute is set to a value less than or equal to <code>minDecibels</code>, + an INDEX_SIZE_ERR exception MUST be thrown.</p> + </dd> +</dl> +<dl> + <dt id="dfn-smoothingTimeConstant"><code>smoothingTimeConstant</code></dt> + <dd><p>A value from 0 -> 1 where 0 represents no time averaging + with the last analysis frame. + The default value is 0.8. + If the value of this attribute is set to a value less than 0 or more than 1, + an INDEX_SIZE_ERR exception MUST be thrown.</p> + </dd> +</dl> +</div> + +<h3 id="methods-and-parameters">4.17.2. Methods and Parameters</h3> +<dl> + <dt id="dfn-getFloatFrequencyData">The <code>getFloatFrequencyData</code> + method</dt> + <dd><p>Copies the current frequency data into the passed floating-point + array. If the array has fewer elements than the frequencyBinCount, the + excess elements will be dropped. If the array has more elements than + the frequencyBinCount, the excess elements will be ignored.</p> + <p>The <dfn id="dfn-array">array</dfn> parameter is where + frequency-domain analysis data will be copied. </p> + </dd> +</dl> +<dl> + <dt id="dfn-getByteFrequencyData">The <code>getByteFrequencyData</code> + method</dt> + <dd><p>Copies the current frequency data into the passed unsigned byte + array. If the array has fewer elements than the frequencyBinCount, the + excess elements will be dropped. If the array has more elements than + the frequencyBinCount, the excess elements will be ignored.</p> + <p>The <dfn id="dfn-array_2">array</dfn> parameter is where + frequency-domain analysis data will be copied. </p> + </dd> +</dl> +<dl> + <dt id="dfn-getByteTimeDomainData">The <code>getByteTimeDomainData</code> + method</dt> + <dd><p>Copies the current time-domain (waveform) data into the passed + unsigned byte array. If the array has fewer elements than the + fftSize, the excess elements will be dropped. If the array has more + elements than fftSize, the excess elements will be ignored.</p> + <p>The <dfn id="dfn-array_3">array</dfn> parameter is where time-domain + analysis data will be copied. </p> + </dd> +</dl> + +<div id="ChannelSplitterNode-section" class="section"> +<h2 id="ChannelSplitterNode">4.18. The ChannelSplitterNode Interface</h2> + +<p>The <code>ChannelSplitterNode</code> is for use in more advanced +applications and would often be used in conjunction with <a +href="#ChannelMergerNode-section"><code>ChannelMergerNode</code></a>. </p> +<pre> + numberOfInputs : 1 + numberOfOutputs : Variable N (defaults to 6) // number of "active" (non-silent) outputs is determined by number of channels in the input + + channelCountMode = "max"; + channelInterpretation = "speakers"; +</pre> + +<p>This interface represents an AudioNode for accessing the individual channels +of an audio stream in the routing graph. It has a single input, and a number of +"active" outputs which equals the number of channels in the input audio stream. +For example, if a stereo input is connected to an +<code>ChannelSplitterNode</code> then the number of active outputs will be two +(one from the left channel and one from the right). There are always a total +number of N outputs (determined by the <code>numberOfOutputs</code> parameter to the AudioContext method <code>createChannelSplitter()</code>), + The default number is 6 if this value is not provided. Any outputs +which are not "active" will output silence and would typically not be connected +to anything. </p> + +<h3 id="example-1">Example:</h3> +<img alt="channel splitter" src="images/channel-splitter.png" /> + +<p>Please note that in this example, the splitter does <b>not</b> interpret the channel identities (such as left, right, etc.), but +simply splits out channels in the order that they are input.</p> + +<p>One application for <code>ChannelSplitterNode</code> is for doing "matrix +mixing" where individual gain control of each channel is desired. </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="channel-splitter-node-idl"> + +interface <dfn id="dfn-ChannelSplitterNode">ChannelSplitterNode</dfn> : AudioNode { + +}; +</code></pre> +</div> +</div> +</div> + +<div id="ChannelMergerNode-section" class="section"> +<h2 id="ChannelMergerNode">4.19. The ChannelMergerNode Interface</h2> + +<p>The <code>ChannelMergerNode</code> is for use in more advanced applications +and would often be used in conjunction with <a +href="#ChannelSplitterNode-section"><code>ChannelSplitterNode</code></a>. </p> +<pre> + numberOfInputs : Variable N (default to 6) // number of connected inputs may be less than this + numberOfOutputs : 1 + + channelCountMode = "max"; + channelInterpretation = "speakers"; +</pre> + +<p>This interface represents an AudioNode for combining channels from multiple +audio streams into a single audio stream. It has a variable number of inputs (defaulting to 6), but not all of them +need be connected. There is a single output whose audio stream has a number of +channels equal to the sum of the numbers of channels of all the connected +inputs. For example, if an <code>ChannelMergerNode</code> has two connected +inputs (both stereo), then the output will be four channels, the first two from +the first input and the second two from the second input. In another example +with two connected inputs (both mono), the output will be two channels +(stereo), with the left channel coming from the first input and the right +channel coming from the second input. </p> + +<h3 id="example-2">Example:</h3> +<img alt="channel merger" src="images/channel-merger.png" /> + +<p>Please note that in this example, the merger does <b>not</b> interpret the channel identities (such as left, right, etc.), but +simply combines channels in the order that they are input.</p> + + +<p>Be aware that it is possible to connect an <code>ChannelMergerNode</code> +in such a way that it outputs an audio stream with a large number of channels +greater than the maximum supported by the audio hardware. In this case where such an output is connected +to the AudioContext .destination (the audio hardware), then the extra channels will be ignored. +Thus, the <code>ChannelMergerNode</code> should be used in situations where the number +of channels is well understood. </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="channel-merger-node-idl"> + +interface <dfn id="dfn-ChannelMergerNode">ChannelMergerNode</dfn> : AudioNode { + +}; +</code></pre> +</div> +</div> +</div> + +<div id="DynamicsCompressorNode-section" class="section"> +<h2 id="DynamicsCompressorNode">4.20. The DynamicsCompressorNode Interface</h2> + +<p>DynamicsCompressorNode is an AudioNode processor implementing a dynamics +compression effect. </p> + +<p>Dynamics compression is very commonly used in musical production and game +audio. It lowers the volume of the loudest parts of the signal and raises the +volume of the softest parts. Overall, a louder, richer, and fuller sound can be +achieved. It is especially important in games and musical applications where +large numbers of individual sounds are played simultaneous to control the +overall signal level and help avoid clipping (distorting) the audio output to +the speakers. </p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 + + channelCount = 2; + channelCountMode = "explicit"; + channelInterpretation = "speakers"; +</pre> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="dynamics-compressor-node-idl"> + +interface <dfn id="dfn-DynamicsCompressorNode">DynamicsCompressorNode</dfn> : AudioNode { + + readonly attribute AudioParam threshold; // in Decibels + readonly attribute AudioParam knee; // in Decibels + readonly attribute AudioParam ratio; // unit-less + readonly attribute AudioParam reduction; // in Decibels + readonly attribute AudioParam attack; // in Seconds + readonly attribute AudioParam release; // in Seconds + +}; +</code> +</pre> +</div> +</div> + +<div id="attributes-DynamicsCompressorNode-section" class="section"> +<h3 id="attributes-DynamicsCompressorNode">4.20.1. Attributes</h3> +<p> +All parameters are <em>k-rate</em> +</p> + +<dl> + <dt id="dfn-threshold"><code>threshold</code></dt> + <dd><p>The decibel value above which the compression will start taking + effect. Its default value is -24, with a nominal range of -100 to 0. </p> + </dd> +</dl> +<dl> + <dt id="dfn-knee"><code>knee</code></dt> + <dd><p>A decibel value representing the range above the threshold where the + curve smoothly transitions to the "ratio" portion. Its default value is 30, with a nominal range of 0 to 40. </p> + </dd> +</dl> +<dl> + <dt id="dfn-ratio"><code>ratio</code></dt> + <dd><p>The amount of dB change in input for a 1 dB change in output. Its default value is 12, with a nominal range of 1 to 20. </p> + </dd> +</dl> +<dl> + <dt id="dfn-reduction"><code>reduction</code></dt> + <dd><p>A read-only decibel value for metering purposes, representing the + current amount of gain reduction that the compressor is applying to the + signal. If fed no signal the value will be 0 (no gain reduction). The nominal range is -20 to 0. </p> + </dd> +</dl> +<dl> + <dt id="dfn-attack"><code>attack</code></dt> + <dd><p>The amount of time (in seconds) to reduce the gain by 10dB. Its default value is 0.003, with a nominal range of 0 to 1. </p> + </dd> +</dl> +<dl> + <dt id="dfn-release"><code>release</code></dt> + <dd><p>The amount of time (in seconds) to increase the gain by 10dB. Its default value is 0.250, with a nominal range of 0 to 1. </p> + </dd> +</dl> +</div> +</div> + +<div id="BiquadFilterNode-section" class="section"> +<h2 id="BiquadFilterNode">4.21. The BiquadFilterNode Interface</h2> + +<p>BiquadFilterNode is an AudioNode processor implementing very common +low-order filters. </p> + +<p>Low-order filters are the building blocks of basic tone controls (bass, mid, +treble), graphic equalizers, and more advanced filters. Multiple +BiquadFilterNode filters can be combined to form more complex filters. The +filter parameters such as "frequency" can be changed over time for filter +sweeps, etc. Each BiquadFilterNode can be configured as one of a number of +common filter types as shown in the IDL below. The default filter type +is "lowpass".</p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 + + channelCountMode = "max"; + channelInterpretation = "speakers"; +</pre> +<p> +The number of channels of the output always equals the number of channels of the input. +</p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="biquad-filter-node-idl"> + +enum <dfn>BiquadFilterType</dfn> { + "lowpass", + "highpass", + "bandpass", + "lowshelf", + "highshelf", + "peaking", + "notch", + "allpass" +}; + +interface <dfn id="dfn-BiquadFilterNode">BiquadFilterNode</dfn> : AudioNode { + + attribute BiquadFilterType type; + readonly attribute AudioParam frequency; // in Hertz + readonly attribute AudioParam detune; // in Cents + readonly attribute AudioParam Q; // Quality factor + readonly attribute AudioParam gain; // in Decibels + + void getFrequencyResponse(Float32Array frequencyHz, + Float32Array magResponse, + Float32Array phaseResponse); + +}; +</code></pre> +</div> +</div> +</div> + +<p>The filter types are briefly described below. We note that all of these +filters are very commonly used in audio processing. In terms of implementation, +they have all been derived from standard analog filter prototypes. For more +technical details, we refer the reader to the excellent <a +href="http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt">reference</a> by +Robert Bristow-Johnson.</p> + +<p> +All parameters are <em>k-rate</em> with the following default parameter values: +</p> + +<blockquote> +<dl> + <dt>frequency</dt> + <dd>350Hz, with a nominal range of 10 to the Nyquist frequency (half the sample-rate). + </dd> + <dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt> + <dd>1, with a nominal range of 0.0001 to 1000.</dd> + <dt>gain</dt> + <dd>0, with a nominal range of -40 to 40.</dd> +</dl> +</blockquote> + + + +<div id="BiquadFilterNode-description-section" class="section"> +<h3 id="BiquadFilterNode-description">4.21.1 "lowpass"</h3> + +<p>A <a href="http://en.wikipedia.org/wiki/Low-pass_filter">lowpass filter</a> +allows frequencies below the cutoff frequency to pass through and attenuates +frequencies above the cutoff. It implements a standard second-order +resonant lowpass filter with 12dB/octave rolloff.</p> + +<blockquote> + <dl> + <dt>frequency</dt> + <dd>The cutoff frequency</dd> + <dt>Q</dt> + <dd>Controls how peaked the response will be at the cutoff frequency. A + large value makes the response more peaked. Please note that for this filter type, this + value is not a traditional Q, but is a resonance value in decibels.</dd> + <dt>gain</dt> + <dd>Not used in this filter type</dd> + </dl> +</blockquote> + +<h3 id="HIGHPASS">4.21.2 "highpass"</h3> + +<p>A <a href="http://en.wikipedia.org/wiki/High-pass_filter">highpass +filter</a> is the opposite of a lowpass filter. Frequencies above the cutoff +frequency are passed through, but frequencies below the cutoff are attenuated. +It implements a standard second-order resonant highpass filter with +12dB/octave rolloff.</p> + +<blockquote> + <dl> + <dt>frequency</dt> + <dd>The cutoff frequency below which the frequencies are attenuated</dd> + <dt>Q</dt> + <dd>Controls how peaked the response will be at the cutoff frequency. A + large value makes the response more peaked. Please note that for this filter type, this + value is not a traditional Q, but is a resonance value in decibels.</dd> + <dt>gain</dt> + <dd>Not used in this filter type</dd> + </dl> +</blockquote> + +<h3 id="BANDPASS">4.21.3 "bandpass"</h3> + +<p>A <a href="http://en.wikipedia.org/wiki/Band-pass_filter">bandpass +filter</a> allows a range of frequencies to pass through and attenuates the +frequencies below and above this frequency range. It implements a +second-order bandpass filter.</p> + +<blockquote> + <dl> + <dt>frequency</dt> + <dd>The center of the frequency band</dd> + <dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt> + <dd>Controls the width of the band. The width becomes narrower as the Q + value increases.</dd> + <dt>gain</dt> + <dd>Not used in this filter type</dd> + </dl> +</blockquote> + +<h3 id="LOWSHELF">4.21.4 "lowshelf"</h3> + +<p>The lowshelf filter allows all frequencies through, but adds a boost (or +attenuation) to the lower frequencies. It implements a second-order +lowshelf filter.</p> + +<blockquote> + <dl> + <dt>frequency</dt> + <dd>The upper limit of the frequences where the boost (or attenuation) is + applied.</dd> + <dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt> + <dd>Not used in this filter type.</dd> + <dt>gain</dt> + <dd>The boost, in dB, to be applied. If the value is negative, the + frequencies are attenuated.</dd> + </dl> +</blockquote> + +<h3 id="L16352">4.21.5 "highshelf"</h3> + +<p>The highshelf filter is the opposite of the lowshelf filter and allows all +frequencies through, but adds a boost to the higher frequencies. It +implements a second-order highshelf filter</p> + +<blockquote> + <dl> + <dt>frequency</dt> + <dd>The lower limit of the frequences where the boost (or attenuation) is + applied.</dd> + <dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt> + <dd>Not used in this filter type.</dd> + <dt>gain</dt> + <dd>The boost, in dB, to be applied. If the value is negative, the + frequencies are attenuated.</dd> + </dl> +</blockquote> + +<h3 id="PEAKING">4.21.6 "peaking"</h3> + +<p>The peaking filter allows all frequencies through, but adds a boost (or +attenuation) to a range of frequencies. </p> + +<blockquote> + <dl> + <dt>frequency</dt> + <dd>The center frequency of where the boost is applied.</dd> + <dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt> + <dd>Controls the width of the band of frequencies that are boosted. A + large value implies a narrow width.</dd> + <dt>gain</dt> + <dd>The boost, in dB, to be applied. If the value is negative, the + frequencies are attenuated.</dd> + </dl> +</blockquote> + +<h3 id="NOTCH">4.21.7 "notch"</h3> + +<p>The notch filter (also known as a <a +href="http://en.wikipedia.org/wiki/Band-stop_filter">band-stop or +band-rejection filter</a>) is the opposite of a bandpass filter. It allows all +frequencies through, except for a set of frequencies.</p> + +<blockquote> + <dl> + <dt>frequency</dt> + <dd>The center frequency of where the notch is applied.</dd> + <dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt> + <dd>Controls the width of the band of frequencies that are attenuated. A + large value implies a narrow width.</dd> + <dt>gain</dt> + <dd>Not used in this filter type.</dd> + </dl> +</blockquote> + +<h3 id="ALLPASS">4.21.8 "allpass"</h3> + +<p>An <a +href="http://en.wikipedia.org/wiki/All-pass_filter#Digital_Implementation">allpass +filter</a> allows all frequencies through, but changes the phase relationship +between the various frequencies. It implements a second-order allpass +filter</p> + +<blockquote> + <dl> + <dt>frequency</dt> + <dd>The frequency where the center of the phase transition occurs. Viewed + another way, this is the frequency with maximal <a + href="http://en.wikipedia.org/wiki/Group_delay">group delay</a>.</dd> + <dt><a href="http://en.wikipedia.org/wiki/Q_factor">Q</a></dt> + <dd>Controls how sharp the phase transition is at the center frequency. A + larger value implies a sharper transition and a larger group delay.</dd> + <dt>gain</dt> + <dd>Not used in this filter type.</dd> + </dl> +</blockquote> + +<h3 id="Methods">4.21.9. Methods</h3> +<dl> + <dt id="dfn-getFrequencyResponse">The <code>getFrequencyResponse</code> + method</dt> + <dd><p>Given the current filter parameter settings, calculates the + frequency response for the specified frequencies. </p> + <p>The <dfn id="dfn-frequencyHz">frequencyHz</dfn> parameter specifies an + array of frequencies at which the response values will be calculated.</p> + <p>The <dfn id="dfn-magResponse">magResponse</dfn> parameter specifies an + output array receiving the linear magnitude response values.</p> + <p>The <dfn id="dfn-phaseResponse">phaseResponse</dfn> parameter + specifies an output array receiving the phase response values in + radians.</p> + </dd> +</dl> +</div> + +<div id="WaveShaperNode-section" class="section"> +<h2 id="WaveShaperNode">4.22. The WaveShaperNode Interface</h2> + +<p>WaveShaperNode is an AudioNode processor implementing non-linear distortion +effects. </p> + +<p>Non-linear waveshaping distortion is commonly used for both subtle +non-linear warming, or more obvious distortion effects. Arbitrary non-linear +shaping curves may be specified.</p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 1 + + channelCountMode = "max"; + channelInterpretation = "speakers"; +</pre> + +<p> +The number of channels of the output always equals the number of channels of the input. +</p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="wave-shaper-node-idl"> + +enum <dfn>OverSampleType</dfn> { + "none", + "2x", + "4x" +}; + +interface <dfn id="dfn-WaveShaperNode">WaveShaperNode</dfn> : AudioNode { + + attribute Float32Array? curve; + attribute OverSampleType oversample; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-WaveShaperNode-section" class="section"> +<h3 id="attributes-WaveShaperNode">4.22.1. Attributes</h3> +<dl> + <dt id="dfn-curve"><code>curve</code></dt> + <dd><p>The shaping curve used for the waveshaping effect. The input signal + is nominally within the range -1 -> +1. Each input sample within this + range will index into the shaping curve with a signal level of zero + corresponding to the center value of the curve array. Any sample value + less than -1 will correspond to the first value in the curve array. Any + sample value greater than +1 will correspond to the last value in + the curve array. The implementation must perform linear interpolation between + adjacent points in the curve. Initially the curve attribute is null, which means that + the WaveShaperNode will pass its input to its output without modification.</p> + </dd> +</dl> + +<dl> + <dt id="dfn-oversample"><code>oversample</code></dt> + <dd><p>Specifies what type of oversampling (if any) should be used when applying the shaping curve. + The default value is "none", meaning the curve will be applied directly to the input samples. + A value of "2x" or "4x" can improve the quality of the processing by avoiding some aliasing, with + the "4x" value yielding the highest quality. For some applications, it's better to use no oversampling + in order to get a very precise shaping curve. + </p> + <p> + A value of "2x" or "4x" means that the following steps must be performed: + <ol> + <li>Up-sample the input samples to 2x or 4x the sample-rate of the AudioContext. Thus for each + processing block of 128 samples, generate 256 (for 2x) or 512 (for 4x) samples.</li> + <li>Apply the shaping curve.</li> + <li>Down-sample the result back to the sample-rate of the AudioContext. Thus taking the 256 (or 512) processed samples, generating 128 as + the final result. + </ol> + The exact up-sampling and down-sampling filters are not specified, and can be tuned for sound quality (low aliasing, etc.), low latency, and performance. + </p> + </dd> +</dl> +</div> +</div> + +<div id="OscillatorNode-section" class="section"> +<h2 id="OscillatorNode">4.23. The OscillatorNode Interface</h2> + +<p>OscillatorNode represents an audio source generating a periodic waveform. It can be set to +a few commonly used waveforms. Additionally, it can be set to an arbitrary periodic +waveform through the use of a <a href="#PeriodicWave-section"><code>PeriodicWave</code></a> object. </p> + +<p>Oscillators are common foundational building blocks in audio synthesis. An OscillatorNode will start emitting sound at the time +specified by the <code>start()</code> method. </p> + +<p> +Mathematically speaking, a <em>continuous-time</em> periodic waveform can have very high (or infinitely high) frequency information when considered +in the frequency domain. When this waveform is sampled as a discrete-time digital audio signal at a particular sample-rate, +then care must be taken to discard (filter out) the high-frequency information higher than the <em>Nyquist</em> frequency (half the sample-rate) +before converting the waveform to a digital form. If this is not done, then <em>aliasing</em> of higher frequencies (than the Nyquist frequency) will fold +back as mirror images into frequencies lower than the Nyquist frequency. In many cases this will cause audibly objectionable artifacts. +This is a basic and well understood principle of audio DSP. +</p> + +<p> +There are several practical approaches that an implementation may take to avoid this aliasing. +But regardless of approach, the <em>idealized</em> discrete-time digital audio signal is well defined mathematically. +The trade-off for the implementation is a matter of implementation cost (in terms of CPU usage) versus fidelity to +achieving this ideal. +</p> + +<p> +It is expected that an implementation will take some care in achieving this ideal, but it is reasonable to consider lower-quality, +less-costly approaches on lower-end hardware. +</p> + +<p> +Both .frequency and .detune are <em>a-rate</em> parameters and are used together to determine a <em>computedFrequency</em> value: +</p> + +<pre> +computedFrequency(t) = frequency(t) * pow(2, detune(t) / 1200) +</pre> + +<p> +The OscillatorNode's instantaneous phase at each time is the time integral of <em>computedFrequency</em>. +</p> + +<pre> numberOfInputs : 0 + numberOfOutputs : 1 (mono output) + </pre> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="oscillator-node-idl"> + +enum <dfn>OscillatorType</dfn> { + "sine", + "square", + "sawtooth", + "triangle", + "custom" +}; + +interface <dfn id="dfn-OscillatorNode">OscillatorNode</dfn> : AudioNode { + + attribute OscillatorType type; + + readonly attribute AudioParam frequency; // in Hertz + readonly attribute AudioParam detune; // in Cents + + void start(double when); + void stop(double when); + void setPeriodicWave(PeriodicWave periodicWave); + + attribute EventHandler onended; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-OscillatorNode-section" class="section"> +<h3 id="attributes-OscillatorNode">4.23.1. Attributes</h3> +<dl> + <dt id="dfn-type"><code>type</code></dt> + <dd><p>The shape of the periodic waveform. It may directly be set to any of the type constant values except for "custom". + The <a href="#dfn-setPeriodicWave"><code>setPeriodicWave()</code></a> method can be used to set a custom waveform, which results in this attribute + being set to "custom". The default value is "sine". </p> + </dd> +</dl> + +<dl> + <dt id="dfn-frequency"><code>frequency</code></dt> + <dd><p>The frequency (in Hertz) of the periodic waveform. This parameter is <em>a-rate</em> </p> + </dd> +</dl> +<dl> + <dt id="dfn-detune"><code>detune</code></dt> + <dd><p>A detuning value (in Cents) which will offset the <code>frequency</code> by the given amount. + This parameter is <em>a-rate</em> </p> + </dd> +</dl> +<dl> + <dt id="dfn-onended"><code>onended</code></dt> + <dd><p>A property used to set the <code>EventHandler</code> (described in <cite><a + href="http://www.whatwg.org/specs/web-apps/current-work/#eventhandler">HTML</a></cite>) + for the ended event that is dispatched to <a + href="#OscillatorNode-section"><code>OscillatorNode</code></a> + node types. When the playback of the buffer for an <code>OscillatorNode</code> + is finished, an event of type <code>Event</code> (described in <cite><a + href="http://www.whatwg.org/specs/web-apps/current-work/#event">HTML</a></cite>) + will be dispatched to the event handler. </p> + </dd> +</dl> +</div> +</div> + +<div id="methodsandparams-OscillatorNode-section" class="section"> +<h3 id="methodsandparams-OscillatorNode">4.23.2. Methods and Parameters</h3> +<dl> + <dt id="dfn-setPeriodicWave">The <code>setPeriodicWave</code> + method</dt> + <dd><p>Sets an arbitrary custom periodic waveform given a <a href="#PeriodicWave-section"><code>PeriodicWave</code></a>.</p> + </dd> +</dl> +<dl> + <dt id="dfn-start-AudioBufferSourceNode">The <code>start</code> + method</dt> + <dd><p>defined as in <a href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>. </p> + </dd> +</dl> +<dl> + <dt id="dfn-stop-AudioBufferSourceNode">The <code>stop</code> + method</dt> + <dd><p>defined as in <a href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>. </p> + </dd> +</dl> +</div> + + +<div id="PeriodicWave-section" class="section"> +<h2 id="PeriodicWave">4.24. The PeriodicWave Interface</h2> + +<p>PeriodicWave represents an arbitrary periodic waveform to be used with an <a href="#OscillatorNode-section"><code>OscillatorNode</code></a>. +Please see <a href="#dfn-createPeriodicWave">createPeriodicWave()</a> and <a href="#dfn-setPeriodicWave">setPeriodicWave()</a> and for more details. </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="wavetable-idl"> + +interface <dfn id="dfn-PeriodicWave">PeriodicWave</dfn> { + +}; +</code></pre> +</div> +</div> +</div> + +<div id="MediaStreamAudioSourceNode-section" class="section"> +<h2 id="MediaStreamAudioSourceNode">4.25. The MediaStreamAudioSourceNode +Interface</h2> + +<p>This interface represents an audio source from a <code>MediaStream</code>. +The first <code>AudioMediaStreamTrack</code> from the <code>MediaStream</code> will be +used as a source of audio.</p> +<pre> numberOfInputs : 0 + numberOfOutputs : 1 +</pre> + + <p> + The number of channels of the output corresponds to the number of channels of the <code>AudioMediaStreamTrack</code>. + If there is no valid audio track, then the number of channels output will be one silent channel. + </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="media-stream-audio-source-node-idl"> + +interface <dfn id="dfn-MediaStreamAudioSourceNode">MediaStreamAudioSourceNode</dfn> : AudioNode { + +}; +</code></pre> +</div> +</div> +</div> + +<div id="MediaStreamAudioDestinationNode-section" class="section"> +<h2 id="MediaStreamAudioDestinationNode">4.26. The MediaStreamAudioDestinationNode +Interface</h2> + +<p>This interface is an audio destination representing a <code>MediaStream</code> with a single <code>AudioMediaStreamTrack</code>. +This MediaStream is created when the node is created and is accessible via the <dfn>stream</dfn> attribute. +This stream can be used in a similar way as a MediaStream obtained via getUserMedia(), and +can, for example, be sent to a remote peer using the RTCPeerConnection addStream() method. +</p> +<pre> + numberOfInputs : 1 + numberOfOutputs : 0 + + channelCount = 2; + channelCountMode = "explicit"; + channelInterpretation = "speakers"; +</pre> + +<p> +The number of channels of the input is by default 2 (stereo). Any connections to the input +are up-mixed/down-mixed to the number of channels of the input. +</p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">Web IDL</span></div> + +<div class="blockContent"> +<pre class="code"><code class="idl-code" id="media-stream-audio-destination-node-idl"> + +interface <dfn id="dfn-MediaStreamAudioDestinationNode">MediaStreamAudioDestinationNode</dfn> : AudioNode { + + readonly attribute MediaStream stream; + +}; +</code></pre> +</div> +</div> + +<div id="attributes-MediaStreamAudioDestinationNode-section" class="section"> +<h3 id="attributes-MediaStreamAudioDestinationNode">4.26.1. Attributes</h3> +<dl> + <dt id="dfn-stream"><code>stream</code></dt> + <dd><p>A MediaStream containing a single AudioMediaStreamTrack with the same number of channels + as the node itself.</p> + </dd> +</dl> +</div> + +</div> + +<div id="MixerGainStructure-section" class="section"> +<h2 id="MixerGainStructure">6. Mixer Gain Structure</h2> + +<p class="norm">This section is informative.</p> + +<h3 id="background">Background</h3> + +<p>One of the most important considerations when dealing with audio processing +graphs is how to adjust the gain (volume) at various points. For example, in a +standard mixing board model, each input bus has pre-gain, post-gain, and +send-gains. Submix and master out busses also have gain control. The gain +control described here can be used to implement standard mixing boards as well +as other architectures. </p> + +<div id="SummingJunction-section" class="section"> +<h3 id="SummingJunction">Summing Inputs</h3> +</div> + +<p>The inputs to <a href="#AudioNode-section"><code>AudioNodes</code></a> have +the ability to accept connections from multiple outputs. The input then acts as +a unity gain summing junction with each output signal being added with the +others: </p> +<img alt="unity gain summing junction" +src="images/unity-gain-summing-junction.png" /> + +<p>In cases where the channel layouts of the outputs do not match, a mix (usually up-mix) will occur according to the <a +href="#UpMix-section">mixing rules</a>. +</p> + +<h3 id="gain-Control">Gain Control</h3> + +<p>But many times, it's important to be able to control the gain for each of +the output signals. The <a +href="#GainNode-section"><code>GainNode</code></a> gives this +control: </p> +<img alt="mixer architecture new" src="images/mixer-architecture-new.png" /> + +<p>Using these two concepts of unity gain summing junctions and GainNodes, +it's possible to construct simple or complex mixing scenarios. </p> + +<h3 id="Example-mixer-with-send-busses">Example: Mixer with Send Busses</h3> + +<p>In a routing scenario involving multiple sends and submixes, explicit +control is needed over the volume or "gain" of each connection to a mixer. Such +routing topologies are very common and exist in even the simplest of electronic +gear sitting around in a basic recording studio. </p> + +<p>Here's an example with two send mixers and a main mixer. Although possible, +for simplicity's sake, pre-gain control and insert effects are not illustrated: +</p> +<img alt="mixer gain structure" src="images/mixer-gain-structure.png" /> + +<p>This diagram is using a shorthand notation where "send 1", "send 2", and +"main bus" are actually inputs to AudioNodes, but here are represented as +summing busses, where the intersections g2_1, g3_1, etc. represent the "gain" +or volume for the given source on the given mixer. In order to expose this +gain, an <a href="#dfn-GainNode"><code>GainNode</code></a> is used: +</p> + +<p>Here's how the above diagram could be constructed in JavaScript: </p> + +<div class="example"> + +<div class="exampleHeader"> +Example</div> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">ECMAScript</span></div> + +<div class="blockContent"> +<pre class="code"><code class="es-code"> + +var context = 0; +var compressor = 0; +var reverb = 0; +var delay = 0; +var s1 = 0; +var s2 = 0; + +var source1 = 0; +var source2 = 0; +var g1_1 = 0; +var g2_1 = 0; +var g3_1 = 0; +var g1_2 = 0; +var g2_2 = 0; +var g3_2 = 0; + +<span class="comment">// Setup routing graph </span> +function setupRoutingGraph() { + context = new AudioContext(); + + compressor = context.createDynamicsCompressor(); + + <span class="comment">// Send1 effect </span> + reverb = context.createConvolver(); + <span class="comment">// Convolver impulse response may be set here or later </span> + + <span class="comment">// Send2 effect </span> + delay = context.createDelay(); + + <span class="comment">// Connect final compressor to final destination </span> + compressor.connect(context.destination); + + <span class="comment">// Connect sends 1 & 2 through effects to main mixer </span> + s1 = context.createGain(); + reverb.connect(s1); + s1.connect(compressor); + + s2 = context.createGain(); + delay.connect(s2); + s2.connect(compressor); + + <span class="comment">// Create a couple of sources </span> + source1 = context.createBufferSource(); + source2 = context.createBufferSource(); + source1.buffer = manTalkingBuffer; + source2.buffer = footstepsBuffer; + + <span class="comment">// Connect source1 </span> + g1_1 = context.createGain(); + g2_1 = context.createGain(); + g3_1 = context.createGain(); + source1.connect(g1_1); + source1.connect(g2_1); + source1.connect(g3_1); + g1_1.connect(compressor); + g2_1.connect(reverb); + g3_1.connect(delay); + + <span class="comment">// Connect source2 </span> + g1_2 = context.createGain(); + g2_2 = context.createGain(); + g3_2 = context.createGain(); + source2.connect(g1_2); + source2.connect(g2_2); + source2.connect(g3_2); + g1_2.connect(compressor); + g2_2.connect(reverb); + g3_2.connect(delay); + + <span class="comment">// We now have explicit control over all the volumes g1_1, g2_1, ..., s1, s2 </span> + g2_1.gain.value = 0.2; <span class="comment"> // For example, set source1 reverb gain </span> + + <span class="comment"> // Because g2_1.gain is an "AudioParam", </span> + <span class="comment"> // an automation curve could also be attached to it. </span> + <span class="comment"> // A "mixing board" UI could be created in canvas or WebGL controlling these gains. </span> +} + + </code></pre> +</div> +</div> +</div> +</div> +<br /> + + +<div id="DynamicLifetime-section"> +<h2 id="DynamicLifetime">7. Dynamic Lifetime</h2> + +<h3 id="DynamicLifetime-background">Background</h3> + +<p class="norm">This section is informative. Please see <a href="#lifetime-AudioContext">AudioContext lifetime</a> +and <a href="#lifetime-AudioNode">AudioNode lifetime</a> for normative requirements +</p> + +<p>In addition to allowing the creation of static routing configurations, it +should also be possible to do custom effect routing on dynamically allocated +voices which have a limited lifetime. For the purposes of this discussion, +let's call these short-lived voices "notes". Many audio applications +incorporate the ideas of notes, examples being drum machines, sequencers, and +3D games with many one-shot sounds being triggered according to game play. </p> + +<p>In a traditional software synthesizer, notes are dynamically allocated and +released from a pool of available resources. The note is allocated when a MIDI +note-on message is received. It is released when the note has finished playing +either due to it having reached the end of its sample-data (if non-looping), it +having reached a sustain phase of its envelope which is zero, or due to a MIDI +note-off message putting it into the release phase of its envelope. In the MIDI +note-off case, the note is not released immediately, but only when the release +envelope phase has finished. At any given time, there can be a large number of +notes playing but the set of notes is constantly changing as new notes are +added into the routing graph, and old ones are released. </p> + +<p>The audio system automatically deals with tearing-down the part of the +routing graph for individual "note" events. A "note" is represented by an +<code>AudioBufferSourceNode</code>, which can be directly connected to other +processing nodes. When the note has finished playing, the context will +automatically release the reference to the <code>AudioBufferSourceNode</code>, +which in turn will release references to any nodes it is connected to, and so +on. The nodes will automatically get disconnected from the graph and will be +deleted when they have no more references. Nodes in the graph which are +long-lived and shared between dynamic voices can be managed explicitly. +Although it sounds complicated, this all happens automatically with no extra +JavaScript handling required. </p> + +<h3 id="Example-DynamicLifetime">Example</h3> + +<div class="example"> + +<div class="exampleHeader"> +Example</div> +<img alt="dynamic allocation" src="images/dynamic-allocation.png" /> + +<p>The low-pass filter, panner, and second gain nodes are directly connected +from the one-shot sound. So when it has finished playing the context will +automatically release them (everything within the dotted line). If there are no +longer any JavaScript references to the one-shot sound and connected nodes, +then they will be immediately removed from the graph and deleted. The streaming +source, has a global reference and will remain connected until it is explicitly +disconnected. Here's how it might look in JavaScript: </p> + +<div class="block"> + +<div class="blockTitleDiv"> +<span class="blockTitle">ECMAScript</span></div> + +<div class="blockContent"> +<pre class="code"><code class="es-code"> + +var context = 0; +var compressor = 0; +var gainNode1 = 0; +var streamingAudioSource = 0; + +<span class="comment">// Initial setup of the "long-lived" part of the routing graph </span> +function setupAudioContext() { + context = new AudioContext(); + + compressor = context.createDynamicsCompressor(); + gainNode1 = context.createGain(); + + // Create a streaming audio source. + var audioElement = document.getElementById('audioTagID'); + streamingAudioSource = context.createMediaElementSource(audioElement); + streamingAudioSource.connect(gainNode1); + + gainNode1.connect(compressor); + compressor.connect(context.destination); +} + +<span class="comment">// Later in response to some user action (typically mouse or key event) </span> +<span class="comment">// a one-shot sound can be played. </span> +function playSound() { + var oneShotSound = context.createBufferSource(); + oneShotSound.buffer = dogBarkingBuffer; + + <span class="comment">// Create a filter, panner, and gain node. </span> + var lowpass = context.createBiquadFilter(); + var panner = context.createPanner(); + var gainNode2 = context.createGain(); + + <span class="comment">// Make connections </span> + oneShotSound.connect(lowpass); + lowpass.connect(panner); + panner.connect(gainNode2); + gainNode2.connect(compressor); + + <span class="comment">// Play 0.75 seconds from now (to play immediately pass in 0)</span> + oneShotSound.start(context.currentTime + 0.75); +} +</code></pre> +</div> +</div> +</div> +</div> + + + +<div id="UpMix-section" class="section"> +<h2 id="UpMix">9. Channel up-mixing and down-mixing</h2> + +<p class="norm">This section is normative.</p> + +<img src="images/unity-gain-summing-junction.png"> + +<p> +<a href="#MixerGainStructure-section">Mixer Gain Structure</a> +describes how an <dfn>input</dfn> to an AudioNode can be connected from one or more <dfn>outputs</dfn> +of an AudioNode. Each of these connections from an output represents a stream with +a specific non-zero number of channels. An input has <em>mixing rules</em> for combining the channels +from all of the connections to it. As a simple example, if an input is connected from a mono output and +a stereo output, then the mono connection will usually be up-mixed to stereo and summed with +the stereo connection. But, of course, it's important to define the exact <em>mixing rules</em> for +every input to every AudioNode. The default mixing rules for all of the inputs have been chosen so that +things "just work" without worrying too much about the details, especially in the very common +case of mono and stereo streams. But the rules can be changed for advanced use cases, especially +multi-channel. +</p> + +<p> +To define some terms, <em>up-mixing</em> refers to the process of taking a stream with a smaller +number of channels and converting it to a stream with a larger number of channels. <em>down-mixing</em> +refers to the process of taking a stream with a larger number of channels and converting it to a stream +with a smaller number of channels. +</p> + +<p> +An AudioNode input use three basic pieces of information to determine how to mix all the outputs +connected to it. As part of this process it computes an internal value <dfn>computedNumberOfChannels</dfn> + representing the actual number of channels of the input at any given time: +</p> + +<p> +The AudioNode attributes involved in channel up-mixing and down-mixing rules are defined +<a href="#attributes-AudioNode-section">above</a>. The following is a more precise specification +on what each of them mean. +</p> + +<ul> +<li><dfn>channelCount</dfn> is used to help compute <dfn>computedNumberOfChannels</dfn>.</li> + +<li><dfn>channelCountMode</dfn> determines how <dfn>computedNumberOfChannels</dfn> will be computed. +Once this number is computed, all of the connections will be up or down-mixed to that many channels. For most nodes, +the default value is "max". +<ul> +<li>“max”: <dfn>computedNumberOfChannels</dfn> is computed as the maximum of the number of channels of all connections. +In this mode <dfn>channelCount</dfn> is ignored.</li> +<li>“clamped-max”: same as “max” up to a limit of the <dfn>channelCount</dfn></li> +<li>“explicit”: <dfn>computedNumberOfChannels</dfn> is the exact value as specified in <dfn>channelCount</dfn></li> +</ul> + +</li> + +<li><dfn>channelInterpretation</dfn> determines how the individual channels will be treated. +For example, will they be treated as speakers having a specific layout, or will they +be treated as simple discrete channels? This value influences exactly how the up and down mixing is +performed. The default value is "speakers". + +<ul> +<li>“speakers”: use <a href="#ChannelLayouts">up-down-mix equations for mono/stereo/quad/5.1</a>. +In cases where the number of channels do not match any of these basic speaker layouts, revert +to "discrete". +</li> +<li>“discrete”: up-mix by filling channels until they run out then zero out remaining channels. + down-mix by filling as many channels as possible, then dropping remaining channels</li> +</ul> + +</li> + +</ul> + +<p> +For each input of an AudioNode, an implementation must: +</p> + +<ol> +<li>Compute <dfn>computedNumberOfChannels</dfn>.</li> +<li>For each connection to the input: +<ul> +<li> up-mix or down-mix the connection to <dfn>computedNumberOfChannels</dfn> according to <dfn>channelInterpretation</dfn>.</li> +<li> Mix it together with all of the other mixed streams (from other connections). This is a straight-forward mixing together of each of the corresponding channels from each +connection.</li> +</ul> +</li> +</ol> + + + + +<div id="ChannelLayouts-section" class="section"> +<h3 id="ChannelLayouts">9.1. Speaker Channel Layouts</h3> + +<p class="norm">This section is normative.</p> + +<p> +When <dfn>channelInterpretation</dfn> is "speakers" then the up-mixing and down-mixing +is defined for specific channel layouts. +</p> + +<p>It's important to define the channel ordering (and define some +abbreviations) for these speaker layouts.</p> + +<p> +For now, only considers cases for mono, stereo, quad, 5.1. Later other channel +layouts can be defined. +</p> + +<h4 id ="ChannelOrdering">9.1.1. Channel ordering</h4> + +<pre> Mono + 0: M: mono + + Stereo + 0: L: left + 1: R: right + </pre> + +<pre> Quad + 0: L: left + 1: R: right + 2: SL: surround left + 3: SR: surround right + + 5.1 + 0: L: left + 1: R: right + 2: C: center + 3: LFE: subwoofer + 4: SL: surround left + 5: SR: surround right + </pre> +</div> + +<h4 id="UpMix-sub">9.1.2. Up Mixing speaker layouts</h4> + +<pre>Mono up-mix: + + 1 -> 2 : up-mix from mono to stereo + output.L = input; + output.R = input; + + 1 -> 4 : up-mix from mono to quad + output.L = input; + output.R = input; + output.SL = 0; + output.SR = 0; + + 1 -> 5.1 : up-mix from mono to 5.1 + output.L = 0; + output.R = 0; + output.C = input; // put in center channel + output.LFE = 0; + output.SL = 0; + output.SR = 0; + +Stereo up-mix: + + 2 -> 4 : up-mix from stereo to quad + output.L = input.L; + output.R = input.R; + output.SL = 0; + output.SR = 0; + + 2 -> 5.1 : up-mix from stereo to 5.1 + output.L = input.L; + output.R = input.R; + output.C = 0; + output.LFE = 0; + output.SL = 0; + output.SR = 0; + +Quad up-mix: + + 4 -> 5.1 : up-mix from stereo to 5.1 + output.L = input.L; + output.R = input.R; + output.C = 0; + output.LFE = 0; + output.SL = input.SL; + output.SR = input.SR;</pre> + +<h4 id="down-mix">9.1.3. Down Mixing speaker layouts</h4> + +<p>A down-mix will be necessary, for example, if processing 5.1 source +material, but playing back stereo. </p> +<pre> +Mono down-mix: + + 2 -> 1 : stereo to mono + output = 0.5 * (input.L + input.R); + + 4 -> 1 : quad to mono + output = 0.25 * (input.L + input.R + input.SL + input.SR); + + 5.1 -> 1 : 5.1 to mono + output = 0.7071 * (input.L + input.R) + input.C + 0.5 * (input.SL + input.SR) + + +Stereo down-mix: + + 4 -> 2 : quad to stereo + output.L = 0.5 * (input.L + input.SL); + output.R = 0.5 * (input.R + input.SR); + + 5.1 -> 2 : 5.1 to stereo + output.L = L + 0.7071 * (input.C + input.SL) + output.R = R + 0.7071 * (input.C + input.SR) + +Quad down-mix: + + 5.1 -> 4 : 5.1 to quad + output.L = L + 0.7071 * input.C + output.R = R + 0.7071 * input.C + output.SL = input.SL + output.SR = input.SR + +</pre> +</div> + +<h3 id="ChannelRules-section">9.2. Channel Rules Examples</h3> + +<p class="norm">This section is informative.</p> + +<div class="block"> +<div class="blockTitleDiv"> +<div class="blockContent"> +<pre class="code"><code class="idl-code"> +// Set gain node to explicit 2-channels (stereo). +gain.channelCount = 2; +gain.channelCountMode = "explicit"; +gain.channelInterpretation = "speakers"; + +// Set "hardware output" to 4-channels for DJ-app with two stereo output busses. +context.destination.channelCount = 4; +context.destination.channelCountMode = "explicit"; +context.destination.channelInterpretation = "discrete"; + +// Set "hardware output" to 8-channels for custom multi-channel speaker array +// with custom matrix mixing. +context.destination.channelCount = 8; +context.destination.channelCountMode = "explicit"; +context.destination.channelInterpretation = "discrete"; + +// Set "hardware output" to 5.1 to play an HTMLAudioElement. +context.destination.channelCount = 6; +context.destination.channelCountMode = "explicit"; +context.destination.channelInterpretation = "speakers"; + +// Explicitly down-mix to mono. +gain.channelCount = 1; +gain.channelCountMode = "explicit"; +gain.channelInterpretation = "speakers"; +</code></pre> +</div> +</div> +</div> + + +<div id="Spatialization-section" class="section"> +<h2 id="Spatialization">11. Spatialization / Panning </h2> + +<h3 id="Spatialization-background">Background</h3> + +<p>A common feature requirement for modern 3D games is the ability to +dynamically spatialize and move multiple audio sources in 3D space. Game audio +engines such as OpenAL, FMOD, Creative's EAX, Microsoft's XACT Audio, etc. have +this ability. </p> + +<p>Using an <code>PannerNode</code>, an audio stream can be spatialized or +positioned in space relative to an <code>AudioListener</code>. An <a +href="#AudioContext-section"><code>AudioContext</code></a> will contain a +single <code>AudioListener</code>. Both panners and listeners have a position +in 3D space using a right-handed cartesian coordinate system. +The units used in the coordinate system are not defined, and do not need to be +because the effects calculated with these coordinates are independent/invariant +of any particular units such as meters or feet. <code>PannerNode</code> +objects (representing the source stream) have an <code>orientation</code> +vector representing in which direction the sound is projecting. Additionally, +they have a <code>sound cone</code> representing how directional the sound is. +For example, the sound could be omnidirectional, in which case it would be +heard anywhere regardless of its orientation, or it can be more directional and +heard only if it is facing the listener. <code>AudioListener</code> objects +(representing a person's ears) have an <code>orientation</code> and +<code>up</code> vector representing in which direction the person is facing. +Because both the source stream and the listener can be moving, they both have a +<code>velocity</code> vector representing both the speed and direction of +movement. Taken together, these two velocities can be used to generate a +doppler shift effect which changes the pitch. </p> + +<p> +During rendering, the <code>PannerNode</code> calculates an <em>azimuth</em> +and <em>elevation</em>. These values are used internally by the implementation in +order to render the spatialization effect. See the <a href="#Spatialization-panning-algorithm">Panning Algorithm</a> section +for details of how these values are used. +</p> + +<p> +The following algorithm must be used to calculate the <em>azimuth</em> +and <em>elevation</em>: +</p> + +<div class="block"> +<div class="blockTitleDiv"> +<div class="blockContent"> +<pre class="code"><code class="es-code"> +// Calculate the source-listener vector. +vec3 sourceListener = source.position - listener.position; + +if (sourceListener.isZero()) { + // Handle degenerate case if source and listener are at the same point. + azimuth = 0; + elevation = 0; + return; +} + +sourceListener.normalize(); + +// Align axes. +vec3 listenerFront = listener.orientation; +vec3 listenerUp = listener.up; +vec3 listenerRight = listenerFront.cross(listenerUp); +listenerRight.normalize(); + +vec3 listenerFrontNorm = listenerFront; +listenerFrontNorm.normalize(); + +vec3 up = listenerRight.cross(listenerFrontNorm); + +float upProjection = sourceListener.dot(up); + +vec3 projectedSource = sourceListener - upProjection * up; +projectedSource.normalize(); + +azimuth = 180 * acos(projectedSource.dot(listenerRight)) / PI; + +// Source in front or behind the listener. +double frontBack = projectedSource.dot(listenerFrontNorm); +if (frontBack < 0) + azimuth = 360 - azimuth; + +// Make azimuth relative to "front" and not "right" listener vector. +if ((azimuth >= 0) && (azimuth <= 270)) + azimuth = 90 - azimuth; +else + azimuth = 450 - azimuth; + +elevation = 90 - 180 * acos(sourceListener.dot(up)) / PI; + +if (elevation > 90) + elevation = 180 - elevation; +else if (elevation < -90) + elevation = -180 - elevation; +</code></pre> +</div> +</div> +</div> + +<h3 id="Spatialization-panning-algorithm">Panning Algorithm</h3> + +<p> +<em>mono->stereo</em> and <em>stereo->stereo</em> panning must be supported. +<em>mono->stereo</em> processing is used when all connections to the input are mono. +Otherwise <em>stereo->stereo</em> processing is used.</p> + +<p>The following algorithms must be implemented: </p> +<ul> + <li>Equal-power (Vector-based) panning + <p>This is a simple and relatively inexpensive algorithm which provides + basic, but reasonable results. It is commonly used when panning musical sources. + </p> + The <em>elevation</em> value is ignored in this panning algorithm. + + <p> + The following steps are used for processing: + </p> + + <ol> + + <li> + <p> + The <em>azimuth</em> value is first contained to be within the range -90 <= <em>azimuth</em> <= +90 according to: + </p> + <pre> + // Clamp azimuth to allowed range of -180 -> +180. + azimuth = max(-180, azimuth); + azimuth = min(180, azimuth); + + // Now wrap to range -90 -> +90. + if (azimuth < -90) + azimuth = -180 - azimuth; + else if (azimuth > 90) + azimuth = 180 - azimuth; + </pre> + </li> + + <li> + <p> + A 0 -> 1 normalized value <em>x</em> is calculated from <em>azimuth</em> for <em>mono->stereo</em> as: + </p> + <pre> + x = (azimuth + 90) / 180 + </pre> + + <p> + Or for <em>stereo->stereo</em> as: + </p> + <pre> + if (azimuth <= 0) { // from -90 -> 0 + // inputL -> outputL and "equal-power pan" inputR as in mono case + // by transforming the "azimuth" value from -90 -> 0 degrees into the range -90 -> +90. + x = (azimuth + 90) / 90; + } else { // from 0 -> +90 + // inputR -> outputR and "equal-power pan" inputL as in mono case + // by transforming the "azimuth" value from 0 -> +90 degrees into the range -90 -> +90. + x = azimuth / 90; + } + </pre> + </li> + + <li> + <p> + Left and right gain values are then calculated: + </p> + <pre> + gainL = cos(0.5 * PI * x); + gainR = sin(0.5 * PI * x); + </pre> + </li> + + <li> + <p>For <em>mono->stereo</em>, the output is calculated as:</p> + <pre> + outputL = input * gainL + outputR = input * gainR + </pre> + <p>Else for <em>stereo->stereo</em>, the output is calculated as:</p> + <pre> + if (azimuth <= 0) { // from -90 -> 0 + outputL = inputL + inputR * gainL; + outputR = inputR * gainR; + } else { // from 0 -> +90 + outputL = inputL * gainL; + outputR = inputR + inputL * gainR; + } + </pre> + </li> + + </ol> + + + + </li> + <li><a + href="http://en.wikipedia.org/wiki/Head-related_transfer_function">HRTF</a> + panning (stereo only) + <p>This requires a set of HRTF impulse responses recorded at a variety of + azimuths and elevations. There are a small number of open/free impulse + responses available. The implementation requires a highly optimized + convolution function. It is somewhat more costly than "equal-power", but + provides a more spatialized sound. </p> + <img alt="HRTF panner" src="images/HRTF_panner.png" /></li> +</ul> + +<h3 id="Spatialization-distance-effects">Distance Effects</h3> +<p> +Sounds which are closer are louder, while sounds further away are quieter. +Exactly <em>how</em> a sound's volume changes according to distance from the listener +depends on the <em>distanceModel</em> attribute. +</p> + + +<p> +During audio rendering, a <em>distance</em> value will be calculated based on the panner and listener positions according to: +</p> +<pre> +v = panner.position - listener.position +</pre> +<pre> +distance = sqrt(dot(v, v)) +</pre> + +<p> +<em>distance</em> will then be used to calculate <em>distanceGain</em> which depends +on the <em>distanceModel</em> attribute. See the <a href="#dfn-distanceModel">distanceModel</a> section for details of +how this is calculated for each distance model. +</p> +<p>As part of its processing, the <code>PannerNode</code> scales/multiplies the input audio signal by <em>distanceGain</em> +to make distant sounds quieter and nearer ones louder. +</p> + + + + +<h3 id="Spatialization-sound-cones">Sound Cones</h3> + +<p>The listener and each sound source have an orientation vector describing +which way they are facing. Each sound source's sound projection characteristics +are described by an inner and outer "cone" describing the sound intensity as a +function of the source/listener angle from the source's orientation vector. +Thus, a sound source pointing directly at the listener will be louder than if +it is pointed off-axis. Sound sources can also be omni-directional. </p> + +<p> +The following algorithm must be used to calculate the gain contribution due +to the cone effect, given the source (the <code>PannerNode</code>) and the listener: +</p> + +<div class="block"> +<div class="blockTitleDiv"> +<div class="blockContent"> +<pre class="code"><code class="idl-code"> +if (source.orientation.isZero() || ((source.coneInnerAngle == 360) && (source.coneOuterAngle == 360))) + return 1; // no cone specified - unity gain + +// Normalized source-listener vector +vec3 sourceToListener = listener.position - source.position; +sourceToListener.normalize(); + +vec3 normalizedSourceOrientation = source.orientation; +normalizedSourceOrientation.normalize(); + +// Angle between the source orientation vector and the source-listener vector +double dotProduct = sourceToListener.dot(normalizedSourceOrientation); +double angle = 180 * acos(dotProduct) / PI; +double absAngle = fabs(angle); + +// Divide by 2 here since API is entire angle (not half-angle) +double absInnerAngle = fabs(source.coneInnerAngle) / 2; +double absOuterAngle = fabs(source.coneOuterAngle) / 2; +double gain = 1; + +if (absAngle <= absInnerAngle) + // No attenuation + gain = 1; +else if (absAngle >= absOuterAngle) + // Max attenuation + gain = source.coneOuterGain; +else { + // Between inner and outer cones + // inner -> outer, x goes from 0 -> 1 + double x = (absAngle - absInnerAngle) / (absOuterAngle - absInnerAngle); + gain = (1 - x) + source.coneOuterGain * x; +} + +return gain; +</code></pre> +</div> +</div> +</div> + +<h3 id="Spatialization-doppler-shift">Doppler Shift</h3> +<ul> + <li>Introduces a pitch shift which can realistically simulate moving + sources.</li> + <li>Depends on: source / listener velocity vectors, speed of sound, doppler + factor.</li> +</ul> + +<p> +The following algorithm must be used to calculate the doppler shift value which is used +as an additional playback rate scalar for all AudioBufferSourceNodes connecting directly or +indirectly to the AudioPannerNode: +</p> + +<div class="block"> +<div class="blockTitleDiv"> +<div class="blockContent"> +<pre class="code"><code class="idl-code"> +double dopplerShift = 1; // Initialize to default value +double dopplerFactor = listener.dopplerFactor; + +if (dopplerFactor > 0) { + double speedOfSound = listener.speedOfSound; + + // Don't bother if both source and listener have no velocity. + if (!source.velocity.isZero() || !listener.velocity.isZero()) { + // Calculate the source to listener vector. + vec3 sourceToListener = source.position - listener.position; + + double sourceListenerMagnitude = sourceToListener.length(); + + double listenerProjection = sourceToListener.dot(listener.velocity) / sourceListenerMagnitude; + double sourceProjection = sourceToListener.dot(source.velocity) / sourceListenerMagnitude; + + listenerProjection = -listenerProjection; + sourceProjection = -sourceProjection; + + double scaledSpeedOfSound = speedOfSound / dopplerFactor; + listenerProjection = min(listenerProjection, scaledSpeedOfSound); + sourceProjection = min(sourceProjection, scaledSpeedOfSound); + + dopplerShift = ((speedOfSound - dopplerFactor * listenerProjection) / (speedOfSound - dopplerFactor * sourceProjection)); + fixNANs(dopplerShift); // Avoid illegal values + + // Limit the pitch shifting to 4 octaves up and 3 octaves down. + dopplerShift = min(dopplerShift, 16); + dopplerShift = max(dopplerShift, 0.125); + } +} +</code></pre> +</div> +</div> +</div> + + + + +</div> + +<div id="Convolution-section" class="section"> +<h2 id="Convolution">12. Linear Effects using Convolution</h2> + +<h3 id="Convolution-background">Background</h3> + +<p><a href="http://en.wikipedia.org/wiki/Convolution">Convolution</a> is a +mathematical process which can be applied to an audio signal to achieve many +interesting high-quality linear effects. Very often, the effect is used to +simulate an acoustic space such as a concert hall, cathedral, or outdoor +amphitheater. It can also be used for complex filter effects, like a muffled +sound coming from inside a closet, sound underwater, sound coming through a +telephone, or playing through a vintage speaker cabinet. This technique is very +commonly used in major motion picture and music production and is considered to +be extremely versatile and of high quality. </p> + +<p>Each unique effect is defined by an <code>impulse response</code>. An +impulse response can be represented as an audio file and <a +href="#recording-impulse-responses">can be recorded</a> from a real acoustic +space such as a cave, or can be synthetically generated through a great variety +of techniques. </p> + +<h3 id="Convolution-motivation">Motivation for use as a Standard</h3> + +<p>A key feature of many game audio engines (OpenAL, FMOD, Creative's EAX, +Microsoft's XACT Audio, etc.) is a reverberation effect for simulating the +sound of being in an acoustic space. But the code used to generate the effect +has generally been custom and algorithmic (generally using a hand-tweaked set +of delay lines and allpass filters which feedback into each other). In nearly +all cases, not only is the implementation custom, but the code is proprietary +and closed-source, each company adding its own "black magic" to achieve its +unique quality. Each implementation being custom with a different set of +parameters makes it impossible to achieve a uniform desired effect. And the +code being proprietary makes it impossible to adopt a single one of the +implementations as a standard. Additionally, algorithmic reverberation effects +are limited to a relatively narrow range of different effects, regardless of +how the parameters are tweaked. </p> + +<p>A convolution effect solves these problems by using a very precisely defined +mathematical algorithm as the basis of its processing. An impulse response +represents an exact sound effect to be applied to an audio stream and is easily +represented by an audio file which can be referenced by URL. The range of +possible effects is enormous. </p> + +<h3 id="Convolution-implementation-guide">Implementation Guide</h3> +<p> +Linear convolution can be implemented efficiently. +Here are some <a href="https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/convolution.html">notes</a> +describing how it can be practically implemented. +</p> + +<h3 id="Convolution-reverb-effect">Reverb Effect (with matrixing)</h3> + +<p class="norm">This section is normative.</p> + +<p> +In the general case the source +has N input channels, the impulse response has K channels, and the playback +system has M output channels. Thus it's a matter of how to matrix these +channels to achieve the final result. +</p> + +<p> +The subset of N, M, K below must be implemented (note that the first image in the diagram is just illustrating +the general case and is not normative, while the following images are normative). +Without loss of generality, developers desiring more complex and arbitrary matrixing can use multiple <code>ConvolverNode</code> +objects in conjunction with an <code>ChannelMergerNode</code>. +</p> + + +<p>Single channel convolution operates on a mono audio input, using a mono +impulse response, and generating a mono output. But to achieve a more spacious sound, 2 channel audio +inputs and 1, 2, or 4 channel impulse responses will be considered. The following diagram, illustrates the +common cases for stereo playback where N and M are 1 or 2 and K is 1, 2, or 4. +</p> +<img alt="reverb matrixing" src="images/reverb-matrixing.png" /> + +<h3 id="recording-impulse-responses">Recording Impulse Responses</h3> + +<p class="norm">This section is informative.</p> +<img alt="impulse response" src="images/impulse-response.png" /> <br /> +<br /> + + +<p>The most <a +href="http://pcfarina.eng.unipr.it/Public/Papers/226-AES122.pdf">modern</a> and +accurate way to record the impulse response of a real acoustic space is to use +a long exponential sine sweep. The test-tone can be as long as 20 or 30 +seconds, or longer. <br /> +Several recordings of the test tone played through a speaker can be made with +microphones placed and oriented at various positions in the room. It's +important to document speaker placement/orientation, the types of microphones, +their settings, placement, and orientations for each recording taken. </p> + +<p>Post-processing is required for each of these recordings by performing an +inverse-convolution with the test tone, yielding the impulse response of the +room with the corresponding microphone placement. These impulse responses are +then ready to be loaded into the convolution reverb engine to re-create the +sound of being in the room. </p> + +<h3 id="tools">Tools</h3> + +<p>Two command-line tools have been written: <br /> +<code>generate_testtones</code> generates an exponential sine-sweep test-tone +and its inverse. Another tool <code>convolve</code> was written for +post-processing. With these tools, anybody with recording equipment can record +their own impulse responses. To test the tools in practice, several recordings +were made in a warehouse space with interesting acoustics. These were later +post-processed with the command-line tools. </p> +<pre>% generate_testtones -h +Usage: generate_testtone + [-o /Path/To/File/To/Create] Two files will be created: .tone and .inverse + [-rate <sample rate>] sample rate of the generated test tones + [-duration <duration>] The duration, in seconds, of the generated files + [-min_freq <min_freq>] The minimum frequency, in hertz, for the sine sweep + +% convolve -h +Usage: convolve input_file impulse_response_file output_file</pre> +<br /> + + +<h3 id="recording-setup">Recording Setup</h3> +<img alt="recording setup" src="images/recording-setup.png" /> <br /> +<br /> +Audio Interface: Metric Halo Mobile I/O 2882 <br /> +<br /> +<br /> +<br /> +<img alt="microphones speaker" src="images/microphones-speaker.png" /> <br /> +<br /> +<img alt="microphone" src="images/microphone.png" /> <img alt="speaker" +src="images/speaker.png" /> <br /> +<br /> +Microphones: AKG 414s, Speaker: Mackie HR824 <br /> +<br /> +<br /> + + +<h3 id="warehouse">The Warehouse Space</h3> +<img alt="warehouse" src="images/warehouse.png" /> <br /> +<br /> +</div> + +<div id="JavaScriptProcessing-section" class="section"> +<h2 id="JavaScriptProcessing">13. JavaScript Synthesis and Processing</h2> + +<p class="norm">This section is informative.</p> + +<p>The Mozilla project has conducted <a +href="https://wiki.mozilla.org/Audio_Data_API">Experiments</a> to synthesize +and process audio directly in JavaScript. This approach is interesting for a +certain class of audio processing and they have produced a number of impressive +demos. This specification includes a means of synthesizing and processing +directly using JavaScript by using a special subtype of <a +href="#AudioNode-section"><code>AudioNode</code></a> called <a +href="#ScriptProcessorNode-section"><code>ScriptProcessorNode</code></a>. </p> + +<p>Here are some interesting examples where direct JavaScript processing can be +useful: </p> + +<h3 id="custom-DSP-effects">Custom DSP Effects</h3> + +<p>Unusual and interesting custom audio processing can be done directly in JS. +It's also a good test-bed for prototyping new algorithms. This is an extremely +rich area. </p> + +<h3 id="educational-applications">Educational Applications</h3> + +<p>JS processing is ideal for illustrating concepts in computer music synthesis +and processing, such as showing the de-composition of a square wave into its +harmonic components, FM synthesis techniques, etc. </p> + +<h3 id="javaScript-performance">JavaScript Performance</h3> + +<p>JavaScript has a variety of <a +href="#JavaScriptPerformance-section">performance issues</a> so it is not +suitable for all types of audio processing. The approach proposed in this +document includes the ability to perform computationally intensive aspects of +the audio processing (too expensive for JavaScript to compute in real-time) +such as multi-source 3D spatialization and convolution in optimized C++ code. +Both direct JavaScript processing and C++ optimized code can be combined due to +the APIs <a href="#ModularRouting-section">modular approach</a>. </p> + +<div id="Performance-section" class="section"> +<h2 id="Performance">15. Performance Considerations</h2> + +<div id="Latency-section" class="section"> +<h3 id="Latency">15.1. Latency: What it is and Why it's Important</h3> +</div> +<img alt="latency" src="images/latency.png" /> + +<p>For web applications, the time delay between mouse and keyboard events +(keydown, mousedown, etc.) and a sound being heard is important. </p> + +<p>This time delay is called latency and is caused by several factors (input +device latency, internal buffering latency, DSP processing latency, output +device latency, distance of user's ears from speakers, etc.), and is +cummulative. The larger this latency is, the less satisfying the user's +experience is going to be. In the extreme, it can make musical production or +game-play impossible. At moderate levels it can affect timing and give the +impression of sounds lagging behind or the game being non-responsive. For +musical applications the timing problems affect rhythm. For gaming, the timing +problems affect precision of gameplay. For interactive applications, it +generally cheapens the users experience much in the same way that very low +animation frame-rates do. Depending on the application, a reasonable latency +can be from as low as 3-6 milliseconds to 25-50 milliseconds. </p> + +<div id="Glitching-section" class="section"> +<h3 id="audio-glitching">15.2. Audio Glitching</h3> +</div> + +<p>Audio glitches are caused by an interruption of the normal continuous audio +stream, resulting in loud clicks and pops. It is considered to be a +catastrophic failure of a multi-media system and must be avoided. It can be +caused by problems with the threads responsible for delivering the audio stream +to the hardware, such as scheduling latencies caused by threads not having the +proper priority and time-constraints. It can also be caused by the audio DSP +trying to do more work than is possible in real-time given the CPU's speed. </p> + +<h3 id="hardware-scalability">15.3. Hardware Scalability</h3> + +<p>The system should gracefully degrade to allow audio processing under +resource constrained conditions without dropping audio frames. </p> + +<p>First of all, it should be clear that regardless of the platform, the audio +processing load should never be enough to completely lock up the machine. +Second, the audio rendering needs to produce a clean, un-interrupted audio +stream without audible <a href="#Glitching-section">glitches</a>. </p> + +<p>The system should be able to run on a range of hardware, from mobile phones +and tablet devices to laptop and desktop computers. But the more limited +compute resources on a phone device make it necessary to consider techniques to +scale back and reduce the complexity of the audio rendering. For example, +voice-dropping algorithms can be implemented to reduce the total number of +notes playing at any given time. </p> + +<p>Here's a list of some techniques which can be used to limit CPU usage: </p> + +<h4 id="CPU-monitoring">15.3.1. CPU monitoring</h4> + +<p>In order to avoid audio breakup, CPU usage must remain below 100%. </p> + +<p>The relative CPU usage can be dynamically measured for each AudioNode (and +chains of connected nodes) as a percentage of the rendering time quantum. In a +single-threaded implementation, overall CPU usage must remain below 100%. The +measured usage may be used internally in the implementation for dynamic +adjustments to the rendering. It may also be exposed through a +<code>cpuUsage</code> attribute of <code>AudioNode</code> for use by +JavaScript. </p> + +<p>In cases where the measured CPU usage is near 100% (or whatever threshold is +considered too high), then an attempt to add additional <code>AudioNodes</code> +into the rendering graph can trigger voice-dropping. </p> + +<h4 id="Voice-dropping">15.3.2. Voice Dropping</h4> + +<p>Voice-dropping is a technique which limits the number of voices (notes) +playing at the same time to keep CPU usage within a reasonable range. There can +either be an upper threshold on the total number of voices allowed at any given +time, or CPU usage can be dynamically monitored and voices dropped when CPU +usage exceeds a threshold. Or a combination of these two techniques can be +applied. When CPU usage is monitored for each voice, it can be measured all the +way from a source node through any effect processing nodes which apply +uniquely to that voice. </p> + +<p>When a voice is "dropped", it needs to happen in such a way that it doesn't +introduce audible clicks or pops into the rendered audio stream. One way to +achieve this is to quickly fade-out the rendered audio for that voice before +completely removing it from the rendering graph. </p> + +<p>When it is determined that one or more voices must be dropped, there are +various strategies for picking which voice(s) to drop out of the total ensemble +of voices currently playing. Here are some of the factors which can be used in +combination to help with this decision: </p> +<ul> + <li>Older voices, which have been playing the longest can be dropped instead + of more recent voices. </li> + <li>Quieter voices, which are contributing less to the overall mix may be + dropped instead of louder ones. </li> + <li>Voices which are consuming relatively more CPU resources may be dropped + instead of less "expensive" voices.</li> + <li>An AudioNode can have a <code>priority</code> attribute to help determine + the relative importance of the voices.</li> +</ul> + +<h4 id="Simplification-of-Effects-Processing">15.3.3. Simplification of Effects +Processing</h4> + +<p>Most of the effects described in this document are relatively inexpensive +and will likely be able to run even on the slower mobile devices. However, the +<a href="#ConvolverNode-section">convolution effect</a> can be configured with +a variety of impulse responses, some of which will likely be too heavy for +mobile devices. Generally speaking, CPU usage scales with the length of the +impulse response and the number of channels it has. Thus, it is reasonable to +consider that impulse responses which exceed a certain length will not be +allowed to run. The exact limit can be determined based on the speed of the +device. Instead of outright rejecting convolution with these long responses, it +may be interesting to consider truncating the impulse responses to the maximum +allowed length and/or reducing the number of channels of the impulse response. +</p> + +<p>In addition to the convolution effect. The <a +href="#PannerNode-section"><code>PannerNode</code></a> may also be +expensive if using the HRTF panning model. For slower devices, a cheaper +algorithm such as EQUALPOWER can be used to conserve compute resources. </p> + +<h4 id="Sample-rate">15.3.4. Sample Rate</h4> + +<p>For very slow devices, it may be worth considering running the rendering at +a lower sample-rate than normal. For example, the sample-rate can be reduced +from 44.1KHz to 22.05KHz. This decision must be made when the +<code>AudioContext</code> is created, because changing the sample-rate +on-the-fly can be difficult to implement and will result in audible glitching +when the transition is made. </p> + +<h4 id="pre-flighting">15.3.5. Pre-flighting</h4> + +<p>It should be possible to invoke some kind of "pre-flighting" code (through +JavaScript) to roughly determine the power of the machine. The JavaScript code +can then use this information to scale back any more intensive processing it +may normally run on a more powerful machine. Also, the underlying +implementation may be able to factor in this information in the voice-dropping +algorithm. </p> + +<p><span class="ednote">TODO: add specification and more detail here </span></p> + +<h4 id="Authoring-for-different-user-agents">15.3.6. Authoring for different +user agents</h4> +JavaScript code can use information about user-agent to scale back any more +intensive processing it may normally run on a more powerful machine. + +<h4 id="Scalability-of-Direct-JavaScript-Synthesis">15.3.7. Scalability of +Direct JavaScript Synthesis / Processing</h4> + +<p>Any audio DSP / processing code done directly in JavaScript should also be +concerned about scalability. To the extent possible, the JavaScript code itself +needs to monitor CPU usage and scale back any more ambitious processing when +run on less powerful devices. If it's an "all or nothing" type of processing, +then user-agent check or pre-flighting should be done to avoid generating an +audio stream with audio breakup. </p> + +<div id="JavaScriptPerformance-section" class="section"> +<h3 id="JavaScriptPerformance">15.4. JavaScript Issues with real-time +Processing and Synthesis: </h3> +</div> +While processing audio in JavaScript, it is extremely challenging to get +reliable, glitch-free audio while achieving a reasonably low-latency, +especially under heavy processor load. +<ul> + <li>JavaScript is very much slower than heavily optimized C++ code and is not + able to take advantage of SSE optimizations and multi-threading which is + critical for getting good performance on today's processors. Optimized + native code can be on the order of twenty times faster for processing FFTs + as compared with JavaScript. It is not efficient enough for heavy-duty + processing of audio such as convolution and 3D spatialization of large + numbers of audio sources. </li> + <li>setInterval() and XHR handling will steal time from the audio processing. + In a reasonably complex game, some JavaScript resources will be needed for + game physics and graphics. This creates challenges because audio rendering + is deadline driven (to avoid glitches and get low enough latency).</li> + <li>JavaScript does not run in a real-time processing thread and thus can be + pre-empted by many other threads running on the system.</li> + <li>Garbage Collection (and autorelease pools on Mac OS X) can cause + unpredictable delay on a JavaScript thread. </li> + <li>Multiple JavaScript contexts can be running on the main thread, stealing + time from the context doing the processing. </li> + <li>Other code (other than JavaScript) such as page rendering runs on the + main thread. </li> + <li>Locks can be taken and memory is allocated on the JavaScript thread. This + can cause additional thread preemption. </li> +</ul> +The problems are even more difficult with today's generation of mobile devices +which have processors with relatively poor performance and power consumption / +battery-life issues. <br /> +<br /> + + +<div id="ExampleApplications-section" class="section"> +<h2 id="ExampleApplications">16. Example Applications</h2> + +<p class="norm">This section is informative.</p> + +<p>Please see the <a +href="http://chromium.googlecode.com/svn/trunk/samples/audio/index.html">demo</a> +page for working examples. </p> + +<p>Here are some of the types of applications a web audio system should be able +to support: </p> + +<h3 id="basic-sound-playback">Basic Sound Playback</h3> + +<p>Simple and <a href="#Latency-section"><strong>low-latency</strong></a> +playback of sound effects in response to simple user actions such as mouse +click, roll-over, key press. </p> +<br /> + + +<h3 id="threeD-environmentse-and-games">3D Environments and Games</h3> +<img alt="quake" src="http://payload48.cargocollective.com/1/2/66805/3278334/redteam_680.jpg" /> +<br /> +<br /> + + +<p>Electronic Arts has produced an impressive immersive game called + <a href="http://sophie-lu.com/Strike-Fortress-EA">Strike Fortress</a>, +taking advantage of 3D spatialization and convolution for room simulation.</p> + +<img alt="beach demo" src="images/beach-demo.png" /> + +<p>3D environments with audio are common in games made for desktop applications +and game consoles. Imagine a 3D island environment with spatialized audio, +seagulls flying overhead, the waves crashing against the shore, the crackling +of the fire, the creaking of the bridge, and the rustling of the trees in the +wind. The sounds can be positioned naturally as one moves through the scene. +Even going underwater, low-pass filters can be tweaked for just the right +underwater sound. </p> +<br /> +<br /> +<img alt="box2d" src="images/box2d.png" /> <img alt="8-ball" +src="images/8-ball.png" /> <br /> +<br /> + + +<p><a href="http://box2d.org/">Box2D</a> is an interesting open-source +library for 2D game physics. It has various implementations, including one +based on Canvas 2D. A demo has been created with dynamic sound effects for each +of the object collisions, taking into account the velocities vectors and +positions to spatialize the sound events, and modulate audio effect parameters +such as filter cutoff. </p> + +<p>A virtual pool game with multi-sampled sound effects has also been created. +</p> +<br /> + + +<h3 id="musical-applications">Musical Applications</h3> +<img alt="garageband" src="images/garage-band.png" /> <img +alt="shiny drum machine" src="images/shiny-drum-machine.png" /> <img +alt="tonecraft" src="images/tonecraft.png" /> <br /> +<br /> +Many music composition and production applications are possible. Applications +requiring tight scheduling of audio events can be implemented and can be both +educational and entertaining. Drum machines, digital DJ applications, and even +timeline-based digital music production software with some of the features of +<a href="http://en.wikipedia.org/wiki/GarageBand">GarageBand</a> can be +written. <br /> +<br /> + + +<h3 id="music-visualizers">Music Visualizers</h3> +<img alt="music visualizer" src="images/music-visualizer.png" /> <br /> +<br /> +When combined with WebGL GLSL shaders, realtime analysis data can be presented +in entertaining ways. These can be as advanced as any found in iTunes. <br /> +<br /> + + +<h3 id="educational-applications_2">Educational Applications</h3> +<img alt="javascript processing" src="images/javascript-processing.png" /> + +<p>A variety of educational applications can be written, illustrating concepts +in music theory and computer music synthesis and processing. </p> +<br /> + + +<h3 id="artistic-audio-exploration">Artistic Audio Exploration</h3> + +<p>There are many creative possibilites for artistic sonic environments for +installation pieces. </p> +<br /> +</div> + +<div id="SecurityConsiderations-section" class="section"> +<h2 id="SecurityConsiderations">17. Security Considerations</h2> + +<p>This section is <em>informative.</em> </p> +</div> + +<div id="PrivacyConsiderations-section" class="section"> +<h2 id="PrivacyConsiderations">18. Privacy Considerations</h2> + +<p>This section is <em>informative</em>. When giving various information on +available AudioNodes, the Web Audio API potentially exposes information on +characteristic features of the client (such as audio hardware sample-rate) to +any page that makes use of the AudioNode interface. Additionally, timing +information can be collected through the RealtimeAnalyzerNode or +ScriptProcessorNode interface. The information could subsequently be used to +create a fingerprint of the client. </p> + +<p>Currently audio input is not specified in this document, but it will involve +gaining access to the client machine's audio input or microphone. This will +require asking the user for permission in an appropriate way, probably via the +<a href="http://developers.whatwg.org/">getUserMedia() +API</a>. </p> +</div> + +<div id="requirements-section" class="section"> +<h2 id="requirements">19. Requirements and Use Cases</h2> + +<p>Please see <a href="#ExampleApplications-section">Example Applications</a> +</p> +</div> + +<div id="oldnames-section" class="section"> +<h2 id="OldNames">20. Old Names</h2> + +<p class="norm">This section is informative.</p> + +<p>Some method and attribute names have been improved during API review. +The new names are described in the main body of this specification in the +description for each node type, etc. Here's a description of the older names +to help content authors migrate to the latest spec. Note that the partial +interfaces are not normative and are only descriptive: +</p> +<blockquote> +<pre> + +partial interface <dfn>AudioBufferSourceNode</dfn> { + // Same as start() + void noteOn(double when); + void noteGrainOn(double when, double grainOffset, double grainDuration); + + // Same as stop() + void noteOff(double when); +}; + +partial interface <dfn>AudioContext</dfn> { + // Same as createGain() + GainNode createGainNode(); + + // Same as createDelay() + DelayNode createDelayNode(optional double maxDelayTime = 1.0); + + // Same as createScriptProcessor() + ScriptProcessorNode createJavaScriptNode(optional unsigned long bufferSize = 0, + optional unsigned long numberOfInputChannels = 2, + optional unsigned long numberOfOutputChannels = 2); +}; + +partial interface <dfn>OscillatorNode</dfn> { + // Same as start() + void noteOn(double when); + + // Same as stop() + void noteOff(double when); +}; + +partial interface <dfn>AudioParam</dfn> { + // Same as setTargetAtTime() + void setTargetValueAtTime(float target, double startTime, double timeConstant); +}; + +</pre> +</blockquote> + +<p>Some attributes taking constant values have changed during API review. +The old way used integer values, while the new way uses Web IDL string values. +</p> + +<blockquote> +<pre> +// PannerNode constants for the .panningModel attribute + +// Old way +const unsigned short EQUALPOWER = 0; +const unsigned short HRTF = 1; + +// New way +enum <dfn>PanningModelType</dfn> { + "equalpower", + "HRTF" +}; +</pre> +</blockquote> + +<blockquote> +<pre> +// PannerNode constants for the .distanceModel attribute + +// Old way +const unsigned short LINEAR_DISTANCE = 0; +const unsigned short INVERSE_DISTANCE = 1; +const unsigned short EXPONENTIAL_DISTANCE = 2; + +// New way +enum <dfn>DistanceModelType</dfn> { + "linear", + "inverse", + "exponential" +}; +</pre> +</blockquote> + + + +<blockquote> +<pre> +// BiquadFilterNode constants for the .type attribute + +// Old way +const unsigned short LOWPASS = 0; +const unsigned short HIGHPASS = 1; +const unsigned short BANDPASS = 2; +const unsigned short LOWSHELF = 3; +const unsigned short HIGHSHELF = 4; +const unsigned short PEAKING = 5; +const unsigned short NOTCH = 6; +const unsigned short ALLPASS = 7; + +// New way +enum <dfn>BiquadFilterType</dfn> { + "lowpass", + "highpass", + "bandpass", + "lowshelf", + "highshelf", + "peaking", + "notch", + "allpass" +}; +</pre> +</blockquote> + +<blockquote> +<pre> +// OscillatorNode constants for the .type attribute + +// Old way +const unsigned short SINE = 0; +const unsigned short SQUARE = 1; +const unsigned short SAWTOOTH = 2; +const unsigned short TRIANGLE = 3; +const unsigned short CUSTOM = 4; + +// New way +enum <dfn>OscillatorType</dfn> { + "sine", + "square", + "sawtooth", + "triangle", + "custom" +}; +</pre> +</blockquote> + + + +</div> + +</div> +</div> + +<div class="appendix section" id="references"> +<h2 id="L17310">A.References</h2> + +<div class="section" id="normative-references"> +<h3 id="Normative-references">A.1 Normative references</h3> +<dl> + <dt id="DOM">[DOM] </dt> + <dd><a href="http://dom.spec.whatwg.org/">DOM</a>, + A. van Kesteren, A. Gregor, Ms2ger. WHATWG.</dd> + <dt id="HTML">[HTML] </dt> + <dd><a href="http://www.whatwg.org/specs/web-apps/current-work/multipage/">HTML</a>, + I. Hickson. WHATWG.</dd> + <dt id="RFC2119">[RFC2119] </dt> + <dd>S. Bradner. <a + href="http://www.ietf.org/rfc/rfc2119.txt"><cite><span>Key words for use + in RFCs to Indicate Requirement Levels.</span></cite></a> Internet RFC + 2119. URL: <a + href="http://www.ietf.org/rfc/rfc2119.txt">http://www.ietf.org/rfc/rfc2119.txt</a> + </dd> +</dl> +</div> + +<div class="section" id="informative-references"> +<h3 id="Informative-references">A.2 Informative references</h3> + +<p>No informative references.</p> +</div> +</div> + +<div class="section" id="acknowledgements"> +<h2 id="L17335">B.Acknowledgements</h2> + +<p>Special thanks to the W3C <a href="http://www.w3.org/2011/audio/">Audio +Working Group</a>. Members of the Working Group are (at the time of writing, +and by alphabetical order): <br /> +Berkovitz, Joe (public Invited expert);Cardoso, Gabriel (INRIA);Carlson, Eric +(Apple, Inc.);Gregan, Matthew (Mozilla Foundation);Jägenstedt, Philip (Opera +Software);Kalliokoski, Jussi (public Invited expert);Lowis, Chris (British +Broadcasting Corporation);MacDonald, Alistair (W3C Invited Experts);Michel, +Thierry (W3C/ERCIM);Noble, Jer (Apple, Inc.);O'Callahan, Robert(Mozilla +Foundation);Paradis, Matthew (British Broadcasting Corporation);Raman, T.V. +(Google, Inc.);Rogers, Chris (Google, Inc.);Schepers, Doug (W3C/MIT);Shires, +Glen (Google, Inc.);Smith, Michael (W3C/Keio);Thereaux, Olivier (British +Broadcasting Corporation);Wei, James (Intel Corporation);Wilson, Chris (Google, +Inc.); </p> +</div> + +<div class="section" id="ChangeLog-section"> +<h2 id="ChangeLog">C. Web Audio API Change Log</h2> +<pre> +user: crogers +date: Sun Dec 09 17:13:56 2012 -0800 +summary: Basic description of OfflineAudioContext + +user: crogers +date: Tue Dec 04 15:59:30 2012 -0800 +summary: minor correction to wording for minValue and maxValue + +user: crogers +date: Tue Dec 04 15:49:29 2012 -0800 +summary: Bug 20161: Make decodeAudioData neuter its array buffer argument when it begins decoding a buffer, and bring it back to normal when the decoding is finished + +user: crogers +date: Tue Dec 04 15:35:17 2012 -0800 +summary: Bug 20039: Refine description of audio decoding + +user: crogers +date: Tue Dec 04 15:23:07 2012 -0800 +summary: elaborate on decoding steps for AudioContext createBuffer() and decodeAudioData() + +user: crogers +date: Tue Dec 04 14:56:19 2012 -0800 +summary: Bug 19770: Note that if the last event for an AudioParam is a setCurveValue event, the computed value after that event will be equal to the latest curve value + +user: crogers +date: Tue Dec 04 14:48:04 2012 -0800 +summary: Bug 19769: Note that before the first automation event, the computed AudioParam value will be AudioParam.value + +user: crogers +date: Tue Dec 04 14:40:51 2012 -0800 +summary: Bug 19768: Explicitly mention that the initial value of AudioParam.value will be defaultValue + +user: crogers +date: Tue Dec 04 14:35:59 2012 -0800 +summary: Bug 19767: Explicitly mention that the 2nd component of AudioParam.computedValue will be 0 if there are no AudioNodes connected to it + +user: crogers +date: Tue Dec 04 14:30:08 2012 -0800 +summary: Bug 19764: Note in the spec that AudioParam.minValue/maxValue are merely informational + +user: crogers +date: Mon Dec 03 18:03:13 2012 -0800 +summary: Convert integer constants to Web IDL enum string constants + +user: crogers +date: Mon Dec 03 15:19:22 2012 -0800 +summary: Bug 17411: (AudioPannerNodeUnits): AudioPannerNode units are underspecified + +user: Ehsan Akhgari (Mozilla) +date: Thu Nov 29 15:59:38 2012 -0500 +summary: Change the Web IDL description of decodeAudioData arguments + +user: crogers +date: Wed Nov 14 13:24:01 2012 -0800 +summary: Bug 17393: (UseDoubles): float/double inconsistency + +user: crogers +date: Wed Nov 14 13:16:57 2012 -0800 +summary: Bug 17356: (AudioListenerOrientation): AudioListener.setOrientation vectors + +user: crogers +date: Wed Nov 14 12:56:06 2012 -0800 +summary: Bug 19957: PannerNode.coneGain is unused + +user: crogers +date: Wed Nov 14 12:40:46 2012 -0800 +summary: Bug 17412: AudioPannerNodeVectorNormalization): AudioPannerNode orientation normalization unspecified + +user: crogers +date: Wed Nov 14 12:16:41 2012 -0800 +summary: Bug 17411: (AudioPannerNodeUnits): AudioPannerNode units are underspecified + +user: crogers +date: Tue Nov 13 16:14:22 2012 -0800 +summary: be more explicit about maxDelayTime units + +user: crogers +date: Tue Nov 13 16:02:50 2012 -0800 +summary: Bug 19766: Clarify that reading AudioParam.computedValue will return the latest computed value for the latest audio quantum + +user: crogers +date: Tue Nov 13 15:47:25 2012 -0800 +summary: Bug 19872: Should specify the defaults for PannerNode's position, ... + +user: crogers +date: Tue Nov 13 15:27:53 2012 -0800 +summary: Bug 17390: (Joe Berkovitz): Loop start/stop points + +user: croger +date: Tue Nov 13 14:49:20 2012 -0800 +summary: Bug 19765: Note that setting AudioParam.value will be ignored when any automation events have been set on the object + +user: crogers +date: Tue Nov 13 14:39:07 2012 -0800 +summary: Bug 19873: Clarify PannerNode.listener + +user: crogers +date: Tue Nov 13 13:35:21 2012 -0800 +summary: Bug 19900: Clarify the default values for the AudioParam attributes of BiquadFilterNode + +user: crogers +date: Tue Nov 13 13:06:38 2012 -0800 +summary: Bug 19884: Specify the default value and ranges for the DynamicsCompressorNode AudioParam members + +user: crogers +date: Tue Nov 13 12:57:02 2012 -0800 +summary: Bug 19910: Disallow AudioContext.createDelay(max) where max <= 0 + +user: crogers +date: Mon Nov 12 12:02:18 2012 -0800 +summary: Add example code for more complex example + +user: Ehsan Akhgari (Mozilla) +date: Thu Nov 01 11:32:39 2012 -0400 +summary: Specify the default value for the AudioContext.createDelay() optional argument in Web IDL + +user: Ehsan Akhgari (Mozilla) +date: Tue Oct 30 20:29:48 2012 -0400 +summary: Mark the AudioParam members as readonly + +user: Ehsan Akhgari (Mozilla) +date: Tue Oct 30 20:24:52 2012 -0400 +summary: Make GainNode and DelayNode valid Web IDL + +user: crogers +date: Mon Oct 29 14:29:23 2012 -0700 +summary: consolidate AudioBufferSourceNode start() method + +user: crogers +date: Fri Oct 19 15:15:28 2012 -0700 +summary: Bug 18332: Node creation method naming inconsistencies + +user: crogers +date: Mon Oct 15 17:22:54 2012 -0700 +summary: Bug 17407: Interface naming inconsistency + +user: crogers +date: Tue Oct 09 17:21:19 2012 -0700 +summary: Bug 17369: Oscillator.detune attribute not defined + +user: crogers +date: Tue Oct 09 16:08:50 2012 -0700 +summary: Bug 17346: HTMLMediaElement integration + +user: crogers +date: Tue Oct 09 15:20:50 2012 -0700 +summary: Bug 17354: AudioListener default position, orientation and velocity + +user: crogers +date: Tue Oct 09 15:02:04 2012 -0700 +summary: Bug 17795: Behavior of multiple connections to same node needs to be explicitly defined + +user: crogers +date: Mon Oct 08 13:18:45 2012 -0700 +summary: Add missing AudioContext.createWaveShaper() method + +user: crogers +date: Fri Oct 05 18:13:44 2012 -0700 +summary: Bug 17399: AudioParam sampling is undefined + +user: crogers +date: Fri Oct 05 17:41:52 2012 -0700 +summary: Bug 17386: Realtime Analysis empty section + +user: crogers +date: Fri Oct 05 17:38:14 2012 -0700 +summary: minor tweak to down-mix section + +user: crogers +date: Fri Oct 05 17:35:05 2012 -0700 +summary: Bug 17380: Channel down mixing incomplete + +user: crogers +date: Fri Oct 05 15:40:57 2012 -0700 +summary: Bug 17375: MixerGainStructure should be marked as informative + +user: crogers +date: Fri Oct 05 14:29:20 2012 -0700 +summary: Bug 17381: (EventScheduling): Event Scheduling ('Need more detail here') + +user: crogers +date: Fri Oct 05 13:12:46 2012 -0700 +summary: Fix 18663: Need a method to get a readonly reading of the combined value when using AudioParam automation curve + +user: crogers +date: Fri Oct 05 12:48:36 2012 -0700 +summary: Fix 18662: Setting audioparam value while there is an automation curve will cancel that automation curve and set value immediately + +user: crogers +date: Fri Oct 05 12:26:28 2012 -0700 +summary: Fix 18661: Use startTime / endTime parameter names for AudioParam automation methods + +user: crogers +date: Wed Oct 03 12:26:39 2012 -0700 +summary: Specify default value for .distanceModel + +user: crogers +date: Tue Oct 02 12:33:36 2012 -0700 +summary: Fix Issues 17338 and 17337: AudioGain interface is not needed (Part 2) + +user: crogers +date: Tue Oct 02 12:28:55 2012 -0700 +summary: Fix Issues 17338 and 17337: AudioGain interface is not needed + +user: Ehsan Akhgari (Mozilla) +date: Wed Sep 26 18:22:36 2012 -0400 +summary: Make AudioBufferSourceNode.buffer nullable + +user: crogers +date: Tue Sep 25 12:56:14 2012 -0700 +summary: noteOn/noteOff changed to start/stop -- added deprecation notes + +user: Ehsan Akhgari (Mozilla) +date: Fri Aug 24 18:27:29 2012 -0400 +summary: Make the AudioContext object have a constructor + +user: Ehsan Akhgari (Mozilla) +date: Fri Aug 24 15:54:10 2012 -0400 +summary: Denote IDL definitions as Web IDL + +user: Ehsan Akhgari (Mozilla) +date: Fri Aug 24 15:04:37 2012 -0400 +summary: Use `long` instead of `int`, since the int type doesn't exist in Web IDL + +user: Ehsan Akhgari (Mozilla) +date: Fri Aug 24 15:02:43 2012 -0400 +summary: Add a missing attribute keyword in AudioProcessingEvent + +user: Ehsan Akhgari (Mozilla) +date: Tue Aug 21 15:36:48 2012 -0400 +summary: Remove the 'raises' notation from the IDLs + +user: crogers +date: Thu Aug 16 16:30:55 2012 -0700 +summary: Issue 17398: Add more detailed information about how AudioParam value is calculated + +user: crogers +date: Thu Aug 16 15:21:38 2012 -0700 +summary: another try with the style sheet + +user: crogers +date: Thu Aug 16 14:53:54 2012 -0700 +summary: use local style sheet to avoid https errors + +user: Ehsan Akhgari (Mozilla) +date: Wed Aug 15 23:05:49 2012 -0400 +summary: Replace the white-space based indentation of Web IDL code with a CSS-based one + +user: Ehsan Akhgari (Mozilla) +date: Wed Aug 15 22:56:03 2012 -0400 +summary: Remove more useless trailing whitespaces + +user: Ehsan Akhgari (Mozilla) +date: Wed Aug 15 22:47:21 2012 -0400 +summary: Remove the optional 'in' keyword from the Web IDL method declarations + +user: Ehsan Akhgari (Mozilla) +date: Wed Aug 15 22:42:03 2012 -0400 +summary: Add trailing semicolons for Web IDL interface declarations + +user: Ehsan Akhgari (Mozilla) +date: Wed Aug 15 22:37:32 2012 -0400 +summary: Remove useless trailing spaces + +user: Ehsan Akhgari (Mozilla) +date: Wed Aug 15 22:35:33 2012 -0400 +summary: Use the correct Web IDL notation for the AudioBufferCallback callback type + +user: Ehsan Akhgari (Mozilla) +date: Wed Aug 15 22:28:37 2012 -0400 +summary: Remove the extra semicolon in the IDL file for AudioContext + +user: Ehsan Akhgari (Mozilla) +date: Wed Aug 15 22:24:02 2012 -0400 +summary: Replace the old [Optional] IDL tag with the Web IDL optional keyword + +user: Ehsan Akhgari (Mozilla) +date: Tue Aug 14 10:18:19 2012 -0400 +summary: Empty changeset to test my commit access + +date: Mon Aug 13 13:26:52 2012 -0700 +* Integrate Thierry Michel's 3rd public working draft edits + +date: Tue Jun 26 15:56:31 2012 -0700 +* add MediaStreamAudioSourceNode + +date: Mon Jun 18 13:26:21 2012 -0700 +* minor formatting fix + +date: Mon Jun 18 13:19:34 2012 -0700 +* Add details for azimuth/elevation calculation + +date: Fri Jun 15 17:35:27 2012 -0700 +* Add equal-power-panning details + +date: Thu Jun 14 17:31:16 2012 -0700 +* Add equations for distance models + +date: Wed Jun 13 17:40:49 2012 -0700 +* Bug 17334: Add precise equations for AudioParam.setTargetValueAtTime() + +date: Fri Jun 08 17:44:26 2012 -0700 +* fix small typo + +date: Fri Jun 08 16:54:04 2012 -0700 +* Bug 17413: AudioBuffers' relationship to AudioContext + +date: Fri Jun 08 16:05:45 2012 -0700 +* Bug 17359: Add much more detail about ConvolverNode + +date: Fri Jun 08 12:59:29 2012 -0700 +* minor formatting fix + +date: Fri Jun 08 12:57:11 2012 -0700 +* Bug 17335: Add much more technical detail to setValueCurveAtTime() + +date: Wed Jun 06 16:34:43 2012 -0700 +*Add much more detail about parameter automation, including an example + +date: Mon Jun 04 17:25:08 2012 -0700 +* ISSUE-85: OscillatorNode folding considerations + +date: Mon Jun 04 17:02:20 2012 -0700 +* ISSUE-45: AudioGain scale underdefined + +date: Mon Jun 04 16:40:43 2012 -0700 +* ISSUE-41: AudioNode as input to AudioParam underdefined + +date: Mon Jun 04 16:14:48 2012 -0700 +* ISSUE-20: Relationship to currentTime + +date: Mon Jun 04 15:48:49 2012 -0700 +* ISSUE-94: Dynamic Lifetime + +date: Mon Jun 04 13:59:31 2012 -0700 +* ISSUE-42: add more detail about AudioParam sampling and block processing + +date: Mon Jun 04 12:28:48 2012 -0700 +* fix typo - minor edits + +date: Thu May 24 18:01:20 2012 -0700 +* ISSUE-69: add implementors guide for linear convolution + +date: Thu May 24 17:35:45 2012 -0700 +* ISSUE-49: better define AudioBuffer audio data access + +date: Thu May 24 17:15:29 2012 -0700 +* fix small typo + +date: Thu May 24 17:13:34 2012 -0700 +* ISSUE-24: define circular routing behavior + +date: Thu May 24 16:35:24 2012 -0700 +* ISSUE-42: specify a-rate or k-rate for each AudioParam + +date: Fri May 18 17:01:36 2012 -0700 +* ISSUE-53: noteOn and noteOff interaction + +date: Fri May 18 16:33:29 2012 -0700 +* ISSUE-34: Remove .name attribute from AudioParam + +date: Fri May 18 16:27:19 2012 -0700 +* ISSUE-33: Add maxNumberOfChannels attribute to AudioDestinationNode + +date: Fri May 18 15:50:08 2012 -0700 +* ISSUE-19: added more info about AudioBuffer - IEEE 32-bit + +date: Fri May 18 15:37:27 2012 -0700 +* ISSUE-29: remove reference to webkitAudioContext + +date: Fri Apr 27 12:36:54 2012 -0700 +* fix two small typos reported by James Wei + +date: Tue Apr 24 12:27:11 2012 -0700 +* small cleanup to ChannelSplitterNode and ChannelMergerNode + +date: Tue Apr 17 11:35:56 2012 -0700 +* small fix to createWaveTable() + +date: Tue Apr 13 2012 +* Cleanup AudioNode connect() and disconnect() method descriptions. +* Add AudioNode connect() to AudioParam method. + +date: Tue Apr 13 2012 +* Add OscillatorNode and WaveTable +* Define default values for optional arguments in createJavaScriptNode(), createChannelSplitter(), createChannelMerger() +* Define default filter type for BiquadFilterNode as LOWPASS + +date: Tue Apr 11 2012 +* add AudioContext .activeSourceCount attribute +* createBuffer() methods can throw exceptions +* add AudioContext method createMediaElementSource() +* update AudioContext methods createJavaScriptNode() (clean up description of parameters) +* update AudioContext method createChannelSplitter() (add numberOfOutputs parameter) +* update AudioContext method createChannelMerger() (add numberOfInputs parameter) +* update description of out-of-bounds AudioParam values (exception will not be thrown) +* remove AudioBuffer .gain attribute +* remove AudioBufferSourceNode .gain attribute +* remove AudioListener .gain attribute +* add AudioBufferSourceNode .playbackState attribute and state constants +* AnalyserNode no longer requires its output be connected to anything +* update ChannelMergerNode section describing numberOfOutputs (defaults to 6 but settable in constructor) +* update ChannelSplitterNode section describing numberOfInputs (defaults to 6 but settable in constructor) +* add note in Spatialization sections about potential to get arbitrary convolution matrixing + +date: Tue Apr 10 2012 +* Rebased editor's draft document based on edits from Thierry Michel (from 2nd public working draft). + +date: Tue Mar 13 12:13:41 2012 -0100 +* fixed all the HTML errors +* added ids to all Headings +* added alt attribute to all img +* fix broken anchors +* added a new status of this document section +* added mandatory spec headers +* generated a new table of content +* added a Reference section +* added an Acknowledgments section +* added a Web Audio API Change Log + +date: Fri Mar 09 15:12:42 2012 -0800 +* add optional maxDelayTime argument to createDelay() +* add more detail about playback state to AudioBufferSourceNode +* upgrade noteOn(), noteGrainOn(), noteOff() times to double from float + +date: Mon Feb 06 16:52:39 2012 -0800 +* Cleanup ScriptProcessorNode section +* Add distance model constants for PannerNode according to the OpenAL spec +* Add .normalize attribute to ConvolverNode +* Add getFrequencyResponse() method to BiquadFilterNode +* Tighten up the up-mix equations + +date: Fri Nov 04 15:40:58 2011 -0700 +summary: Add more technical detail to BiquadFilterNode description (contributed by Raymond Toy) + +date: Sat Oct 15 19:08:15 2011 -0700 +summary: small edits to the introduction + +date: Sat Oct 15 19:00:15 2011 -0700 +summary: initial commit + +date: Tue Sep 13 12:49:11 2011 -0700 +summary: add convolution reverb design document + +date: Mon Aug 29 17:05:58 2011 -0700 +summary: document the decodeAudioData() method + +date: Mon Aug 22 14:36:33 2011 -0700 +summary: fix broken MediaElementAudioSourceNode link + +date: Mon Aug 22 14:33:57 2011 -0700 +summary: refine section describing integration with HTMLMediaElement + +date: Mon Aug 01 12:05:53 2011 -0700 +summary: add Privacy section + +date: Mon Jul 18 17:53:50 2011 -0700 +summary: small update - tweak musical applications thumbnail images + +date: Mon Jul 18 17:23:00 2011 -0700 +summary: initial commit of Web Audio API specification</pre> +</div> +</body> +</html> |